Attachment 347839 Details for Bug 188745 – reduced patch that contains only the changes after resyncing libwebrtc

[patch] reduced patch that contains only the changes after resyncing libwebrtc

patch-after-libwebrtc-resync (text/plain), 844.46 KB, created by youenn fablet on 2018-08-22 13:35:46 PDT

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Creator: youenn fablet

Created: 2018-08-22 13:35:46 PDT

Size: 844.46 KB

patch

obsolete

>diff --git a/LayoutTests/ChangeLog b/LayoutTests/ChangeLog
>index b54f89ec9ec..c687430a12a 100644
>--- a/LayoutTests/ChangeLog
>+++ b/LayoutTests/ChangeLog
>@@ -1,3 +1,15 @@
>+2018-08-20  Youenn Fablet  <youenn@apple.com>
>+
>+        Update libwebrtc up to 984f1a80c0
>+        https://bugs.webkit.org/show_bug.cgi?id=188745
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Updated tests according new webrtc backend.
>+
>+        * webrtc/libwebrtc/setLocalDescriptionCrash.html:
>+        * webrtc/video-getParameters.html:
>+
> 2018-08-21  Yusuke Suzuki  <yusukesuzuki@slowstart.org>
> 
>         Support "name" option for dedicated workers
>diff --git a/LayoutTests/imported/w3c/ChangeLog b/LayoutTests/imported/w3c/ChangeLog
>index 14e2c7c12e3..0e4dda7c09e 100644
>--- a/LayoutTests/imported/w3c/ChangeLog
>+++ b/LayoutTests/imported/w3c/ChangeLog
>@@ -1,3 +1,14 @@
>+2018-08-21  Youenn Fablet  <youenn@apple.com>
>+
>+        Update libwebrtc up to 984f1a80c0
>+        https://bugs.webkit.org/show_bug.cgi?id=188745
>+        <rdar://problem/43539177>
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        * web-platform-tests/webrtc/RTCPeerConnection-addIceCandidate-expected.txt:
>+        * web-platform-tests/webrtc/no-media-call-expected.txt:
>+
> 2018-08-21  Yusuke Suzuki  <yusukesuzuki@slowstart.org>
> 
>         Support "name" option for dedicated workers
>diff --git a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addIceCandidate-expected.txt b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addIceCandidate-expected.txt
>index ca4912e349b..259885390dc 100644
>--- a/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addIceCandidate-expected.txt
>+++ b/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCPeerConnection-addIceCandidate-expected.txt
>@@ -1,25 +1,25 @@
> 
>-FAIL Add null candidate should reject with TypeError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>+FAIL Add null candidate should reject with TypeError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
> FAIL Add ICE candidate before setting remote description should reject with InvalidStateError assert_unreached: Should have rejected: undefined Reached unreachable code
>-FAIL Add ICE candidate after setting remote description should succeed promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Add ICE candidate with RTCIceCandidate should succeed promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Add candidate with only valid sdpMid should succeed promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Add candidate with only valid sdpMLineIndex should succeed promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL addIceCandidate with first sdpMid and sdpMLineIndex add candidate to first media stream promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL addIceCandidate with second sdpMid and sdpMLineIndex should add candidate to second media stream promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Add candidate for first media stream with null ufrag should add candidate to first media stream promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Adding multiple candidates should add candidates to their corresponding media stream promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Add with empty candidate string (end of candidate) should succeed promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Add candidate with both sdpMid and sdpMLineIndex manually set to null should reject with TypeError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Add candidate with only valid candidate string should reject with TypeError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Add candidate with invalid candidate string and both sdpMid and sdpMLineIndex null should reject with TypeError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Add candidate with empty dict should reject with TypeError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Add candidate with manually filled default values should reject with TypeError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Add candidate with invalid sdpMid should reject with OperationError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Add candidate with invalid sdpMLineIndex should reject with OperationError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Invalid sdpMLineIndex should be ignored if valid sdpMid is provided promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Add candidate for media stream 2 with null ufrag should succeed promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Add candidate with invalid ufrag should reject with OperationError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Add candidate with invalid candidate string should reject with OperationError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>-FAIL Add candidate with sdpMid belonging to different ufrag should reject with OperationError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set remote video description send parameters.."
>+FAIL Add ICE candidate after setting remote description should succeed promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Add ICE candidate with RTCIceCandidate should succeed promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Add candidate with only valid sdpMid should succeed promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Add candidate with only valid sdpMLineIndex should succeed promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL addIceCandidate with first sdpMid and sdpMLineIndex add candidate to first media stream promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL addIceCandidate with second sdpMid and sdpMLineIndex should add candidate to second media stream promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Add candidate for first media stream with null ufrag should add candidate to first media stream promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Adding multiple candidates should add candidates to their corresponding media stream promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Add with empty candidate string (end of candidate) should succeed promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Add candidate with both sdpMid and sdpMLineIndex manually set to null should reject with TypeError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Add candidate with only valid candidate string should reject with TypeError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Add candidate with invalid candidate string and both sdpMid and sdpMLineIndex null should reject with TypeError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Add candidate with empty dict should reject with TypeError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Add candidate with manually filled default values should reject with TypeError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Add candidate with invalid sdpMid should reject with OperationError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Add candidate with invalid sdpMLineIndex should reject with OperationError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Invalid sdpMLineIndex should be ignored if valid sdpMid is provided promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Add candidate for media stream 2 with null ufrag should succeed promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Add candidate with invalid ufrag should reject with OperationError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Add candidate with invalid candidate string should reject with OperationError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
>+FAIL Add candidate with sdpMid belonging to different ufrag should reject with OperationError promise_test: Unhandled rejection with value: object "OperationError: Failed to set remote offer sdp: Failed to set remote video description send parameters."
> 
>diff --git a/LayoutTests/imported/w3c/web-platform-tests/webrtc/no-media-call-expected.txt b/LayoutTests/imported/w3c/web-platform-tests/webrtc/no-media-call-expected.txt
>index 033022f81bb..15d01b9e6a5 100644
>--- a/LayoutTests/imported/w3c/web-platform-tests/webrtc/no-media-call-expected.txt
>+++ b/LayoutTests/imported/w3c/web-platform-tests/webrtc/no-media-call-expected.txt
>@@ -4,5 +4,4 @@ Harness Error (TIMEOUT), message = null
> TIMEOUT Can set up a basic WebRTC call with no data. Test timed out
> iceConnectionState info
> 
>-First: checking
>-Second: new
>+
>diff --git a/LayoutTests/webrtc/libwebrtc/setLocalDescriptionCrash.html b/LayoutTests/webrtc/libwebrtc/setLocalDescriptionCrash.html
>index 7a0142c2a47..886a97c52df 100644
>--- a/LayoutTests/webrtc/libwebrtc/setLocalDescriptionCrash.html
>+++ b/LayoutTests/webrtc/libwebrtc/setLocalDescriptionCrash.html
>@@ -10,7 +10,7 @@
>         <script>
> function badifyAnswer(answer)
> {
>-    return {type: answer.type, sdp: answer.sdp.replace("BUNDLE video", "BUNDLE")};
>+    return {type: answer.type, sdp: answer.sdp.replace("v=0", "v")};
> }
> 
> promise_test((test) => {
>@@ -29,9 +29,9 @@ promise_test((test) => {
>         badAnswer = badifyAnswer(answer);
>         return pc2.setLocalDescription(badAnswer);
>     }).then(assert_unreached, (e) => {
>-        assert_equals(e.message, "Failed to set local answer sdp: Failed to enable BUNDLE.");
>+        assert_equals(e.message, "Expect line: v=");
>         return pc2.setLocalDescription(badAnswer).then(assert_unreached, (e) => {
>-            assert_equals(e.message, "Failed to set local answer sdp: no pending remote description.");
>+            assert_equals(e.message, "Expect line: v=");
>         });
>     });
> }, "Testing calling twice setLocalDescription with a bad SDP");
>diff --git a/LayoutTests/webrtc/video-getParameters.html b/LayoutTests/webrtc/video-getParameters.html
>index c9c949f8a5e..3b269080b3d 100644
>--- a/LayoutTests/webrtc/video-getParameters.html
>+++ b/LayoutTests/webrtc/video-getParameters.html
>@@ -45,6 +45,9 @@ promise_test((test) => {
>         receiverParameters.encodings[0].fec.ssrc = 1;
>         receiverParameters.encodings[0].rtx.ssrc = 1;
> 
>+        senderParameters.transactionId = "";
>+        receiverParameters.transactionId = "";
>+
>         assert_equals(JSON.stringify(senderParameters), JSON.stringify(receiverParameters), "testing sender vs. receiver parameters");
> 
>         senderParameters.encodings[0].ssrc = 1;
>diff --git a/Source/ThirdParty/libwebrtc/CMakeLists.txt b/Source/ThirdParty/libwebrtc/CMakeLists.txt
>index 550cf2a16de..5552a6ef63f 100644
>--- a/Source/ThirdParty/libwebrtc/CMakeLists.txt
>+++ b/Source/ThirdParty/libwebrtc/CMakeLists.txt
>@@ -17,6 +17,11 @@ if (NOT ALSALIB_FOUND)
> 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/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
>     Source/third_party/boringssl/err_data.c
>     Source/third_party/boringssl/src/crypto/asn1/a_bitstr.c
>     Source/third_party/boringssl/src/crypto/asn1/a_bool.c
>@@ -294,17 +299,14 @@ set(webrtc_SOURCES
>     Source/third_party/libyuv/source/rotate_any.cc
>     Source/third_party/libyuv/source/rotate_argb.cc
>     Source/third_party/libyuv/source/rotate_common.cc
>-    Source/third_party/libyuv/source/rotate_dspr2.cc
>     Source/third_party/libyuv/source/rotate_gcc.cc
>     Source/third_party/libyuv/source/row_any.cc
>     Source/third_party/libyuv/source/row_common.cc
>-    Source/third_party/libyuv/source/row_dspr2.cc
>     Source/third_party/libyuv/source/row_gcc.cc
>     Source/third_party/libyuv/source/scale.cc
>     Source/third_party/libyuv/source/scale_any.cc
>     Source/third_party/libyuv/source/scale_argb.cc
>     Source/third_party/libyuv/source/scale_common.cc
>-    Source/third_party/libyuv/source/scale_dspr2.cc
>     Source/third_party/libyuv/source/scale_gcc.cc
>     Source/third_party/libyuv/source/video_common.cc
>     Source/third_party/opus/src/celt/bands.c
>@@ -442,6 +444,8 @@ set(webrtc_SOURCES
>     Source/third_party/opus/src/src/opus_multistream_decoder.c
>     Source/third_party/opus/src/src/opus_multistream_encoder.c
>     Source/third_party/opus/src/src/repacketizer.c
>+    Source/third_party/rnnoise/src/kiss_fft.cc
>+    Source/third_party/rnnoise/src/rnn_vad_weights.cc
>     Source/third_party/usrsctp/usrsctplib/netinet/sctp_asconf.c
>     Source/third_party/usrsctp/usrsctplib/netinet/sctp_auth.c
>     Source/third_party/usrsctp/usrsctplib/netinet/sctp_bsd_addr.c
>@@ -465,8 +469,10 @@ set(webrtc_SOURCES
>     Source/third_party/usrsctp/usrsctplib/user_mbuf.c
>     Source/third_party/usrsctp/usrsctplib/user_recv_thread.c
>     Source/third_party/usrsctp/usrsctplib/user_socket.c
>+    Source/webrtc/api/audio/audio_frame.cc
>     Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.cc
>     Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc
>+    Source/webrtc/api/audio_codecs/audio_codec_pair_id.cc
>     Source/webrtc/api/audio_codecs/audio_decoder.cc
>     Source/webrtc/api/audio_codecs/audio_encoder.cc
>     Source/webrtc/api/audio_codecs/audio_format.cc
>@@ -485,46 +491,87 @@ set(webrtc_SOURCES
>     Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.cc
>     Source/webrtc/api/audio_codecs/opus/audio_encoder_opus.cc
>     Source/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc
>+    Source/webrtc/api/audio_options.cc
>+    Source/webrtc/api/call/transport.cc
>     Source/webrtc/api/candidate.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/optional.cc
>+    Source/webrtc/api/peerconnectioninterface.cc
>     Source/webrtc/api/proxy.cc
>     Source/webrtc/api/rtcerror.cc
>     Source/webrtc/api/rtp_headers.cc
>     Source/webrtc/api/rtpparameters.cc
>+    Source/webrtc/api/rtpreceiverinterface.cc
>+    Source/webrtc/api/rtptransceiverinterface.cc
>     Source/webrtc/api/statstypes.cc
>-    Source/webrtc/api/umametrics.cc
>+    Source/webrtc/api/transport/bitrate_settings.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/color_space.cc
>+    Source/webrtc/api/video/encoded_frame.cc
>+    Source/webrtc/api/video/i010_buffer.cc
>     Source/webrtc/api/video/i420_buffer.cc
>+    Source/webrtc/api/video/video_bitrate_allocation.cc
>     Source/webrtc/api/video/video_content_type.cc
>     Source/webrtc/api/video/video_frame.cc
>     Source/webrtc/api/video/video_frame_buffer.cc
>+    Source/webrtc/api/video/video_source_interface.cc
>+    Source/webrtc/api/video/video_stream_decoder_create.cc
>+    Source/webrtc/api/video/video_stream_encoder_create.cc
>     Source/webrtc/api/video/video_timing.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/sdp_video_format.cc
>+    Source/webrtc/api/video_codecs/video_codec.cc
>+    Source/webrtc/api/video_codecs/video_decoder.cc
>+    Source/webrtc/api/video_codecs/video_decoder_software_fallback_wrapper.cc
>     Source/webrtc/api/video_codecs/video_encoder.cc
>-    Source/webrtc/api/videosourceinterface.cc
>+    Source/webrtc/api/video_codecs/video_encoder_config.cc
>+    Source/webrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc
>+    Source/webrtc/audio/audio_level.cc
>     Source/webrtc/audio/audio_receive_stream.cc
>     Source/webrtc/audio/audio_send_stream.cc
>     Source/webrtc/audio/audio_state.cc
>     Source/webrtc/audio/audio_transport_impl.cc
>+    Source/webrtc/audio/channel.cc
>+    Source/webrtc/audio/channel_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
>     Source/webrtc/call/audio_send_stream.cc
>+    Source/webrtc/call/audio_state.cc
>     Source/webrtc/call/bitrate_allocator.cc
>     Source/webrtc/call/call.cc
>+    Source/webrtc/call/call_config.cc
>     Source/webrtc/call/callfactory.cc
>+    Source/webrtc/call/degraded_call.cc
>+    Source/webrtc/call/fake_network_pipe.cc
>+    Source/webrtc/call/flexfec_receive_stream.cc
>     Source/webrtc/call/flexfec_receive_stream_impl.cc
>+    Source/webrtc/call/packet_receiver.cc
>+    Source/webrtc/call/receive_time_calculator.cc
>     Source/webrtc/call/rtcp_demuxer.cc
>+    Source/webrtc/call/rtp_bitrate_configurator.cc
>     Source/webrtc/call/rtp_config.cc
>     Source/webrtc/call/rtp_demuxer.cc
>+    Source/webrtc/call/rtp_payload_params.cc
>     Source/webrtc/call/rtp_rtcp_demuxer_helper.cc
>     Source/webrtc/call/rtp_stream_receiver_controller.cc
>     Source/webrtc/call/rtp_transport_controller_send.cc
>+    Source/webrtc/call/rtp_video_sender.cc
>     Source/webrtc/call/rtx_receive_stream.cc
>+    Source/webrtc/call/simulated_network.cc
>     Source/webrtc/call/syncable.cc
>-    Source/webrtc/call/video_config.cc
>     Source/webrtc/call/video_receive_stream.cc
>     Source/webrtc/call/video_send_stream.cc
>     Source/webrtc/common_audio/audio_converter.cc
>@@ -532,7 +579,6 @@ set(webrtc_SOURCES
>     Source/webrtc/common_audio/audio_util.cc
>     Source/webrtc/common_audio/blocker.cc
>     Source/webrtc/common_audio/channel_buffer.cc
>-    Source/webrtc/common_audio/fft4g.c
>     Source/webrtc/common_audio/fir_filter_c.cc
>     Source/webrtc/common_audio/fir_filter_factory.cc
>     Source/webrtc/common_audio/lapped_transform.cc
>@@ -573,12 +619,13 @@ set(webrtc_SOURCES
>     Source/webrtc/common_audio/signal_processing/spl_init.c
>     Source/webrtc/common_audio/signal_processing/spl_inl.c
>     Source/webrtc/common_audio/signal_processing/spl_sqrt.c
>-    Source/webrtc/common_audio/signal_processing/spl_sqrt_floor.c
>     Source/webrtc/common_audio/signal_processing/splitting_filter.c
>     Source/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
>     Source/webrtc/common_audio/signal_processing/vector_scaling_operations.c
>     Source/webrtc/common_audio/smoothing_filter.cc
>     Source/webrtc/common_audio/sparse_fir_filter.cc
>+    Source/webrtc/common_audio/third_party/fft4g/fft4g.c
>+    Source/webrtc/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.c
>     Source/webrtc/common_audio/vad/vad.cc
>     Source/webrtc/common_audio/vad/vad_core.c
>     Source/webrtc/common_audio/vad/vad_filterbank.c
>@@ -588,7 +635,6 @@ set(webrtc_SOURCES
>     Source/webrtc/common_audio/wav_file.cc
>     Source/webrtc/common_audio/wav_header.cc
>     Source/webrtc/common_audio/window_generator.cc
>-    Source/webrtc/common_types.cc
>     Source/webrtc/common_video/bitrate_adjuster.cc
>     Source/webrtc/common_video/h264/h264_bitstream_parser.cc
>     Source/webrtc/common_video/h264/h264_common.cc
>@@ -608,6 +654,8 @@ set(webrtc_SOURCES
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.cc
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.cc
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.cc
>+    Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.cc
>+    Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.cc
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.cc
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.cc
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.cc
>@@ -617,13 +665,16 @@ set(webrtc_SOURCES
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.cc
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.cc
>+    Source/webrtc/logging/rtc_event_log/icelogger.cc
>     Source/webrtc/logging/rtc_event_log/output/rtc_event_log_output_file.cc
>     Source/webrtc/logging/rtc_event_log/rtc_event_log.cc
>     Source/webrtc/logging/rtc_event_log/rtc_event_log_factory.cc
>+    Source/webrtc/logging/rtc_event_log/rtc_event_log_impl.cc
>     Source/webrtc/logging/rtc_event_log/rtc_stream_config.cc
>     Source/webrtc/media/base/adaptedvideotracksource.cc
>     Source/webrtc/media/base/codec.cc
>     Source/webrtc/media/base/h264_profile_level_id.cc
>+    Source/webrtc/media/base/mediachannel.cc
>     Source/webrtc/media/base/mediaconstants.cc
>     Source/webrtc/media/base/mediaengine.cc
>     Source/webrtc/media/base/rtpdataengine.cc
>@@ -646,8 +697,6 @@ set(webrtc_SOURCES
>     Source/webrtc/media/engine/scopedvideoencoder.cc
>     Source/webrtc/media/engine/simulcast.cc
>     Source/webrtc/media/engine/simulcast_encoder_adapter.cc
>-    Source/webrtc/media/engine/videodecodersoftwarefallbackwrapper.cc
>-    Source/webrtc/media/engine/videoencodersoftwarefallbackwrapper.cc
>     Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.cc
>     Source/webrtc/media/engine/webrtcmediaengine.cc
>     Source/webrtc/media/engine/webrtcvideocapturer.cc
>@@ -681,12 +730,9 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc
>     Source/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
>     Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
>-    Source/webrtc/modules/audio_coding/codecs/g711/g711.c
>     Source/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
>     Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
>     Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
>-    Source/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
>-    Source/webrtc/modules/audio_coding/codecs/g722/g722_encode.c
>     Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
>     Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
>     Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
>@@ -802,12 +848,11 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/encode.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
>-    Source/webrtc/modules/audio_coding/codecs/isac/main/source/fft.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c
>-    Source/webrtc/modules/audio_coding/codecs/isac/main/source/filterbank_tables.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/filterbanks.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/intialize.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
>+    Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/lattice.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c
>@@ -838,8 +883,6 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_coding/neteq/comfort_noise.cc
>     Source/webrtc/modules/audio_coding/neteq/cross_correlation.cc
>     Source/webrtc/modules/audio_coding/neteq/decision_logic.cc
>-    Source/webrtc/modules/audio_coding/neteq/decision_logic_fax.cc
>-    Source/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
>     Source/webrtc/modules/audio_coding/neteq/decoder_database.cc
>     Source/webrtc/modules/audio_coding/neteq/delay_manager.cc
>     Source/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc
>@@ -847,6 +890,7 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc
>     Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
>     Source/webrtc/modules/audio_coding/neteq/expand.cc
>+    Source/webrtc/modules/audio_coding/neteq/expand_uma_logger.cc
>     Source/webrtc/modules/audio_coding/neteq/merge.cc
>     Source/webrtc/modules/audio_coding/neteq/nack_tracker.cc
>     Source/webrtc/modules/audio_coding/neteq/neteq.cc
>@@ -911,7 +955,6 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_processing/aec3/matched_filter.cc
>     Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc
>     Source/webrtc/modules/audio_processing/aec3/matrix_buffer.cc
>-    Source/webrtc/modules/audio_processing/aec3/output_selector.cc
>     Source/webrtc/modules/audio_processing/aec3/render_buffer.cc
>     Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.cc
>     Source/webrtc/modules/audio_processing/aec3/render_delay_controller.cc
>@@ -929,38 +972,54 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc
>     Source/webrtc/modules/audio_processing/agc/agc.cc
>     Source/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
>+    Source/webrtc/modules/audio_processing/agc/utility.cc
>     Source/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
>     Source/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
>     Source/webrtc/modules/audio_processing/agc/loudness_histogram.cc
>-    Source/webrtc/modules/audio_processing/agc/utility.cc
>-    Source/webrtc/modules/audio_processing/agc2/gain_controller2.cc
>+    Source/webrtc/modules/audio_processing/agc2/adaptive_agc.cc
>+    Source/webrtc/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
>+    Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
>+    Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.cc
>+    Source/webrtc/modules/audio_processing/agc2/biquad_filter.cc
>+    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
>+    Source/webrtc/modules/audio_processing/agc2/noise_spectrum_estimator.cc
>+    Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
>+    Source/webrtc/modules/audio_processing/agc2/rnn_vad/fft_util.cc
>+    Source/webrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
>+    Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
>+    Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
>+    Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
>+    Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.cc
>+    Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
>+    Source/webrtc/modules/audio_processing/agc2/saturation_protector.cc
>+    Source/webrtc/modules/audio_processing/agc2/signal_classifier.cc
>+    Source/webrtc/modules/audio_processing/agc2/vad_with_level.cc
>+    Source/webrtc/modules/audio_processing/agc2/vector_float_frame.cc
>     Source/webrtc/modules/audio_processing/audio_buffer.cc
>     Source/webrtc/modules/audio_processing/audio_processing_impl.cc
>-    Source/webrtc/modules/audio_processing/beamformer/array_util.cc
>-    Source/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc
>-    Source/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
>     Source/webrtc/modules/audio_processing/echo_cancellation_impl.cc
>+    Source/webrtc/modules/audio_processing/echo_cancellation_proxy.cc
>     Source/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
>+    Source/webrtc/modules/audio_processing/echo_control_mobile_proxy.cc
>     Source/webrtc/modules/audio_processing/echo_detector/circular_buffer.cc
>     Source/webrtc/modules/audio_processing/echo_detector/mean_variance_estimator.cc
>     Source/webrtc/modules/audio_processing/echo_detector/moving_max.cc
>     Source/webrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.cc
>     Source/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc
>     Source/webrtc/modules/audio_processing/gain_control_impl.cc
>+    Source/webrtc/modules/audio_processing/gain_controller2.cc
>     Source/webrtc/modules/audio_processing/include/aec_dump.cc
>     Source/webrtc/modules/audio_processing/include/audio_processing.cc
>     Source/webrtc/modules/audio_processing/include/audio_processing_statistics.cc
>     Source/webrtc/modules/audio_processing/include/config.cc
>-    Source/webrtc/modules/audio_processing/level_controller/biquad_filter.cc
>-    Source/webrtc/modules/audio_processing/level_controller/down_sampler.cc
>-    Source/webrtc/modules/audio_processing/level_controller/gain_applier.cc
>-    Source/webrtc/modules/audio_processing/level_controller/gain_selector.cc
>-    Source/webrtc/modules/audio_processing/level_controller/level_controller.cc
>-    Source/webrtc/modules/audio_processing/level_controller/noise_level_estimator.cc
>-    Source/webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.cc
>-    Source/webrtc/modules/audio_processing/level_controller/peak_level_estimator.cc
>-    Source/webrtc/modules/audio_processing/level_controller/saturating_gain_estimator.cc
>-    Source/webrtc/modules/audio_processing/level_controller/signal_classifier.cc
>     Source/webrtc/modules/audio_processing/level_estimator_impl.cc
>     Source/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
>     Source/webrtc/modules/audio_processing/low_cut_filter.cc
>@@ -992,25 +1051,37 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_processing/voice_detection_impl.cc
>     Source/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
>     Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
>-    Source/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.cc
>-    Source/webrtc/modules/congestion_controller/bitrate_estimator.cc
>-    Source/webrtc/modules/congestion_controller/delay_based_bwe.cc
>-    Source/webrtc/modules/congestion_controller/median_slope_estimator.cc
>-    Source/webrtc/modules/congestion_controller/probe_bitrate_estimator.cc
>-    Source/webrtc/modules/congestion_controller/probe_controller.cc
>+    Source/webrtc/modules/congestion_controller/bbr/bandwidth_sampler.cc
>+    Source/webrtc/modules/congestion_controller/bbr/bbr_factory.cc
>+    Source/webrtc/modules/congestion_controller/bbr/bbr_network_controller.cc
>+    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/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/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
>     Source/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
>     Source/webrtc/modules/congestion_controller/transport_feedback_adapter.cc
>-    Source/webrtc/modules/congestion_controller/trendline_estimator.cc
>-    Source/webrtc/modules/media_file/media_file_impl.cc
>-    Source/webrtc/modules/media_file/media_file_utility.cc
>-    Source/webrtc/modules/pacing/alr_detector.cc
>     Source/webrtc/modules/pacing/bitrate_prober.cc
>     Source/webrtc/modules/pacing/interval_budget.cc
>     Source/webrtc/modules/pacing/paced_sender.cc
>     Source/webrtc/modules/pacing/packet_queue.cc
>-    Source/webrtc/modules/pacing/packet_queue2.cc
>+    Source/webrtc/modules/pacing/packet_queue_interface.cc
>     Source/webrtc/modules/pacing/packet_router.cc
>+    Source/webrtc/modules/pacing/round_robin_packet_queue.cc
>     Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
>     Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h
>     Source/webrtc/modules/remote_bitrate_estimator/bwe_defines.cc
>@@ -1020,9 +1091,11 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
>     Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
>     Source/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
>-    Source/webrtc/modules/remote_bitrate_estimator/send_time_history.cc
>     Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
>+    Source/webrtc/modules/rtp_rtcp/source/contributing_sources.cc
>     Source/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
>+    Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.cc
>+    Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.cc
>     Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc
>     Source/webrtc/modules/rtp_rtcp/source/flexfec_receiver.cc
>     Source/webrtc/modules/rtp_rtcp/source/flexfec_sender.cc
>@@ -1067,15 +1140,17 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/rtp_rtcp/source/rtp_format.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
>-    Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc
>+    Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc
>+    Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.cc
>+    Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
>@@ -1086,11 +1161,17 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
>+    Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.cc
>     Source/webrtc/modules/rtp_rtcp/source/time_util.cc
>     Source/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
>     Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.cc
>     Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc
>     Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc
>+    Source/webrtc/modules/third_party/fft/fft.c
>+    Source/webrtc/modules/third_party/g711/g711.c
>+    Source/webrtc/modules/third_party/g722/g722_decode.c
>+    Source/webrtc/modules/third_party/g722/g722_encode.c
>+    Source/webrtc/modules/third_party/portaudio/pa_ringbuffer.c
>     Source/webrtc/modules/utility/source/process_thread_impl.cc
>     Source/webrtc/modules/video_capture/device_info_impl.cc
>     Source/webrtc/modules/video_capture/external/device_info_external.cc
>@@ -1099,20 +1180,17 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/video_capture/linux/video_capture_linux.cc
>     Source/webrtc/modules/video_capture/video_capture_factory.cc
>     Source/webrtc/modules/video_capture/video_capture_impl.cc
>-    Source/webrtc/modules/video_coding/codec_database.cc
>     Source/webrtc/modules/video_coding/codec_timer.cc
>     Source/webrtc/modules/video_coding/codecs/h264/h264.cc
>     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/simulcast_rate_allocator.cc
>     Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.cc
>-    Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
>-    Source/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.cc
>-    Source/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
>-    Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
>+    Source/webrtc/modules/video_coding/decoder_database.cc
>     Source/webrtc/modules/video_coding/decoding_state.cc
>     Source/webrtc/modules/video_coding/encoded_frame.cc
>+    Source/webrtc/modules/video_coding/encoder_database.cc
>+    Source/webrtc/modules/video_coding/fec_controller_default.cc
>     Source/webrtc/modules/video_coding/frame_buffer.cc
>     Source/webrtc/modules/video_coding/frame_buffer2.cc
>     Source/webrtc/modules/video_coding/frame_object.cc
>@@ -1129,7 +1207,6 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/video_coding/nack_module.cc
>     Source/webrtc/modules/video_coding/packet.cc
>     Source/webrtc/modules/video_coding/packet_buffer.cc
>-    Source/webrtc/modules/video_coding/protection_bitrate_calculator.cc
>     Source/webrtc/modules/video_coding/qp_parser.cc
>     Source/webrtc/modules/video_coding/receiver.cc
>     Source/webrtc/modules/video_coding/rtp_frame_reference_finder.cc
>@@ -1142,9 +1219,12 @@ set(webrtc_SOURCES
>     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
>     Source/webrtc/modules/video_coding/utility/vp8_header_parser.cc
>     Source/webrtc/modules/video_coding/utility/vp9_uncompressed_header_parser.cc
>     Source/webrtc/modules/video_coding/video_codec_initializer.cc
>+    Source/webrtc/modules/video_coding/video_coding_defines.cc
>     Source/webrtc/modules/video_coding/video_coding_impl.cc
>     Source/webrtc/modules/video_coding/video_receiver.cc
>     Source/webrtc/modules/video_coding/video_sender.cc
>@@ -1154,6 +1234,7 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/video_processing/util/skin_detection.cc
>     Source/webrtc/modules/video_processing/video_denoiser.cc
>     Source/webrtc/p2p/base/asyncstuntcpsocket.cc
>+    Source/webrtc/p2p/base/basicasyncresolverfactory.cc
>     Source/webrtc/p2p/base/basicpacketsocketfactory.cc
>     Source/webrtc/p2p/base/dtlstransport.cc
>     Source/webrtc/p2p/base/dtlstransportinternal.cc
>@@ -1167,9 +1248,9 @@ set(webrtc_SOURCES
>     Source/webrtc/p2p/base/portallocator.cc
>     Source/webrtc/p2p/base/portinterface.cc
>     Source/webrtc/p2p/base/pseudotcp.cc
>+    Source/webrtc/p2p/base/regatheringcontroller.cc
>     Source/webrtc/p2p/base/relayport.cc
>     Source/webrtc/p2p/base/relayserver.cc
>-    Source/webrtc/p2p/base/session.cc
>     Source/webrtc/p2p/base/stun.cc
>     Source/webrtc/p2p/base/stunport.cc
>     Source/webrtc/p2p/base/stunrequest.cc
>@@ -1181,16 +1262,13 @@ set(webrtc_SOURCES
>     Source/webrtc/p2p/base/turnserver.cc
>     Source/webrtc/p2p/base/udptransport.cc
>     Source/webrtc/p2p/client/basicportallocator.cc
>-    Source/webrtc/p2p/client/socketmonitor.cc
>     Source/webrtc/p2p/client/turnportfactory.cc
>     Source/webrtc/p2p/stunprober/stunprober.cc
>-    Source/webrtc/pc/audiomonitor.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/currentspeakermonitor.cc
>     Source/webrtc/pc/datachannel.cc
>     Source/webrtc/pc/dtlssrtptransport.cc
>     Source/webrtc/pc/dtmfsender.cc
>@@ -1199,8 +1277,8 @@ set(webrtc_SOURCES
>     Source/webrtc/pc/jsepicecandidate.cc
>     Source/webrtc/pc/jsepsessiondescription.cc
>     Source/webrtc/pc/jseptransport.cc
>+    Source/webrtc/pc/jseptransportcontroller.cc
>     Source/webrtc/pc/localaudiosource.cc
>-    Source/webrtc/pc/mediamonitor.cc
>     Source/webrtc/pc/mediasession.cc
>     Source/webrtc/pc/mediastream.cc
>     Source/webrtc/pc/mediastreamobserver.cc
>@@ -1209,7 +1287,9 @@ set(webrtc_SOURCES
>     Source/webrtc/pc/remoteaudiosource.cc
>     Source/webrtc/pc/rtcpmuxfilter.cc
>     Source/webrtc/pc/rtcstatscollector.cc
>+    Source/webrtc/pc/rtcstatstraversal.cc
>     Source/webrtc/pc/rtpmediautils.cc
>+    Source/webrtc/pc/rtpparametersconversion.cc
>     Source/webrtc/pc/rtpreceiver.cc
>     Source/webrtc/pc/rtpsender.cc
>     Source/webrtc/pc/rtptransceiver.cc
>@@ -1222,7 +1302,7 @@ set(webrtc_SOURCES
>     Source/webrtc/pc/srtptransport.cc
>     Source/webrtc/pc/statscollector.cc
>     Source/webrtc/pc/trackmediainfomap.cc
>-    Source/webrtc/pc/transportcontroller.cc
>+    Source/webrtc/pc/transportstats.cc
>     Source/webrtc/pc/videocapturertracksource.cc
>     Source/webrtc/pc/videotrack.cc
>     Source/webrtc/pc/videotracksource.cc
>@@ -1234,7 +1314,6 @@ set(webrtc_SOURCES
>     Source/webrtc/rtc_base/asyncsocket.cc
>     Source/webrtc/rtc_base/asynctcpsocket.cc
>     Source/webrtc/rtc_base/asyncudpsocket.cc
>-    Source/webrtc/rtc_base/base64.cc
>     Source/webrtc/rtc_base/bitbuffer.cc
>     Source/webrtc/rtc_base/bitrateallocationstrategy.cc
>     Source/webrtc/rtc_base/bufferqueue.cc
>@@ -1245,9 +1324,14 @@ set(webrtc_SOURCES
>     Source/webrtc/rtc_base/crc32.cc
>     Source/webrtc/rtc_base/criticalsection.cc
>     Source/webrtc/rtc_base/cryptstring.cc
>+    Source/webrtc/rtc_base/data_rate_limiter.cc
>     Source/webrtc/rtc_base/event.cc
>     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/field_trial_parser.cc
>+    Source/webrtc/rtc_base/experiments/field_trial_units.cc
>+    Source/webrtc/rtc_base/experiments/quality_scaling_experiment.cc
>     Source/webrtc/rtc_base/file.cc
>     Source/webrtc/rtc_base/file_posix.cc
>     Source/webrtc/rtc_base/filerotatingstream.cc
>@@ -1264,8 +1348,7 @@ set(webrtc_SOURCES
>     Source/webrtc/rtc_base/location.cc
>     Source/webrtc/rtc_base/logging.cc
>     Source/webrtc/rtc_base/logsinks.cc
>-    Source/webrtc/rtc_base/md5.cc
>-    Source/webrtc/rtc_base/md5digest.cc
>+    Source/webrtc/rtc_base/memory/aligned_malloc.cc
>     Source/webrtc/rtc_base/memory_usage.cc
>     Source/webrtc/rtc_base/messagedigest.cc
>     Source/webrtc/rtc_base/messagehandler.cc
>@@ -1281,52 +1364,63 @@ set(webrtc_SOURCES
>     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/sample_counter.cc
>     Source/webrtc/rtc_base/openssladapter.cc
>+    Source/webrtc/rtc_base/opensslcertificate.cc
>     Source/webrtc/rtc_base/openssldigest.cc
>     Source/webrtc/rtc_base/opensslidentity.cc
>+    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/physicalsocketserver.cc
>     Source/webrtc/rtc_base/platform_file.cc
>     Source/webrtc/rtc_base/platform_thread.cc
>+    Source/webrtc/rtc_base/platform_thread_types.cc
>     Source/webrtc/rtc_base/proxyinfo.cc
>     Source/webrtc/rtc_base/proxyserver.cc
>     Source/webrtc/rtc_base/race_checker.cc
>     Source/webrtc/rtc_base/random.cc
>     Source/webrtc/rtc_base/rate_limiter.cc
>     Source/webrtc/rtc_base/rate_statistics.cc
>-    Source/webrtc/rtc_base/ratelimiter.cc
>     Source/webrtc/rtc_base/ratetracker.cc
>     Source/webrtc/rtc_base/rtccertificate.cc
>     Source/webrtc/rtc_base/rtccertificategenerator.cc
>     Source/webrtc/rtc_base/sequenced_task_checker_impl.cc
>-    Source/webrtc/rtc_base/sha1.cc
>-    Source/webrtc/rtc_base/sha1digest.cc
>     Source/webrtc/rtc_base/signalthread.cc
>-    Source/webrtc/rtc_base/sigslot.cc
>+    Source/webrtc/rtc_base/socket.cc
>     Source/webrtc/rtc_base/socketadapters.cc
>     Source/webrtc/rtc_base/socketaddress.cc
>     Source/webrtc/rtc_base/socketaddresspair.cc
>     Source/webrtc/rtc_base/socketstream.cc
>     Source/webrtc/rtc_base/ssladapter.cc
>+    Source/webrtc/rtc_base/sslcertificate.cc
>     Source/webrtc/rtc_base/sslfingerprint.cc
>     Source/webrtc/rtc_base/sslidentity.cc
>     Source/webrtc/rtc_base/sslstreamadapter.cc
>     Source/webrtc/rtc_base/stream.cc
>     Source/webrtc/rtc_base/string_to_number.cc
>     Source/webrtc/rtc_base/stringencode.cc
>+    Source/webrtc/rtc_base/strings/audio_format_to_string.cc
>+    Source/webrtc/rtc_base/strings/string_builder.cc
>     Source/webrtc/rtc_base/stringutils.cc
>+    Source/webrtc/rtc_base/synchronization/rw_lock_posix.cc
>+    Source/webrtc/rtc_base/synchronization/rw_lock_wrapper.cc
>+    Source/webrtc/rtc_base/system/file_wrapper.cc
>     Source/webrtc/rtc_base/task_queue_libevent.cc
>     Source/webrtc/rtc_base/task_queue_posix.cc
>+    Source/webrtc/rtc_base/third_party/base64/base64.cc
>+    Source/webrtc/rtc_base/third_party/sigslot/sigslot.cc
>     Source/webrtc/rtc_base/thread.cc
>     Source/webrtc/rtc_base/thread_checker_impl.cc
>+    Source/webrtc/rtc_base/time/timestamp_extrapolator.cc
>     Source/webrtc/rtc_base/timestampaligner.cc
>     Source/webrtc/rtc_base/timeutils.cc
>-    Source/webrtc/rtc_base/transformadapter.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
>     Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis_lib.cc
>     Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis.cc
>     Source/webrtc/rtc_tools/frame_editing/frame_editing_lib.cc
>@@ -1334,7 +1428,6 @@ set(webrtc_SOURCES
>     Source/webrtc/stats/rtcstats.cc
>     Source/webrtc/stats/rtcstats_objects.cc
>     Source/webrtc/stats/rtcstatsreport.cc
>-    Source/webrtc/system_wrappers/source/aligned_malloc.cc
>     Source/webrtc/system_wrappers/source/clock.cc
>     Source/webrtc/system_wrappers/source/cpu_features.cc
>     Source/webrtc/system_wrappers/source/cpu_features_linux.c
>@@ -1342,18 +1435,13 @@ set(webrtc_SOURCES
>     Source/webrtc/system_wrappers/source/event.cc
>     Source/webrtc/system_wrappers/source/event_timer_posix.cc
>     Source/webrtc/system_wrappers/source/field_trial_default.cc
>-    Source/webrtc/system_wrappers/source/file_impl.cc
>     Source/webrtc/system_wrappers/source/metrics_default.cc
>     Source/webrtc/system_wrappers/source/rtp_to_ntp_estimator.cc
>     Source/webrtc/system_wrappers/source/runtime_enabled_features_default.cc
>-    Source/webrtc/system_wrappers/source/rw_lock.cc
>-    Source/webrtc/system_wrappers/source/rw_lock_posix.cc
>     Source/webrtc/system_wrappers/source/sleep.cc
>-    Source/webrtc/system_wrappers/source/timestamp_extrapolator.cc
>     Source/webrtc/video/call_stats.cc
>     Source/webrtc/video/encoder_rtcp_feedback.cc
>     Source/webrtc/video/overuse_frame_detector.cc
>-    Source/webrtc/video/payload_router.cc
>     Source/webrtc/video/quality_threshold.cc
>     Source/webrtc/video/receive_statistics_proxy.cc
>     Source/webrtc/video/report_block_stats.cc
>@@ -1364,15 +1452,13 @@ set(webrtc_SOURCES
>     Source/webrtc/video/stats_counter.cc
>     Source/webrtc/video/stream_synchronization.cc
>     Source/webrtc/video/transport_adapter.cc
>+    Source/webrtc/video/video_quality_observer.cc
>     Source/webrtc/video/video_receive_stream.cc
>     Source/webrtc/video/video_send_stream.cc
>+    Source/webrtc/video/video_send_stream_impl.cc
>     Source/webrtc/video/video_stream_decoder.cc
>+    Source/webrtc/video/video_stream_decoder_impl.cc
>     Source/webrtc/video/video_stream_encoder.cc
>-    Source/webrtc/voice_engine/audio_level.cc
>-    Source/webrtc/voice_engine/channel.cc
>-    Source/webrtc/voice_engine/channel_proxy.cc
>-    Source/webrtc/voice_engine/transport_feedback_packet_loss_tracker.cc
>-    Source/webrtc/voice_engine/utility.cc
>     $<TARGET_OBJECTS:libsrtp>
> )
> 
>@@ -1429,6 +1515,8 @@ target_compile_definitions(webrtc PRIVATE
>   WEBRTC_USE_BUILTIN_ISAC_FIX=1
>   WEBRTC_USE_BUILTIN_ISAC_FLOAT=0
>   WTF_USE_DYNAMIC_ANNOTATIONS=1
>+  LINUX_ALSA
>+  RTC_DISABLE_VP9
>   _GNU_SOURCE
>   __Userspace__
>   __Userspace_os_Linux
>@@ -1436,6 +1524,7 @@ target_compile_definitions(webrtc PRIVATE
> 
> target_include_directories(webrtc PRIVATE
>     Source
>+    Source/third_party/abseil-cpp
>     Source/third_party/boringssl/src/include
>     Source/third_party/jsoncpp/overrides/include
>     Source/third_party/jsoncpp/source/include
>diff --git a/Source/ThirdParty/libwebrtc/ChangeLog b/Source/ThirdParty/libwebrtc/ChangeLog
>index 06163977c73..d288f96b6dc 100644
>--- a/Source/ThirdParty/libwebrtc/ChangeLog
>+++ b/Source/ThirdParty/libwebrtc/ChangeLog
>@@ -1,3 +1,73 @@
>+2018-08-20  Youenn Fablet  <youenn@apple.com> and Alejandro G. Castro  <alex@igalia.com>
>+
>+        Update libwebrtc up to 984f1a80c0
>+        https://bugs.webkit.org/show_bug.cgi?id=188745
>+        <rdar://problem/43539177>
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Update libwebrtc main code.
>+        Update exported symbols and related applied modifications.
>+
>+        * CMakeLists.txt:
>+        * Configurations/libwebrtc.iOS.exp:
>+        * Configurations/libwebrtc.iOSsim.exp:
>+        * Configurations/libwebrtc.mac.exp:
>+        * Configurations/libwebrtc.xcconfig:
>+        * Source/webrtc: refreshed
>+        * WebKit/0001-Updating-webrtc.patch: Added.
>+        * WebKit/0001-Adapting-libwebrtc-H264-codec.patch: Removed.
>+        * WebKit/0001-Disable-SIGPIPE-for-WebRTC-sockets.patch: Removed.
>+        * WebKit/0001-Update-RTCVideoEncoderH264.mm-for-WebKit.patch: Removed.
>+        * WebKit/0001-Using-VCP.patch: Removed.
>+        * WebKit/0003-Fixing-VP8-files.patch: Removed.
>+        * WebKit/0004-Removing-parameter-names-from-files-included-from-We.patch: Removed.
>+        * WebKit/0005-Fix-RTC_FATAL.patch: Removed.
>+        * WebKit/0006-Disabling-VP8.patch: Removed.
>+        * WebKit/0007-Fix-RTC_STRINGIZE.patch: Removed.
>+        * WebKit/0008-Fix-sanitizer.patch: Removed.
>+        * WebKit/0009-Remove-dispatch_set_target_queue.patch: Removed.
>+        * WebKit/0010-Fix-RTCVideoEncoderH264-CVPixelBuffer-leak.patch: Removed.
>+        * WebKit/0011-Fix-AudioDeviceID-array-leak.patch: Removed.
>+        * WebKit/0012-Add-WK-prefix-to-Objective-C-classes-and-protocols.patch: Removed.
>+        * WebKit/0013-Fix-SafeSetError-use-after-move.patch: Removed.
>+        * libwebrtc.xcodeproj/project.pbxproj:
>+
>+2018-08-20  Youenn Fablet  <youenn@apple.com>
>+
>+        Update libwebrtc up to 984f1a80c0
>+        https://bugs.webkit.org/show_bug.cgi?id=188745
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Updated webrtc to 984f1a80c0.
>+        Updated libyuv accordingly.
>+        Added abseil-cpp and rnnoise third party source code.
>+        Removed WebKit specific patches that are no longer needed.
>+
>+        * Configurations/libwebrtc.iOS.exp: Updated to include new required symbols.
>+        * Configurations/libwebrtc.iOSsim.exp: Ditto.
>+        * Configurations/libwebrtc.mac.exp: Ditto.
>+        * Source/third_party/abseil-cpp: Added.
>+        * Source/third_party/rnnoise: Added.
>+        * Source/webrtc: Updated.
>+        * WebKit/0001-Adapting-libwebrtc-H264-codec.patch: Removed.
>+        * WebKit/0001-Disable-SIGPIPE-for-WebRTC-sockets.patch: Removed.
>+        * WebKit/0001-Update-RTCVideoEncoderH264.mm-for-WebKit.patch: Removed.
>+        * WebKit/0001-Using-VCP.patch: Removed.
>+        * WebKit/0003-Fixing-VP8-files.patch: Removed.
>+        * WebKit/0004-Removing-parameter-names-from-files-included-from-We.patch: Removed.
>+        * WebKit/0005-Fix-RTC_FATAL.patch: Removed.
>+        * WebKit/0006-Disabling-VP8.patch: Removed.
>+        * WebKit/0007-Fix-RTC_STRINGIZE.patch: Removed.
>+        * WebKit/0008-Fix-sanitizer.patch: Removed.
>+        * WebKit/0009-Remove-dispatch_set_target_queue.patch: Removed.
>+        * WebKit/0010-Fix-RTCVideoEncoderH264-CVPixelBuffer-leak.patch: Removed.
>+        * WebKit/0011-Fix-AudioDeviceID-array-leak.patch: Removed.
>+        * WebKit/0012-Add-WK-prefix-to-Objective-C-classes-and-protocols.patch: Removed.
>+        * WebKit/0013-Fix-SafeSetError-use-after-move.patch: Removed.
>+        * libwebrtc.xcodeproj/project.pbxproj:
>+
> 2018-08-21  Youenn Fablet  <youenn@apple.com>
> 
>         Update some libwebrtc third party libraries as per libwebrtc 984f1a80c0c
>@@ -552,7 +622,54 @@
>         * Source/webrtc/sdk/WebKit/WebKitUtilities.mm:
>         (webrtc::pixelBufferToFrame):
> 
>-2018-04-02  Alejandro G. Castro  <alex@igalia.com>
>+2018-04-02  Alejandro G. Castro  <a2018-08-20  Youenn Fablet  <youenn@apple.com>
>+
>+        Update libwebrtc up to 984f1a80c0
>+        https://bugs.webkit.org/show_bug.cgi?id=188745
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Covered by existing tests.
>+
>+        Updated implementation according new webrtc backend.
>+        Instead of modifying libwebrtc header files, we disable unused parameter warning for such headers included in WebCore.
>+        WebCore implementation is updated according new webrtc API, mostly AddRef/Release.
>+        RealtimeOutgoingVideoSource now uses a pixel convolver to convert RGB pixel buffers as I420.
>+
>+        * Configurations/Base.xcconfig:
>+        * Configurations/WebCoreTestSupport.xcconfig:
>+        * Modules/mediastream/libwebrtc/LibWebRTCDataChannelHandler.h:
>+        * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
>+        (WebCore::LibWebRTCMediaEndpoint::addTrack):
>+        (WebCore::LibWebRTCMediaEndpoint::mediaStreamFromRTCStream):
>+        (WebCore::fillEncodingParameters):
>+        (WebCore::fillRtpParameters):
>+        * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
>+        (WebCore::LibWebRTCMediaEndpoint::Release const):
>+        * WebCore.xcodeproj/project.pbxproj:
>+        * platform/mediastream/RealtimeIncomingAudioSource.h:
>+        * platform/mediastream/RealtimeIncomingVideoSource.h:
>+        * platform/mediastream/RealtimeOutgoingAudioSource.h:
>+        * platform/mediastream/RealtimeOutgoingVideoSource.h:
>+        * platform/mediastream/libwebrtc/LibWebRTCAudioModule.h:
>+        * platform/mediastream/libwebrtc/LibWebRTCProvider.cpp:
>+        * platform/mediastream/libwebrtc/LibWebRTCProvider.h:
>+        * platform/mediastream/mac/MockRealtimeVideoSourceMac.mm:
>+        (WebCore::MockRealtimeVideoSourceMac::CMSampleBufferFromPixelBuffer):
>+        * platform/mediastream/mac/RealtimeIncomingVideoSourceCocoa.mm:
>+        * platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.cpp:
>+        (WebCore::ConvertToI420):
>+        (WebCore::RealtimeOutgoingVideoSourceCocoa::sampleBufferUpdated):
>+        * platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.h:
>+        * platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.mm: Copied from Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.h.
>+        (WebCore::RealtimeOutgoingVideoSourceCocoa::convertToYUV):
>+        * testing/MockLibWebRTCPeerConnection.cpp:
>+        (WebCore::MockLibWebRTCPeerConnection::AddTrack):
>+        * testing/MockLibWebRTCPeerConnection.h:
>+        (WebCore::MockRtpSender::GetParameters):
>+        (WebCore::MockRtpSender::SetParameters):
>+
>+lex@igalia.com>
> 
>         Unreviewed fixing GTK port X86 32bits compilation after r230152.
> 
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/Base.xcconfig b/Source/ThirdParty/libwebrtc/Configurations/Base.xcconfig
>index b04182e31f8..8796a86a93a 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/Base.xcconfig
>+++ b/Source/ThirdParty/libwebrtc/Configurations/Base.xcconfig
>@@ -88,7 +88,7 @@ OTHER_CFLAGS = $(ASAN_OTHER_CFLAGS) -fvisibility=default;
> OTHER_CPLUSPLUSFLAGS = $(ASAN_OTHER_CPLUSPLUSFLAGS) -fvisibility=default;
> OTHER_LDFLAGS = $(ASAN_OTHER_LDFLAGS) -fvisibility=default;
> 
>-GCC_PREPROCESSOR_DEFINITIONS = $(DEBUG_DEFINES) $(inherited);
>+GCC_PREPROCESSOR_DEFINITIONS = $(DEBUG_DEFINES) $(inherited) ABSL_ALLOCATOR_NOTHROW;
> 
> EXCLUDED_INSTALLSRC_SUBDIRECTORY_PATTERNS = $(inherited) "$(PROJECT_DIR)/Source/webrtc/examples";
> 
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp
>index 89f4e8b8a9c..8478b284152 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp
>@@ -1,7 +1,4 @@
> __ZN3rtc10LogMessage10LogToDebugENS_15LoggingSeverityE
>-__ZN3rtc12FatalMessageC1EPKci
>-__ZN3rtc12FatalMessageC1EPKciPNSt3__112basic_stringIcNS3_11char_traitsIcEENS3_9allocatorIcEEEE
>-__ZN3rtc12FatalMessageD1Ev
> __ZN3rtc14MessageHandlerD2Ev
> __ZN3rtc14VideoSinkWantsC1Ev
> __ZN3rtc14VideoSinkWantsD1Ev
>@@ -9,7 +6,6 @@ __ZN3rtc17CopyOnWriteBuffer21CloneDataIfReferencedEm
> __ZN3rtc17CopyOnWriteBufferC1ERKS0_
> __ZN3rtc17CopyOnWriteBufferC1Emm
> __ZN3rtc17CopyOnWriteBufferD1Ev
>-__ZN3rtc6Thread14InvokeInternalERKNS_8LocationEPNS_14MessageHandlerE
> __ZN3rtc6Thread22CreateWithSocketServerEv
> __ZN3rtc6Thread5StartEPNS_8RunnableE
> __ZN3rtc6Thread6CreateEv
>@@ -34,15 +30,11 @@ __ZN6webrtc16ConvertVideoTypeENS_9VideoTypeE
> __ZN6webrtc18CreateIceCandidateERKNSt3__112basic_stringIcNS0_11char_traitsIcEENS0_9allocatorIcEEEEiS8_PNS_13SdpParseErrorE
> __ZN6webrtc19RTCCertificateStats5kTypeE
> __ZN6webrtc19RTCDataChannelStats5kTypeE
>-__ZN6webrtc20CoreVideoFrameBuffer6ToI420Ev
>-__ZN6webrtc20CoreVideoFrameBufferC2EP10__CVBuffer
>-__ZN6webrtc20CoreVideoFrameBufferD2Ev
> __ZN6webrtc24CreateSessionDescriptionERKNSt3__112basic_stringIcNS0_11char_traitsIcEENS0_9allocatorIcEEEES8_PNS_13SdpParseErrorE
> __ZN6webrtc24RTCIceCandidatePairStats5kTypeE
> __ZN6webrtc24RTCInboundRTPStreamStats5kTypeE
> __ZN6webrtc24RTCMediaStreamTrackStats5kTypeE
> __ZN6webrtc25RTCOutboundRTPStreamStats5kTypeE
>-__ZN6webrtc27CreatePeerConnectionFactoryEPN3rtc6ThreadES2_S2_NS0_13scoped_refptrINS_17AudioDeviceModuleEEENS3_INS_19AudioEncoderFactoryEEENS3_INS_19AudioDecoderFactoryEEENSt3__110unique_ptrINS_19VideoEncoderFactoryENSA_14default_deleteISC_EEEENSB_INS_19VideoDecoderFactoryENSD_ISG_EEEENS3_INS_10AudioMixerEEENS3_INS_15AudioProcessingEEE
> __ZN6webrtc27SessionDescriptionInterface6kOfferE
> __ZN6webrtc27SessionDescriptionInterface7kAnswerE
> __ZN6webrtc27SessionDescriptionInterface9kPrAnswerE
>@@ -50,7 +42,6 @@ __ZN6webrtc8internal21SynchronousMethodCall6InvokeERKN3rtc8LocationEPNS2_6Thread
> __ZN6webrtc8internal21SynchronousMethodCallC1EPN3rtc14MessageHandlerE
> __ZN6webrtc8internal21SynchronousMethodCallD1Ev
> __ZN7cricket17MediaTypeToStringENS_9MediaTypeE
>-__ZN7cricket18BasicPortAllocatorC1EPN3rtc14NetworkManagerEPNS1_19PacketSocketFactoryEPN6webrtc14TurnCustomizerEPNS_25RelayPortFactoryInterfaceE
> __ZNK6webrtc10VideoFrame18video_frame_bufferEv
> __ZNK6webrtc10VideoFrame5widthEv
> __ZNK6webrtc10VideoFrame6heightEv
>@@ -59,9 +50,6 @@ __ZNK6webrtc14RTCStatsReport13ConstIteratorneERKS1_
> __ZNK6webrtc14RTCStatsReport13ConstIteratorptEv
> __ZNK6webrtc14RTCStatsReport3endEv
> __ZNK6webrtc14RTCStatsReport5beginEv
>-__ZNK6webrtc20CoreVideoFrameBuffer4typeEv
>-__ZNK6webrtc20CoreVideoFrameBuffer5widthEv
>-__ZNK6webrtc20CoreVideoFrameBuffer6heightEv
> __ZNK6webrtc8RTCStats6ToJsonEv
> __ZTVN3rtc14MessageHandlerE
> _ConvertToI420
>@@ -95,13 +83,11 @@ __ZN3rtc13SocketAddressC1ERKS0_
> __ZN3rtc13SocketAddressaSERKS0_
> __ZNK3rtc13SocketAddress6ipaddrEv
> __ZNK3rtc18NetworkManagerBase11GetNetworksEPNSt3__16vectorIPNS_7NetworkENS1_9allocatorIS4_EEEE
>-__ZN3rtc19PacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
> __ZN3rtc22PacketTimeUpdateParamsC1Ev
> __ZN3rtc17AsyncPacketSocketD2Ev
> __ZN3rtc9ProxyInfoD1Ev
> __ZN3rtc18NetworkManagerBaseC2Ev
> __ZN3rtc7NetworkD1Ev
>-__ZTVN3rtc19PacketSocketFactoryE
> __ZN3rtc22PacketTimeUpdateParamsD1Ev
> __ZN3rtc22AsyncResolverInterfaceC2Ev
> __ZN3rtc18NetworkManagerBase16MergeNetworkListERKNSt3__16vectorIPNS_7NetworkENS1_9allocatorIS4_EEEEPb
>@@ -118,9 +104,6 @@ __ZN6webrtc32CreateBuiltinAudioDecoderFactoryEv
> __ZN6webrtc32CreateBuiltinAudioEncoderFactoryEv
> __ZN6webrtc27SessionDescriptionInterface16RemoveCandidatesERKNSt3__16vectorIN7cricket9CandidateENS1_9allocatorIS4_EEEE
> __ZNK6webrtc21IceCandidateInterface10server_urlEv
>-__ZNK6webrtc27SessionDescriptionInterface7GetTypeEv
>-__ZTVN6webrtc21IceCandidateInterfaceE
>-__ZTVN6webrtc27SessionDescriptionInterfaceE
> __ZN6webrtc20setApplicationStatusEb
> __ZN6webrtc32createVideoToolboxDecoderFactoryEv
> __ZN6webrtc32createVideoToolboxEncoderFactoryEv
>@@ -128,6 +111,98 @@ __ZN6webrtc29setH264HardwareEncoderAllowedEb
> __ZN6webrtc20pixelBufferFromFrameERKNS_10VideoFrameE
> __ZN6webrtc18pixelBufferToFrameEP10__CVBuffer
> __ZN3rtc24BasicPacketSocketFactory19CreateAsyncResolverEv
>-__ZN3rtc24BasicPacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
> __ZN3rtc24BasicPacketSocketFactoryC2Ev
> __ZN3rtc24BasicPacketSocketFactoryD2Ev
>+__ZNK3rtc6Thread9IsCurrentEv
>+__ZN3rtc6Thread14InvokeInternalERKNS_8LocationEPNS_14MessageHandlerE
>+__ZN3rtc5EventD1Ev
>+__ZN3rtc5EventC1Ebb
>+__ZN3rtc5Event4WaitEi
>+__ZN3rtc5Event3SetEv
>+__ZN3rtc18CreateRandomStringEm
>+__ZN6webrtc8RTCError2OKEv
>+__ZN3rtc18webrtc_checks_impl8FatalLogEPKciS2_PKNS0_12CheckArgTypeEz
>+__ZTVN6webrtc32CreateSessionDescriptionObserverE
>+__ZTVN6webrtc30PeerConnectionFactoryInterfaceE
>+__ZTVN6webrtc29SetSessionDescriptionObserverE
>+__ZNK3rtc17ThreadCheckerImpl19CalledOnValidThreadEv
>+__ZN7cricket18BasicPortAllocatorC1EPN3rtc14NetworkManagerEPNS1_19PacketSocketFactoryEPN6webrtc14TurnCustomizerEPNS_25RelayPortFactoryInterfaceE
>+__ZN7cricket12AudioOptionsD1Ev
>+__ZN6webrtc32CreateSessionDescriptionObserver9OnFailureENS_8RTCErrorE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceEPN7cricket13VideoCapturerEPKNS_25MediaConstraintsInterfaceE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceEPN7cricket13VideoCapturerE
>+__ZN6webrtc29SetSessionDescriptionObserver9OnFailureENS_8RTCErrorE
>+__ZN6webrtc28RtpHeaderExtensionCapabilityD1Ev
>+__ZN6webrtc26PeerConnectionDependenciesD1Ev
>+__ZN3rtc24BasicPacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
>+__ZN6webrtc12RtcpFeedbackC1ERKS0_
>+__ZN6webrtc12RtcpFeedbackD1Ev
>+__ZN6webrtc14I420BufferPoolC1Ev
>+__ZN6webrtc14I420BufferPoolD1Ev
>+__ZN6webrtc15RtpCapabilitiesC1Ev
>+__ZN6webrtc15RtpCapabilitiesD1Ev
>+__ZN6webrtc18RtpCodecCapabilityD1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1ERKS1_
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationD1Ev
>+__ZN6webrtc23PeerConnectionInterface9IceServerC1ERKS1_
>+__ZN6webrtc23PeerConnectionInterface9IceServerC1Ev
>+__ZN6webrtc23PeerConnectionInterface9IceServerD1Ev
>+__ZN6webrtc26PeerConnectionDependenciesC1EOS0_
>+__ZN3rtc18webrtc_checks_impl8FatalLogEPKciS2_PKNS0_12CheckArgTypeEz
>+__ZN6webrtc12RtcpFeedbackC1ERKS0_
>+__ZN6webrtc12RtcpFeedbackD1Ev
>+__ZN6webrtc14I420BufferPoolC1Ev
>+__ZN6webrtc14I420BufferPoolD1Ev
>+__ZN6webrtc15RtpCapabilitiesC1Ev
>+__ZN6webrtc15RtpCapabilitiesD1Ev
>+__ZN6webrtc18RtpCodecCapabilityD1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1ERKS1_
>+__ZTVN6webrtc27SessionDescriptionInterfaceE
>+__ZTVN6webrtc23PeerConnectionInterfaceE
>+__ZTVN6webrtc21IceCandidateInterfaceE
>+__ZTVN6webrtc20DataChannelInterfaceE
>+__ZTVN6webrtc19VideoTrackInterfaceE
>+__ZTVN6webrtc19AudioTrackInterfaceE
>+__ZNK6webrtc30PeerConnectionFactoryInterface26GetRtpReceiverCapabilitiesEN7cricket9MediaTypeE
>+__ZNK6webrtc30PeerConnectionFactoryInterface24GetRtpSenderCapabilitiesEN7cricket9MediaTypeE
>+__ZNK6webrtc27SessionDescriptionInterface7GetTypeEv
>+__ZNK6webrtc23PeerConnectionInterface26pending_remote_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface26current_remote_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface25pending_local_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface25current_local_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface15GetTransceiversEv
>+__ZNK6webrtc23PeerConnectionInterface12GetReceiversEv
>+__ZNK6webrtc23PeerConnectionInterface10GetSendersEv
>+__ZNK6webrtc20DataChannelInterface8protocolEv
>+__ZNK6webrtc20DataChannelInterface17maxRetransmitTimeEv
>+__ZNK6webrtc20DataChannelInterface14maxRetransmitsEv
>+__ZNK6webrtc20DataChannelInterface10negotiatedEv
>+__ZNK6webrtc19VideoTrackInterface12content_hintEv
>+__ZN6webrtc30PeerConnectionFactoryInterface20CreatePeerConnectionERKNS_23PeerConnectionInterface16RTCConfigurationENS_26PeerConnectionDependenciesE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceENSt3__110unique_ptrIN7cricket13VideoCapturerENS1_14default_deleteIS4_EEEEPKNS_25MediaConstraintsInterfaceE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceENSt3__110unique_ptrIN7cricket13VideoCapturerENS1_14default_deleteIS4_EEEE
>+__ZN6webrtc23PeerConnectionInterface19RemoveIceCandidatesERKNSt3__16vectorIN7cricket9CandidateENS1_9allocatorIS4_EEEE
>+__ZN6webrtc23PeerConnectionInterface16StartRtcEventLogEix
>+__ZN6webrtc23PeerConnectionInterface16StartRtcEventLogENSt3__110unique_ptrINS_17RtcEventLogOutputENS1_14default_deleteIS3_EEEEx
>+__ZN6webrtc23PeerConnectionInterface16SetConfigurationERKNS0_16RTCConfigurationEPNS_8RTCErrorE
>+__ZN6webrtc23PeerConnectionInterface16SetConfigurationERKNS0_16RTCConfigurationE
>+__ZN6webrtc19AudioTrackInterface14GetSignalLevelEPi
>+__ZN6webrtc19AudioTrackInterface17GetAudioProcessorEv
>+__ZN6webrtc23PeerConnectionInterface10SetBitrateERKNS_15BitrateSettingsE
>+__ZN6webrtc23PeerConnectionInterface12CreateSenderERKNSt3__112basic_stringIcNS1_11char_traitsIcEENS1_9allocatorIcEEEES9_
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN3rtc13scoped_refptrINS_25MediaStreamTrackInterfaceEEE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN3rtc13scoped_refptrINS_25MediaStreamTrackInterfaceEEERKNS_18RtpTransceiverInitE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN7cricket9MediaTypeE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN7cricket9MediaTypeERKNS_18RtpTransceiverInitE
>+__ZN6webrtc23PeerConnectionInterface16GetConfigurationEv
>+__ZN6webrtc23PeerConnectionInterface14RemoveTrackNewEN3rtc13scoped_refptrINS_18RtpSenderInterfaceEEE
>+__ZN3rtc10SentPacketC1Exx
>+__ZN3rtc13PacketOptionsC1ERKS0_
>+__ZN3rtc10PacketInfoD1Ev
>+__ZN3rtc10PacketInfoC1ERKS0_
>+__ZTVN3rtc19PacketSocketFactoryE
>+__ZN3rtc13PacketOptionsD1Ev
>+__ZN3rtc19PacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
>+__ZN3rtc13PacketOptionsC1Ev
>+__ZN6webrtc27CreatePeerConnectionFactoryEPN3rtc6ThreadES2_S2_NS0_13scoped_refptrINS_17AudioDeviceModuleEEENS3_INS_19AudioEncoderFactoryEEENS3_INS_19AudioDecoderFactoryEEENSt3__110unique_ptrINS_19VideoEncoderFactoryENSA_14default_deleteISC_EEEENSB_INS_19VideoDecoderFactoryENSD_ISG_EEEENS3_INS_10AudioMixerEEENS3_INS_15AudioProcessingEEE
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp
>index 89f4e8b8a9c..8478b284152 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp
>@@ -1,7 +1,4 @@
> __ZN3rtc10LogMessage10LogToDebugENS_15LoggingSeverityE
>-__ZN3rtc12FatalMessageC1EPKci
>-__ZN3rtc12FatalMessageC1EPKciPNSt3__112basic_stringIcNS3_11char_traitsIcEENS3_9allocatorIcEEEE
>-__ZN3rtc12FatalMessageD1Ev
> __ZN3rtc14MessageHandlerD2Ev
> __ZN3rtc14VideoSinkWantsC1Ev
> __ZN3rtc14VideoSinkWantsD1Ev
>@@ -9,7 +6,6 @@ __ZN3rtc17CopyOnWriteBuffer21CloneDataIfReferencedEm
> __ZN3rtc17CopyOnWriteBufferC1ERKS0_
> __ZN3rtc17CopyOnWriteBufferC1Emm
> __ZN3rtc17CopyOnWriteBufferD1Ev
>-__ZN3rtc6Thread14InvokeInternalERKNS_8LocationEPNS_14MessageHandlerE
> __ZN3rtc6Thread22CreateWithSocketServerEv
> __ZN3rtc6Thread5StartEPNS_8RunnableE
> __ZN3rtc6Thread6CreateEv
>@@ -34,15 +30,11 @@ __ZN6webrtc16ConvertVideoTypeENS_9VideoTypeE
> __ZN6webrtc18CreateIceCandidateERKNSt3__112basic_stringIcNS0_11char_traitsIcEENS0_9allocatorIcEEEEiS8_PNS_13SdpParseErrorE
> __ZN6webrtc19RTCCertificateStats5kTypeE
> __ZN6webrtc19RTCDataChannelStats5kTypeE
>-__ZN6webrtc20CoreVideoFrameBuffer6ToI420Ev
>-__ZN6webrtc20CoreVideoFrameBufferC2EP10__CVBuffer
>-__ZN6webrtc20CoreVideoFrameBufferD2Ev
> __ZN6webrtc24CreateSessionDescriptionERKNSt3__112basic_stringIcNS0_11char_traitsIcEENS0_9allocatorIcEEEES8_PNS_13SdpParseErrorE
> __ZN6webrtc24RTCIceCandidatePairStats5kTypeE
> __ZN6webrtc24RTCInboundRTPStreamStats5kTypeE
> __ZN6webrtc24RTCMediaStreamTrackStats5kTypeE
> __ZN6webrtc25RTCOutboundRTPStreamStats5kTypeE
>-__ZN6webrtc27CreatePeerConnectionFactoryEPN3rtc6ThreadES2_S2_NS0_13scoped_refptrINS_17AudioDeviceModuleEEENS3_INS_19AudioEncoderFactoryEEENS3_INS_19AudioDecoderFactoryEEENSt3__110unique_ptrINS_19VideoEncoderFactoryENSA_14default_deleteISC_EEEENSB_INS_19VideoDecoderFactoryENSD_ISG_EEEENS3_INS_10AudioMixerEEENS3_INS_15AudioProcessingEEE
> __ZN6webrtc27SessionDescriptionInterface6kOfferE
> __ZN6webrtc27SessionDescriptionInterface7kAnswerE
> __ZN6webrtc27SessionDescriptionInterface9kPrAnswerE
>@@ -50,7 +42,6 @@ __ZN6webrtc8internal21SynchronousMethodCall6InvokeERKN3rtc8LocationEPNS2_6Thread
> __ZN6webrtc8internal21SynchronousMethodCallC1EPN3rtc14MessageHandlerE
> __ZN6webrtc8internal21SynchronousMethodCallD1Ev
> __ZN7cricket17MediaTypeToStringENS_9MediaTypeE
>-__ZN7cricket18BasicPortAllocatorC1EPN3rtc14NetworkManagerEPNS1_19PacketSocketFactoryEPN6webrtc14TurnCustomizerEPNS_25RelayPortFactoryInterfaceE
> __ZNK6webrtc10VideoFrame18video_frame_bufferEv
> __ZNK6webrtc10VideoFrame5widthEv
> __ZNK6webrtc10VideoFrame6heightEv
>@@ -59,9 +50,6 @@ __ZNK6webrtc14RTCStatsReport13ConstIteratorneERKS1_
> __ZNK6webrtc14RTCStatsReport13ConstIteratorptEv
> __ZNK6webrtc14RTCStatsReport3endEv
> __ZNK6webrtc14RTCStatsReport5beginEv
>-__ZNK6webrtc20CoreVideoFrameBuffer4typeEv
>-__ZNK6webrtc20CoreVideoFrameBuffer5widthEv
>-__ZNK6webrtc20CoreVideoFrameBuffer6heightEv
> __ZNK6webrtc8RTCStats6ToJsonEv
> __ZTVN3rtc14MessageHandlerE
> _ConvertToI420
>@@ -95,13 +83,11 @@ __ZN3rtc13SocketAddressC1ERKS0_
> __ZN3rtc13SocketAddressaSERKS0_
> __ZNK3rtc13SocketAddress6ipaddrEv
> __ZNK3rtc18NetworkManagerBase11GetNetworksEPNSt3__16vectorIPNS_7NetworkENS1_9allocatorIS4_EEEE
>-__ZN3rtc19PacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
> __ZN3rtc22PacketTimeUpdateParamsC1Ev
> __ZN3rtc17AsyncPacketSocketD2Ev
> __ZN3rtc9ProxyInfoD1Ev
> __ZN3rtc18NetworkManagerBaseC2Ev
> __ZN3rtc7NetworkD1Ev
>-__ZTVN3rtc19PacketSocketFactoryE
> __ZN3rtc22PacketTimeUpdateParamsD1Ev
> __ZN3rtc22AsyncResolverInterfaceC2Ev
> __ZN3rtc18NetworkManagerBase16MergeNetworkListERKNSt3__16vectorIPNS_7NetworkENS1_9allocatorIS4_EEEEPb
>@@ -118,9 +104,6 @@ __ZN6webrtc32CreateBuiltinAudioDecoderFactoryEv
> __ZN6webrtc32CreateBuiltinAudioEncoderFactoryEv
> __ZN6webrtc27SessionDescriptionInterface16RemoveCandidatesERKNSt3__16vectorIN7cricket9CandidateENS1_9allocatorIS4_EEEE
> __ZNK6webrtc21IceCandidateInterface10server_urlEv
>-__ZNK6webrtc27SessionDescriptionInterface7GetTypeEv
>-__ZTVN6webrtc21IceCandidateInterfaceE
>-__ZTVN6webrtc27SessionDescriptionInterfaceE
> __ZN6webrtc20setApplicationStatusEb
> __ZN6webrtc32createVideoToolboxDecoderFactoryEv
> __ZN6webrtc32createVideoToolboxEncoderFactoryEv
>@@ -128,6 +111,98 @@ __ZN6webrtc29setH264HardwareEncoderAllowedEb
> __ZN6webrtc20pixelBufferFromFrameERKNS_10VideoFrameE
> __ZN6webrtc18pixelBufferToFrameEP10__CVBuffer
> __ZN3rtc24BasicPacketSocketFactory19CreateAsyncResolverEv
>-__ZN3rtc24BasicPacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
> __ZN3rtc24BasicPacketSocketFactoryC2Ev
> __ZN3rtc24BasicPacketSocketFactoryD2Ev
>+__ZNK3rtc6Thread9IsCurrentEv
>+__ZN3rtc6Thread14InvokeInternalERKNS_8LocationEPNS_14MessageHandlerE
>+__ZN3rtc5EventD1Ev
>+__ZN3rtc5EventC1Ebb
>+__ZN3rtc5Event4WaitEi
>+__ZN3rtc5Event3SetEv
>+__ZN3rtc18CreateRandomStringEm
>+__ZN6webrtc8RTCError2OKEv
>+__ZN3rtc18webrtc_checks_impl8FatalLogEPKciS2_PKNS0_12CheckArgTypeEz
>+__ZTVN6webrtc32CreateSessionDescriptionObserverE
>+__ZTVN6webrtc30PeerConnectionFactoryInterfaceE
>+__ZTVN6webrtc29SetSessionDescriptionObserverE
>+__ZNK3rtc17ThreadCheckerImpl19CalledOnValidThreadEv
>+__ZN7cricket18BasicPortAllocatorC1EPN3rtc14NetworkManagerEPNS1_19PacketSocketFactoryEPN6webrtc14TurnCustomizerEPNS_25RelayPortFactoryInterfaceE
>+__ZN7cricket12AudioOptionsD1Ev
>+__ZN6webrtc32CreateSessionDescriptionObserver9OnFailureENS_8RTCErrorE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceEPN7cricket13VideoCapturerEPKNS_25MediaConstraintsInterfaceE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceEPN7cricket13VideoCapturerE
>+__ZN6webrtc29SetSessionDescriptionObserver9OnFailureENS_8RTCErrorE
>+__ZN6webrtc28RtpHeaderExtensionCapabilityD1Ev
>+__ZN6webrtc26PeerConnectionDependenciesD1Ev
>+__ZN3rtc24BasicPacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
>+__ZN6webrtc12RtcpFeedbackC1ERKS0_
>+__ZN6webrtc12RtcpFeedbackD1Ev
>+__ZN6webrtc14I420BufferPoolC1Ev
>+__ZN6webrtc14I420BufferPoolD1Ev
>+__ZN6webrtc15RtpCapabilitiesC1Ev
>+__ZN6webrtc15RtpCapabilitiesD1Ev
>+__ZN6webrtc18RtpCodecCapabilityD1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1ERKS1_
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationD1Ev
>+__ZN6webrtc23PeerConnectionInterface9IceServerC1ERKS1_
>+__ZN6webrtc23PeerConnectionInterface9IceServerC1Ev
>+__ZN6webrtc23PeerConnectionInterface9IceServerD1Ev
>+__ZN6webrtc26PeerConnectionDependenciesC1EOS0_
>+__ZN3rtc18webrtc_checks_impl8FatalLogEPKciS2_PKNS0_12CheckArgTypeEz
>+__ZN6webrtc12RtcpFeedbackC1ERKS0_
>+__ZN6webrtc12RtcpFeedbackD1Ev
>+__ZN6webrtc14I420BufferPoolC1Ev
>+__ZN6webrtc14I420BufferPoolD1Ev
>+__ZN6webrtc15RtpCapabilitiesC1Ev
>+__ZN6webrtc15RtpCapabilitiesD1Ev
>+__ZN6webrtc18RtpCodecCapabilityD1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1ERKS1_
>+__ZTVN6webrtc27SessionDescriptionInterfaceE
>+__ZTVN6webrtc23PeerConnectionInterfaceE
>+__ZTVN6webrtc21IceCandidateInterfaceE
>+__ZTVN6webrtc20DataChannelInterfaceE
>+__ZTVN6webrtc19VideoTrackInterfaceE
>+__ZTVN6webrtc19AudioTrackInterfaceE
>+__ZNK6webrtc30PeerConnectionFactoryInterface26GetRtpReceiverCapabilitiesEN7cricket9MediaTypeE
>+__ZNK6webrtc30PeerConnectionFactoryInterface24GetRtpSenderCapabilitiesEN7cricket9MediaTypeE
>+__ZNK6webrtc27SessionDescriptionInterface7GetTypeEv
>+__ZNK6webrtc23PeerConnectionInterface26pending_remote_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface26current_remote_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface25pending_local_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface25current_local_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface15GetTransceiversEv
>+__ZNK6webrtc23PeerConnectionInterface12GetReceiversEv
>+__ZNK6webrtc23PeerConnectionInterface10GetSendersEv
>+__ZNK6webrtc20DataChannelInterface8protocolEv
>+__ZNK6webrtc20DataChannelInterface17maxRetransmitTimeEv
>+__ZNK6webrtc20DataChannelInterface14maxRetransmitsEv
>+__ZNK6webrtc20DataChannelInterface10negotiatedEv
>+__ZNK6webrtc19VideoTrackInterface12content_hintEv
>+__ZN6webrtc30PeerConnectionFactoryInterface20CreatePeerConnectionERKNS_23PeerConnectionInterface16RTCConfigurationENS_26PeerConnectionDependenciesE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceENSt3__110unique_ptrIN7cricket13VideoCapturerENS1_14default_deleteIS4_EEEEPKNS_25MediaConstraintsInterfaceE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceENSt3__110unique_ptrIN7cricket13VideoCapturerENS1_14default_deleteIS4_EEEE
>+__ZN6webrtc23PeerConnectionInterface19RemoveIceCandidatesERKNSt3__16vectorIN7cricket9CandidateENS1_9allocatorIS4_EEEE
>+__ZN6webrtc23PeerConnectionInterface16StartRtcEventLogEix
>+__ZN6webrtc23PeerConnectionInterface16StartRtcEventLogENSt3__110unique_ptrINS_17RtcEventLogOutputENS1_14default_deleteIS3_EEEEx
>+__ZN6webrtc23PeerConnectionInterface16SetConfigurationERKNS0_16RTCConfigurationEPNS_8RTCErrorE
>+__ZN6webrtc23PeerConnectionInterface16SetConfigurationERKNS0_16RTCConfigurationE
>+__ZN6webrtc19AudioTrackInterface14GetSignalLevelEPi
>+__ZN6webrtc19AudioTrackInterface17GetAudioProcessorEv
>+__ZN6webrtc23PeerConnectionInterface10SetBitrateERKNS_15BitrateSettingsE
>+__ZN6webrtc23PeerConnectionInterface12CreateSenderERKNSt3__112basic_stringIcNS1_11char_traitsIcEENS1_9allocatorIcEEEES9_
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN3rtc13scoped_refptrINS_25MediaStreamTrackInterfaceEEE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN3rtc13scoped_refptrINS_25MediaStreamTrackInterfaceEEERKNS_18RtpTransceiverInitE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN7cricket9MediaTypeE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN7cricket9MediaTypeERKNS_18RtpTransceiverInitE
>+__ZN6webrtc23PeerConnectionInterface16GetConfigurationEv
>+__ZN6webrtc23PeerConnectionInterface14RemoveTrackNewEN3rtc13scoped_refptrINS_18RtpSenderInterfaceEEE
>+__ZN3rtc10SentPacketC1Exx
>+__ZN3rtc13PacketOptionsC1ERKS0_
>+__ZN3rtc10PacketInfoD1Ev
>+__ZN3rtc10PacketInfoC1ERKS0_
>+__ZTVN3rtc19PacketSocketFactoryE
>+__ZN3rtc13PacketOptionsD1Ev
>+__ZN3rtc19PacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
>+__ZN3rtc13PacketOptionsC1Ev
>+__ZN6webrtc27CreatePeerConnectionFactoryEPN3rtc6ThreadES2_S2_NS0_13scoped_refptrINS_17AudioDeviceModuleEEENS3_INS_19AudioEncoderFactoryEEENS3_INS_19AudioDecoderFactoryEEENSt3__110unique_ptrINS_19VideoEncoderFactoryENSA_14default_deleteISC_EEEENSB_INS_19VideoDecoderFactoryENSD_ISG_EEEENS3_INS_10AudioMixerEEENS3_INS_15AudioProcessingEEE
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp
>index 89f4e8b8a9c..8478b284152 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp
>@@ -1,7 +1,4 @@
> __ZN3rtc10LogMessage10LogToDebugENS_15LoggingSeverityE
>-__ZN3rtc12FatalMessageC1EPKci
>-__ZN3rtc12FatalMessageC1EPKciPNSt3__112basic_stringIcNS3_11char_traitsIcEENS3_9allocatorIcEEEE
>-__ZN3rtc12FatalMessageD1Ev
> __ZN3rtc14MessageHandlerD2Ev
> __ZN3rtc14VideoSinkWantsC1Ev
> __ZN3rtc14VideoSinkWantsD1Ev
>@@ -9,7 +6,6 @@ __ZN3rtc17CopyOnWriteBuffer21CloneDataIfReferencedEm
> __ZN3rtc17CopyOnWriteBufferC1ERKS0_
> __ZN3rtc17CopyOnWriteBufferC1Emm
> __ZN3rtc17CopyOnWriteBufferD1Ev
>-__ZN3rtc6Thread14InvokeInternalERKNS_8LocationEPNS_14MessageHandlerE
> __ZN3rtc6Thread22CreateWithSocketServerEv
> __ZN3rtc6Thread5StartEPNS_8RunnableE
> __ZN3rtc6Thread6CreateEv
>@@ -34,15 +30,11 @@ __ZN6webrtc16ConvertVideoTypeENS_9VideoTypeE
> __ZN6webrtc18CreateIceCandidateERKNSt3__112basic_stringIcNS0_11char_traitsIcEENS0_9allocatorIcEEEEiS8_PNS_13SdpParseErrorE
> __ZN6webrtc19RTCCertificateStats5kTypeE
> __ZN6webrtc19RTCDataChannelStats5kTypeE
>-__ZN6webrtc20CoreVideoFrameBuffer6ToI420Ev
>-__ZN6webrtc20CoreVideoFrameBufferC2EP10__CVBuffer
>-__ZN6webrtc20CoreVideoFrameBufferD2Ev
> __ZN6webrtc24CreateSessionDescriptionERKNSt3__112basic_stringIcNS0_11char_traitsIcEENS0_9allocatorIcEEEES8_PNS_13SdpParseErrorE
> __ZN6webrtc24RTCIceCandidatePairStats5kTypeE
> __ZN6webrtc24RTCInboundRTPStreamStats5kTypeE
> __ZN6webrtc24RTCMediaStreamTrackStats5kTypeE
> __ZN6webrtc25RTCOutboundRTPStreamStats5kTypeE
>-__ZN6webrtc27CreatePeerConnectionFactoryEPN3rtc6ThreadES2_S2_NS0_13scoped_refptrINS_17AudioDeviceModuleEEENS3_INS_19AudioEncoderFactoryEEENS3_INS_19AudioDecoderFactoryEEENSt3__110unique_ptrINS_19VideoEncoderFactoryENSA_14default_deleteISC_EEEENSB_INS_19VideoDecoderFactoryENSD_ISG_EEEENS3_INS_10AudioMixerEEENS3_INS_15AudioProcessingEEE
> __ZN6webrtc27SessionDescriptionInterface6kOfferE
> __ZN6webrtc27SessionDescriptionInterface7kAnswerE
> __ZN6webrtc27SessionDescriptionInterface9kPrAnswerE
>@@ -50,7 +42,6 @@ __ZN6webrtc8internal21SynchronousMethodCall6InvokeERKN3rtc8LocationEPNS2_6Thread
> __ZN6webrtc8internal21SynchronousMethodCallC1EPN3rtc14MessageHandlerE
> __ZN6webrtc8internal21SynchronousMethodCallD1Ev
> __ZN7cricket17MediaTypeToStringENS_9MediaTypeE
>-__ZN7cricket18BasicPortAllocatorC1EPN3rtc14NetworkManagerEPNS1_19PacketSocketFactoryEPN6webrtc14TurnCustomizerEPNS_25RelayPortFactoryInterfaceE
> __ZNK6webrtc10VideoFrame18video_frame_bufferEv
> __ZNK6webrtc10VideoFrame5widthEv
> __ZNK6webrtc10VideoFrame6heightEv
>@@ -59,9 +50,6 @@ __ZNK6webrtc14RTCStatsReport13ConstIteratorneERKS1_
> __ZNK6webrtc14RTCStatsReport13ConstIteratorptEv
> __ZNK6webrtc14RTCStatsReport3endEv
> __ZNK6webrtc14RTCStatsReport5beginEv
>-__ZNK6webrtc20CoreVideoFrameBuffer4typeEv
>-__ZNK6webrtc20CoreVideoFrameBuffer5widthEv
>-__ZNK6webrtc20CoreVideoFrameBuffer6heightEv
> __ZNK6webrtc8RTCStats6ToJsonEv
> __ZTVN3rtc14MessageHandlerE
> _ConvertToI420
>@@ -95,13 +83,11 @@ __ZN3rtc13SocketAddressC1ERKS0_
> __ZN3rtc13SocketAddressaSERKS0_
> __ZNK3rtc13SocketAddress6ipaddrEv
> __ZNK3rtc18NetworkManagerBase11GetNetworksEPNSt3__16vectorIPNS_7NetworkENS1_9allocatorIS4_EEEE
>-__ZN3rtc19PacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
> __ZN3rtc22PacketTimeUpdateParamsC1Ev
> __ZN3rtc17AsyncPacketSocketD2Ev
> __ZN3rtc9ProxyInfoD1Ev
> __ZN3rtc18NetworkManagerBaseC2Ev
> __ZN3rtc7NetworkD1Ev
>-__ZTVN3rtc19PacketSocketFactoryE
> __ZN3rtc22PacketTimeUpdateParamsD1Ev
> __ZN3rtc22AsyncResolverInterfaceC2Ev
> __ZN3rtc18NetworkManagerBase16MergeNetworkListERKNSt3__16vectorIPNS_7NetworkENS1_9allocatorIS4_EEEEPb
>@@ -118,9 +104,6 @@ __ZN6webrtc32CreateBuiltinAudioDecoderFactoryEv
> __ZN6webrtc32CreateBuiltinAudioEncoderFactoryEv
> __ZN6webrtc27SessionDescriptionInterface16RemoveCandidatesERKNSt3__16vectorIN7cricket9CandidateENS1_9allocatorIS4_EEEE
> __ZNK6webrtc21IceCandidateInterface10server_urlEv
>-__ZNK6webrtc27SessionDescriptionInterface7GetTypeEv
>-__ZTVN6webrtc21IceCandidateInterfaceE
>-__ZTVN6webrtc27SessionDescriptionInterfaceE
> __ZN6webrtc20setApplicationStatusEb
> __ZN6webrtc32createVideoToolboxDecoderFactoryEv
> __ZN6webrtc32createVideoToolboxEncoderFactoryEv
>@@ -128,6 +111,98 @@ __ZN6webrtc29setH264HardwareEncoderAllowedEb
> __ZN6webrtc20pixelBufferFromFrameERKNS_10VideoFrameE
> __ZN6webrtc18pixelBufferToFrameEP10__CVBuffer
> __ZN3rtc24BasicPacketSocketFactory19CreateAsyncResolverEv
>-__ZN3rtc24BasicPacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
> __ZN3rtc24BasicPacketSocketFactoryC2Ev
> __ZN3rtc24BasicPacketSocketFactoryD2Ev
>+__ZNK3rtc6Thread9IsCurrentEv
>+__ZN3rtc6Thread14InvokeInternalERKNS_8LocationEPNS_14MessageHandlerE
>+__ZN3rtc5EventD1Ev
>+__ZN3rtc5EventC1Ebb
>+__ZN3rtc5Event4WaitEi
>+__ZN3rtc5Event3SetEv
>+__ZN3rtc18CreateRandomStringEm
>+__ZN6webrtc8RTCError2OKEv
>+__ZN3rtc18webrtc_checks_impl8FatalLogEPKciS2_PKNS0_12CheckArgTypeEz
>+__ZTVN6webrtc32CreateSessionDescriptionObserverE
>+__ZTVN6webrtc30PeerConnectionFactoryInterfaceE
>+__ZTVN6webrtc29SetSessionDescriptionObserverE
>+__ZNK3rtc17ThreadCheckerImpl19CalledOnValidThreadEv
>+__ZN7cricket18BasicPortAllocatorC1EPN3rtc14NetworkManagerEPNS1_19PacketSocketFactoryEPN6webrtc14TurnCustomizerEPNS_25RelayPortFactoryInterfaceE
>+__ZN7cricket12AudioOptionsD1Ev
>+__ZN6webrtc32CreateSessionDescriptionObserver9OnFailureENS_8RTCErrorE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceEPN7cricket13VideoCapturerEPKNS_25MediaConstraintsInterfaceE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceEPN7cricket13VideoCapturerE
>+__ZN6webrtc29SetSessionDescriptionObserver9OnFailureENS_8RTCErrorE
>+__ZN6webrtc28RtpHeaderExtensionCapabilityD1Ev
>+__ZN6webrtc26PeerConnectionDependenciesD1Ev
>+__ZN3rtc24BasicPacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
>+__ZN6webrtc12RtcpFeedbackC1ERKS0_
>+__ZN6webrtc12RtcpFeedbackD1Ev
>+__ZN6webrtc14I420BufferPoolC1Ev
>+__ZN6webrtc14I420BufferPoolD1Ev
>+__ZN6webrtc15RtpCapabilitiesC1Ev
>+__ZN6webrtc15RtpCapabilitiesD1Ev
>+__ZN6webrtc18RtpCodecCapabilityD1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1ERKS1_
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationD1Ev
>+__ZN6webrtc23PeerConnectionInterface9IceServerC1ERKS1_
>+__ZN6webrtc23PeerConnectionInterface9IceServerC1Ev
>+__ZN6webrtc23PeerConnectionInterface9IceServerD1Ev
>+__ZN6webrtc26PeerConnectionDependenciesC1EOS0_
>+__ZN3rtc18webrtc_checks_impl8FatalLogEPKciS2_PKNS0_12CheckArgTypeEz
>+__ZN6webrtc12RtcpFeedbackC1ERKS0_
>+__ZN6webrtc12RtcpFeedbackD1Ev
>+__ZN6webrtc14I420BufferPoolC1Ev
>+__ZN6webrtc14I420BufferPoolD1Ev
>+__ZN6webrtc15RtpCapabilitiesC1Ev
>+__ZN6webrtc15RtpCapabilitiesD1Ev
>+__ZN6webrtc18RtpCodecCapabilityD1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1ERKS1_
>+__ZTVN6webrtc27SessionDescriptionInterfaceE
>+__ZTVN6webrtc23PeerConnectionInterfaceE
>+__ZTVN6webrtc21IceCandidateInterfaceE
>+__ZTVN6webrtc20DataChannelInterfaceE
>+__ZTVN6webrtc19VideoTrackInterfaceE
>+__ZTVN6webrtc19AudioTrackInterfaceE
>+__ZNK6webrtc30PeerConnectionFactoryInterface26GetRtpReceiverCapabilitiesEN7cricket9MediaTypeE
>+__ZNK6webrtc30PeerConnectionFactoryInterface24GetRtpSenderCapabilitiesEN7cricket9MediaTypeE
>+__ZNK6webrtc27SessionDescriptionInterface7GetTypeEv
>+__ZNK6webrtc23PeerConnectionInterface26pending_remote_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface26current_remote_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface25pending_local_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface25current_local_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface15GetTransceiversEv
>+__ZNK6webrtc23PeerConnectionInterface12GetReceiversEv
>+__ZNK6webrtc23PeerConnectionInterface10GetSendersEv
>+__ZNK6webrtc20DataChannelInterface8protocolEv
>+__ZNK6webrtc20DataChannelInterface17maxRetransmitTimeEv
>+__ZNK6webrtc20DataChannelInterface14maxRetransmitsEv
>+__ZNK6webrtc20DataChannelInterface10negotiatedEv
>+__ZNK6webrtc19VideoTrackInterface12content_hintEv
>+__ZN6webrtc30PeerConnectionFactoryInterface20CreatePeerConnectionERKNS_23PeerConnectionInterface16RTCConfigurationENS_26PeerConnectionDependenciesE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceENSt3__110unique_ptrIN7cricket13VideoCapturerENS1_14default_deleteIS4_EEEEPKNS_25MediaConstraintsInterfaceE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceENSt3__110unique_ptrIN7cricket13VideoCapturerENS1_14default_deleteIS4_EEEE
>+__ZN6webrtc23PeerConnectionInterface19RemoveIceCandidatesERKNSt3__16vectorIN7cricket9CandidateENS1_9allocatorIS4_EEEE
>+__ZN6webrtc23PeerConnectionInterface16StartRtcEventLogEix
>+__ZN6webrtc23PeerConnectionInterface16StartRtcEventLogENSt3__110unique_ptrINS_17RtcEventLogOutputENS1_14default_deleteIS3_EEEEx
>+__ZN6webrtc23PeerConnectionInterface16SetConfigurationERKNS0_16RTCConfigurationEPNS_8RTCErrorE
>+__ZN6webrtc23PeerConnectionInterface16SetConfigurationERKNS0_16RTCConfigurationE
>+__ZN6webrtc19AudioTrackInterface14GetSignalLevelEPi
>+__ZN6webrtc19AudioTrackInterface17GetAudioProcessorEv
>+__ZN6webrtc23PeerConnectionInterface10SetBitrateERKNS_15BitrateSettingsE
>+__ZN6webrtc23PeerConnectionInterface12CreateSenderERKNSt3__112basic_stringIcNS1_11char_traitsIcEENS1_9allocatorIcEEEES9_
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN3rtc13scoped_refptrINS_25MediaStreamTrackInterfaceEEE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN3rtc13scoped_refptrINS_25MediaStreamTrackInterfaceEEERKNS_18RtpTransceiverInitE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN7cricket9MediaTypeE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN7cricket9MediaTypeERKNS_18RtpTransceiverInitE
>+__ZN6webrtc23PeerConnectionInterface16GetConfigurationEv
>+__ZN6webrtc23PeerConnectionInterface14RemoveTrackNewEN3rtc13scoped_refptrINS_18RtpSenderInterfaceEEE
>+__ZN3rtc10SentPacketC1Exx
>+__ZN3rtc13PacketOptionsC1ERKS0_
>+__ZN3rtc10PacketInfoD1Ev
>+__ZN3rtc10PacketInfoC1ERKS0_
>+__ZTVN3rtc19PacketSocketFactoryE
>+__ZN3rtc13PacketOptionsD1Ev
>+__ZN3rtc19PacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
>+__ZN3rtc13PacketOptionsC1Ev
>+__ZN6webrtc27CreatePeerConnectionFactoryEPN3rtc6ThreadES2_S2_NS0_13scoped_refptrINS_17AudioDeviceModuleEEENS3_INS_19AudioEncoderFactoryEEENS3_INS_19AudioDecoderFactoryEEENSt3__110unique_ptrINS_19VideoEncoderFactoryENSA_14default_deleteISC_EEEENSB_INS_19VideoDecoderFactoryENSD_ISG_EEEENS3_INS_10AudioMixerEEENS3_INS_15AudioProcessingEEE
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig
>index 0306586af6b..62a1196e337 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig
>@@ -26,7 +26,7 @@ INSTALLHDRS_SCRIPT_PHASE = YES;
> WARNING_CFLAGS = -Wno-deprecated-declarations $(inherited);
> 
> // FIXME: Set WEBRTC_USE_BUILTIN_ISAC_FIX and WEBRTC_USE_BUILTIN_ISAC_FLOAT for iOS and Mac
>-GCC_PREPROCESSOR_DEFINITIONS = GTEST_RELATIVE_PATH WEBRTC_OPUS_SUPPORT_120MS_PTIME=0 WEBRTC_POSIX WEBRTC_MAC SSL_USE_OPENSSL FEATURE_ENABLE_SSL HAVE_SRTP HAVE_OPENSSL_SSL_H SCTP_PROCESS_LEVEL_LOCKS SCTP_SIMPLE_ALLOCATOR SCTP_USE_OPENSSL_SHA1 __Userspace__ HAVE_SA_LEN HAVE_SCONN_LEN __APPLE_USE_RFC_2292 __Userspace_os_Darwin NON_WINDOWS_DEFINE HAVE_WEBRTC_VIDEO HAVE_WEBRTC_VOICE WEBRTC_INTELLIGIBILITY_ENHANCER=0 WEBRTC_APM_DEBUG_DUMP=0 WEBRTC_NS_FLOAT WEBRTC_USE_BUILTIN_ILBC WEBRTC_CODEC_ILBC WEBRTC_USE_BUILTIN_OPUS WEBRTC_CODEC_OPUS WEBRTC_CODEC_ISAC WEBRTC_CODEC_RED RTC_DISABLE_VP9 RTC_DISABLE_VP8 HAVE_STDINT_H HAVE_STDLIB_H HAVE_UINT64_T OPENSSL HAVE_CONFIG_H WEBRTC_WEBKIT_BUILD HAVE_PTHREAD_COND_TIMEDWAIT_RELATIVE  HAVE_SCTP WEBRTC_CODEC_G711 WEBRTC_CODEC_G722 WEBRTC_OPUS_VARIABLE_COMPLEXITY=0 WEBRTC_USE_BUILTIN_ISAC_FIX=1 WEBRTC_USE_BUILTIN_ISAC_FLOAT=0 $(inherited);
>+GCC_PREPROCESSOR_DEFINITIONS = GTEST_RELATIVE_PATH WEBRTC_OPUS_SUPPORT_120MS_PTIME=0 WEBRTC_POSIX WEBRTC_MAC SSL_USE_OPENSSL FEATURE_ENABLE_SSL HAVE_SRTP HAVE_OPENSSL_SSL_H SCTP_PROCESS_LEVEL_LOCKS SCTP_SIMPLE_ALLOCATOR SCTP_USE_OPENSSL_SHA1 __Userspace__ HAVE_SA_LEN HAVE_SCONN_LEN __APPLE_USE_RFC_2292 __Userspace_os_Darwin NON_WINDOWS_DEFINE HAVE_WEBRTC_VIDEO HAVE_WEBRTC_VOICE WEBRTC_INTELLIGIBILITY_ENHANCER=0 WEBRTC_APM_DEBUG_DUMP=0 WEBRTC_NS_FLOAT WEBRTC_USE_BUILTIN_ILBC WEBRTC_CODEC_ILBC WEBRTC_USE_BUILTIN_OPUS WEBRTC_CODEC_OPUS WEBRTC_CODEC_ISAC WEBRTC_CODEC_RED RTC_DISABLE_VP9 RTC_DISABLE_VP8 HAVE_STDINT_H HAVE_STDLIB_H HAVE_UINT64_T OPENSSL HAVE_CONFIG_H WEBRTC_WEBKIT_BUILD HAVE_PTHREAD_COND_TIMEDWAIT_RELATIVE  HAVE_SCTP WEBRTC_CODEC_G711 WEBRTC_CODEC_G722 WEBRTC_OPUS_VARIABLE_COMPLEXITY=0 WEBRTC_USE_BUILTIN_ISAC_FIX=1 WEBRTC_USE_BUILTIN_ISAC_FLOAT=0 USE_BUILTIN_SW_CODECS $(inherited);
> 
> GCC_PREPROCESSOR_DEFINITIONS[sdk=macosx*] = $(inherited) WEBRTC_USE_VTB_HARDWARE_ENCODER;
> GCC_PREPROCESSOR_DEFINITIONS[sdk=iphoneos*] = $(inherited) WEBRTC_IOS;
>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 d3b1a7c5cb1..cbed3334d77 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
>@@ -16,7 +16,14 @@
> #include <unistd.h>
> #include <list>
> 
>+#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/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec.mm
>index d7521262d4d..955d9745863 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec.mm
>@@ -43,6 +43,7 @@ namespace {
> using namespace webrtc::H264;
> 
> NSString *MaxSupportedLevelForProfile(Profile profile) {
>+#if !defined(WEBRTC_WEBKIT_BUILD)
>   const absl::optional<ProfileLevelId> profileLevelId = [UIDevice maxSupportedH264Profile];
>   if (profileLevelId && profileLevelId->profile >= profile) {
>     const absl::optional<std::string> profileString =
>@@ -51,6 +52,7 @@ NSString *MaxSupportedLevelForProfile(Profile profile) {
>       return [NSString stringForStdString:*profileString];
>     }
>   }
>+#endif
>   return nil;
> }
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm
>index 2871b8a5ef1..4c9576e150d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm
>@@ -260,9 +260,13 @@ - (void)configureDecompressionSession {
> - (void)destroyDecompressionSession {
>   if (_decompressionSession) {
> #if defined(WEBRTC_IOS)
>+#if !defined(WEBRTC_WEBKIT_BUILD)
>     if ([UIDevice isIOS11OrLater]) {
>       VTDecompressionSessionWaitForAsynchronousFrames(_decompressionSession);
>     }
>+#else
>+    VTDecompressionSessionWaitForAsynchronousFrames(_decompressionSession);
>+#endif
> #endif
>     VTDecompressionSessionInvalidate(_decompressionSession);
>     CFRelease(_decompressionSession);
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0001-Adapting-libwebrtc-H264-codec.patch b/Source/ThirdParty/libwebrtc/WebKit/0001-Adapting-libwebrtc-H264-codec.patch
>deleted file mode 100644
>index 03638614812..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0001-Adapting-libwebrtc-H264-codec.patch
>+++ /dev/null
>@@ -1,118 +0,0 @@
>-From c5debd83bb3933413d0a3dd2417f5fd9d64a3205 Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Thu, 22 Mar 2018 16:38:14 -0700
>-Subject: [PATCH] Adapting-libwebrtc-H264-codec
>-
>----
>- .../sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm      |  1 +
>- .../Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm     | 11 +++++++----
>- .../Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm     | 10 ++++++++--
>- 3 files changed, 16 insertions(+), 6 deletions(-)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm
>-index 203896758e8..537a65e17d3 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm
>-@@ -73,6 +73,7 @@ - (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
>- 
>- - (void)dealloc {
>-   CVBufferRelease(_pixelBuffer);
>-+  [super dealloc];
>- }
>- 
>- - (int)width {
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm
>-index 2f4b6bd72c6..34d08db0e3b 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm
>-@@ -90,6 +90,10 @@ - (instancetype)init {
>- - (void)dealloc {
>-   [self destroyDecompressionSession];
>-   [self setVideoFormat:nullptr];
>-+  if (_callback) {
>-+    Block_release(_callback);
>-+  }
>-+  [super dealloc];
>- }
>- 
>- - (NSInteger)startDecodeWithSettings:(RTCVideoEncoderSettings *)settings
>-@@ -178,7 +182,7 @@ - (NSInteger)decode:(RTCEncodedImage *)inputImage
>- }
>- 
>- - (void)setCallback:(RTCVideoDecoderCallback)callback {
>--  _callback = callback;
>-+  _callback = Block_copy(callback);
>- }
>- 
>- - (void)setError:(OSStatus)error {
>-@@ -190,6 +194,7 @@ - (NSInteger)releaseDecoder {
>-   // is safe to null out the callback.
>-   [self destroyDecompressionSession];
>-   [self setVideoFormat:nullptr];
>-+  Block_release(_callback);
>-   _callback = nullptr;
>-   return WEBRTC_VIDEO_CODEC_OK;
>- }
>-@@ -260,9 +265,7 @@ - (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/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-index a818c27d1e6..809948011a6 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-@@ -34,6 +34,7 @@
>- #include "sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h"
>- #include "system_wrappers/include/clock.h"
>- #include "third_party/libyuv/include/libyuv/convert_from.h"
>-+#include "sdk/WebKit/WebKitUtilities.h"
>- 
>- @interface RTCVideoEncoderH264 ()
>- 
>-@@ -318,6 +319,10 @@ - (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo {
>- 
>- - (void)dealloc {
>-   [self destroyCompressionSession];
>-+  if (_callback) {
>-+    Block_release(_callback);
>-+  }
>-+  [super dealloc];
>- }
>- 
>- - (NSInteger)startEncodeWithSettings:(RTCVideoEncoderSettings *)settings
>-@@ -463,7 +468,7 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>- }
>- 
>- - (void)setCallback:(RTCVideoEncoderCallback)callback {
>--  _callback = callback;
>-+  _callback = Block_copy(callback);
>- }
>- 
>- - (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate {
>-@@ -480,6 +485,7 @@ - (NSInteger)releaseEncoder {
>-   // callback anymore. Do not remove callback until the session is invalidated
>-   // since async encoder callbacks can occur until invalidation.
>-   [self destroyCompressionSession];
>-+  Block_release(_callback);
>-   _callback = nullptr;
>-   return WEBRTC_VIDEO_CODEC_OK;
>- }
>-@@ -573,7 +579,7 @@ - (int)resetCompressionSessionWithPixelFormat:(OSType)framePixelFormat {
>-       nullptr, 1, &kCFTypeDictionaryKeyCallBacks, &kCFTypeDictionaryValueCallBacks);
>-   CFDictionarySetValue(encoder_specs,
>-                        kVTVideoEncoderSpecification_EnableHardwareAcceleratedVideoEncoder,
>--                       kCFBooleanTrue);
>-+                       webrtc::isH264HardwareEncoderAllowed() ? kCFBooleanTrue : kCFBooleanFalse);
>- #endif
>-   OSStatus status =
>-       VTCompressionSessionCreate(nullptr,  // use default allocator
>--- 
>-2.15.1 (Apple Git-101)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0001-Disable-SIGPIPE-for-WebRTC-sockets.patch b/Source/ThirdParty/libwebrtc/WebKit/0001-Disable-SIGPIPE-for-WebRTC-sockets.patch
>deleted file mode 100644
>index 3a12db8e198..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0001-Disable-SIGPIPE-for-WebRTC-sockets.patch
>+++ /dev/null
>@@ -1,25 +0,0 @@
>-From ca7c3a4683bbafaaf8a7bbe9fcf7b99cd0a55441 Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Mon, 9 Apr 2018 13:27:57 -0700
>-Subject: [PATCH] Disable SIGPIPE for WebRTC sockets
>-
>----
>- .../ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc | 2 +-
>- 1 file changed, 1 insertion(+), 1 deletion(-)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc
>-index 0d5fc525242..bba921b6692 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc
>-@@ -632,7 +632,7 @@ 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
>--- 
>-2.15.1 (Apple Git-101)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0001-Update-RTCVideoEncoderH264.mm-for-WebKit.patch b/Source/ThirdParty/libwebrtc/WebKit/0001-Update-RTCVideoEncoderH264.mm-for-WebKit.patch
>deleted file mode 100644
>index 168be45c14f..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0001-Update-RTCVideoEncoderH264.mm-for-WebKit.patch
>+++ /dev/null
>@@ -1,274 +0,0 @@
>-From 6526d1daf054748cd0983f994ec0f988df235dc1 Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Mon, 30 Apr 2018 11:26:26 -0700
>-Subject: [PATCH] Update RTCVideoEncoderH264 for WebKit
>-
>----
>- .../Classes/VideoToolbox/RTCVideoEncoderH264.mm    | 159 ++++++++++++++++++---
>- 1 file changed, 140 insertions(+), 19 deletions(-)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-index a818c27d1e6..68118f8f56a 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-@@ -35,6 +35,43 @@
>- #include "system_wrappers/include/clock.h"
>- #include "third_party/libyuv/include/libyuv/convert_from.h"
>- 
>-+#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
>-+
>- @interface RTCVideoEncoderH264 ()
>- 
>- - (void)frameWasEncoded:(OSStatus)status
>-@@ -285,7 +322,7 @@ @implementation RTCVideoEncoderH264 {
>-   RTCVideoEncoderCallback _callback;
>-   int32_t _width;
>-   int32_t _height;
>--  VTCompressionSessionRef _compressionSession;
>-+  CompressionSessionRef _compressionSession;
>-   RTCVideoCodecMode _mode;
>- 
>-   webrtc::H264BitstreamParser _h264BitstreamParser;
>-@@ -318,6 +355,10 @@ - (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo {
>- 
>- - (void)dealloc {
>-   [self destroyCompressionSession];
>-+  if (_callback) {
>-+    Block_release(_callback);
>-+  }
>-+  [super dealloc];
>- }
>- 
>- - (NSInteger)startEncodeWithSettings:(RTCVideoEncoderSettings *)settings
>-@@ -358,9 +399,9 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>- 
>-   // Get a pixel buffer from the pool and copy frame data over.
>-   CVPixelBufferPoolRef pixelBufferPool =
>--      VTCompressionSessionGetPixelBufferPool(_compressionSession);
>-+      CompressionSessionGetPixelBufferPool(_compressionSession);
>-   if ([self resetCompressionSessionIfNeededForPool:pixelBufferPool withFrame:frame]) {
>--    pixelBufferPool = VTCompressionSessionGetPixelBufferPool(_compressionSession);
>-+    pixelBufferPool = CompressionSessionGetPixelBufferPool(_compressionSession);
>-     isKeyframeRequired = YES;
>-   }
>- 
>-@@ -442,7 +483,7 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>-   // Update the bitrate if needed.
>-   [self setBitrateBps:_bitrateAdjuster->GetAdjustedBitrateBps()];
>- 
>--  OSStatus status = VTCompressionSessionEncodeFrame(_compressionSession,
>-+  OSStatus status = CompressionSessionEncodeFrame(_compressionSession,
>-                                                     pixelBuffer,
>-                                                     presentationTimeStamp,
>-                                                     kCMTimeInvalid,
>-@@ -463,7 +504,7 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>- }
>- 
>- - (void)setCallback:(RTCVideoEncoderCallback)callback {
>--  _callback = callback;
>-+  _callback = Block_copy(callback);
>- }
>- 
>- - (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate {
>-@@ -480,6 +521,7 @@ - (NSInteger)releaseEncoder {
>-   // callback anymore. Do not remove callback until the session is invalidated
>-   // since async encoder callbacks can occur until invalidation.
>-   [self destroyCompressionSession];
>-+  Block_release(_callback);
>-   _callback = nullptr;
>-   return WEBRTC_VIDEO_CODEC_OK;
>- }
>-@@ -565,22 +607,21 @@ - (int)resetCompressionSessionWithPixelFormat:(OSType)framePixelFormat {
>-     CFRelease(pixelFormat);
>-     pixelFormat = nullptr;
>-   }
>--  CFMutableDictionaryRef encoder_specs = nullptr;
>-+  CFDictionaryRef encoderSpecs = nullptr;
>- #if defined(WEBRTC_MAC) && !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);
>- #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,
>-@@ -590,24 +631,101 @@ - (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) << "RTCVideoEncoderH264 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, 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];
>-@@ -619,6 +737,9 @@ - (void)configureCompressionSession {
>-   SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_RealTime, true);
>-   SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_ProfileLevel, _profile);
>-   SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_AllowFrameReordering, false);
>-+#if ENABLE_VCP_ENCODER
>-+  SetVTSessionProperty(_compressionSession, (__bridge CFStringRef)getkVTVideoEncoderSpecification_Usage(), 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.
>-@@ -636,7 +757,7 @@ - (void)configureCompressionSession {
>- 
>- - (void)destroyCompressionSession {
>-   if (_compressionSession) {
>--    VTCompressionSessionInvalidate(_compressionSession);
>-+    CompressionSessionInvalidate(_compressionSession);
>-     CFRelease(_compressionSession);
>-     _compressionSession = nullptr;
>-   }
>--- 
>-2.16.1 (Apple Git-102)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0001-Updating-webrtc.patch b/Source/ThirdParty/libwebrtc/WebKit/0001-Updating-webrtc.patch
>new file mode 100644
>index 00000000000..db29fe8a9df
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/WebKit/0001-Updating-webrtc.patch
>@@ -0,0 +1,1769 @@
>+From 97d8ac39e6fc0e8dc1b7e25a1210d6aed91fe452 Mon Sep 17 00:00:00 2001
>+From: Youenn Fablet <youenn@apple.com>
>+Date: Tue, 21 Aug 2018 14:23:22 -0700
>+Subject: [PATCH 1/2] Updating webrtc
>+
>+---
>+ .../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 +-
>+ .../Source/webrtc/media/base/codec.cc         |   4 +
>+ .../audio_coding/acm2/audio_coding_module.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 +-
>+ .../video_coding/codecs/vp8/include/vp8.h     |   2 +
>+ .../webrtc/modules/video_coding/qp_parser.cc  |   2 +
>+ .../video_coding/video_codec_initializer.cc   |   4 +
>+ .../Source/webrtc/pc/peerconnectionfactory.cc |   4 +-
>+ .../webrtc/pc/rtpparametersconversion.cc      |   2 +-
>+ .../libwebrtc/Source/webrtc/rtc_base/checks.h |   4 +-
>+ .../libwebrtc/Source/webrtc/rtc_base/flags.cc |   6 +-
>+ .../Source/webrtc/rtc_base/location.h         |   2 +-
>+ .../webrtc/rtc_base/logging_unittest.cc       |   2 +-
>+ .../rtc_base/numerics/safe_conversions.h      |   4 +-
>+ .../webrtc/rtc_base/opensslcertificate.cc     |   8 +-
>+ .../webrtc/rtc_base/physicalsocketserver.cc   |   2 +-
>+ .../Source/webrtc/rtc_base/stringize_macros.h |   4 +-
>+ .../Source/webrtc/rtc_base/task_queue_gcd.cc  |   5 +-
>+ .../sdk/WebKit/VideoProcessingSoftLink.h      |   2 +-
>+ .../sdk/WebKit/VideoToolBoxDecoderFactory.cpp |   2 +-
>+ .../sdk/WebKit/VideoToolBoxEncoderFactory.cpp |   3 +-
>+ .../webrtc/sdk/WebKit/WebKitUtilities.h       |   3 +-
>+ .../webrtc/sdk/WebKit/WebKitUtilities.mm      |  34 ++-
>+ .../Source/webrtc/sdk/WebKit/decoder.h        |   1 -
>+ .../Source/webrtc/sdk/WebKit/decoder.mm       |   1 -
>+ .../Common/RTCUIApplicationStatusObserver.h   |   1 +
>+ .../VideoToolbox/RTCVideoEncoderH264.mm       | 235 +++++++++++++-----
>+ .../Headers/WebRTC/RTCAudioSession.h          |   4 +
>+ .../WebRTC/RTCAudioSessionConfiguration.h     |   1 +
>+ .../Framework/Headers/WebRTC/RTCAudioSource.h |   1 +
>+ .../Framework/Headers/WebRTC/RTCAudioTrack.h  |   1 +
>+ .../Headers/WebRTC/RTCCameraPreviewView.h     |   1 +
>+ .../Headers/WebRTC/RTCConfiguration.h         |   1 +
>+ .../Framework/Headers/WebRTC/RTCDataChannel.h |   2 +
>+ .../WebRTC/RTCDataChannelConfiguration.h      |   1 +
>+ .../Framework/Headers/WebRTC/RTCDispatcher.h  |   1 +
>+ .../Framework/Headers/WebRTC/RTCFileLogger.h  |   1 +
>+ .../Headers/WebRTC/RTCIceCandidate.h          |   1 +
>+ .../Framework/Headers/WebRTC/RTCIceServer.h   |   1 +
>+ .../Headers/WebRTC/RTCIntervalRange.h         |   1 +
>+ .../Headers/WebRTC/RTCLegacyStatsReport.h     |   1 +
>+ .../Headers/WebRTC/RTCMediaConstraints.h      |   1 +
>+ .../Framework/Headers/WebRTC/RTCMediaSource.h |   1 +
>+ .../Framework/Headers/WebRTC/RTCMediaStream.h |   1 +
>+ .../Headers/WebRTC/RTCMediaStreamTrack.h      |   1 +
>+ .../Headers/WebRTC/RTCMetricsSampleInfo.h     |   1 +
>+ .../Headers/WebRTC/RTCPeerConnection.h        |   4 +
>+ .../Headers/WebRTC/RTCPeerConnectionFactory.h |   1 +
>+ .../Headers/WebRTC/RTCRtpCodecParameters.h    |   1 +
>+ .../Headers/WebRTC/RTCRtpEncodingParameters.h |   1 +
>+ .../Headers/WebRTC/RTCRtpParameters.h         |   1 +
>+ .../Framework/Headers/WebRTC/RTCRtpReceiver.h |   1 +
>+ .../Framework/Headers/WebRTC/RTCRtpSender.h   |   1 +
>+ .../Headers/WebRTC/RTCSessionDescription.h    |   1 +
>+ .../Headers/WebRTC/RTCVideoCapturer.h         |   1 +
>+ .../Framework/Headers/WebRTC/RTCVideoCodec.h  |   5 +
>+ .../Headers/WebRTC/RTCVideoCodecFactory.h     |   2 +
>+ .../Headers/WebRTC/RTCVideoCodecH264.h        |   6 +
>+ .../Headers/WebRTC/RTCVideoDecoderVP8.h       |   1 +
>+ .../Headers/WebRTC/RTCVideoDecoderVP9.h       |   1 +
>+ .../Headers/WebRTC/RTCVideoEncoderVP8.h       |   1 +
>+ .../Headers/WebRTC/RTCVideoEncoderVP9.h       |   1 +
>+ .../Framework/Headers/WebRTC/RTCVideoFrame.h  |   1 +
>+ .../Headers/WebRTC/RTCVideoFrameBuffer.h      |   3 +
>+ .../Framework/Headers/WebRTC/RTCVideoSource.h |   1 +
>+ .../Framework/Headers/WebRTC/RTCVideoTrack.h  |   1 +
>+ 73 files changed, 307 insertions(+), 118 deletions(-)
>+
>+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/media/base/codec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.cc
>+index caf32121125..ee7eda343b3 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.cc
>+@@ -271,8 +271,10 @@ bool VideoCodec::Matches(const VideoCodec& other) const {
>+   if (CodecNamesEq(name.c_str(), kH264CodecName))
>+     return webrtc::H264::IsSameH264Profile(params, other.params) &&
>+            IsSameH264PacketizationMode(params, other.params);
>++#if !defined(RTC_DISABLE_VP9)
>+   if (CodecNamesEq(name.c_str(), kVp9CodecName))
>+     return webrtc::IsSameVP9Profile(params, other.params);
>++#endif
>+   return true;
>+ }
>+ 
>+@@ -392,8 +394,10 @@ bool IsSameCodec(const std::string& name1,
>+   // For every format besides H264 and VP9, comparing names is enough.
>+   if (CodecNamesEq(name1.c_str(), kH264CodecName))
>+     return webrtc::H264::IsSameH264Profile(params1, params2);
>++#if !defined(RTC_DISABLE_VP9)
>+   if (CodecNamesEq(name1.c_str(), kVp9CodecName))
>+     return webrtc::IsSameVP9Profile(params1, params2);
>++#endif
>+   return true;
>+ }
>+ 
>+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 7f652a215d6..ef4250eb40e 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
>+@@ -1149,7 +1149,7 @@ int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
>+       app = AudioEncoder::Application::kAudio;
>+       break;
>+     default:
>+-      FATAL();
>++      RTC_FATAL();
>+       return 0;
>+   }
>+   return encoder_stack_->SetApplication(app) ? 0 : -1;
>+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 818e17f3b57..7eff320b723 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
>+@@ -215,7 +215,7 @@ 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)));
>+ }
>+@@ -230,7 +230,7 @@ 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/modules/video_coding/codecs/vp8/include/vp8.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
>+index 57be521320d..580755f08de 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
>+@@ -22,6 +22,7 @@ namespace webrtc {
>+ class VP8Encoder : public VideoEncoder {
>+  public:
>+   static std::unique_ptr<VP8Encoder> Create();
>++  static bool IsSupported();
>+ 
>+   ~VP8Encoder() override {}
>+ };  // end of VP8Encoder class
>+@@ -29,6 +30,7 @@ class VP8Encoder : public VideoEncoder {
>+ class VP8Decoder : public VideoDecoder {
>+  public:
>+   static std::unique_ptr<VP8Decoder> Create();
>++  static bool IsSupported();
>+ 
>+   ~VP8Decoder() override {}
>+ };  // end of VP8Decoder class
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc
>+index 2adbca9b712..2db0d9cbf53 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc
>+@@ -21,9 +21,11 @@ bool QpParser::GetQp(const VCMEncodedFrame& frame, int* qp) {
>+     case kVideoCodecVP8:
>+       // QP range: [0, 127].
>+       return vp8::GetQp(frame.Buffer(), frame.Length(), qp);
>++#if !defined(RTC_DISABLE_VP9)
>+     case kVideoCodecVP9:
>+       // QP range: [0, 255].
>+       return vp9::GetQp(frame.Buffer(), frame.Length(), qp);
>++#endif
>+     default:
>+       return false;
>+   }
>+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 fe59afc184c..cd73551891a 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
>+@@ -56,9 +56,11 @@ VideoCodecInitializer::CreateBitrateAllocator(const VideoCodec& codec) {
>+     case kVideoCodecH264:
>+       rate_allocator.reset(new SimulcastRateAllocator(codec));
>+       break;
>++#if !defined(RTC_DISABLE_VP9)
>+     case kVideoCodecVP9:
>+       rate_allocator.reset(new SvcRateAllocator(codec));
>+       break;
>++#endif
>+     default:
>+       rate_allocator.reset(new DefaultVideoBitrateAllocator(codec));
>+   }
>+@@ -177,6 +179,7 @@ VideoCodec VideoCodecInitializer::VideoEncoderConfigToVideoCodec(
>+ 
>+       break;
>+     }
>++#if !defined(RTC_DISABLE_VP9)
>+     case kVideoCodecVP9: {
>+       if (!config.encoder_specific_settings) {
>+         *video_codec.VP9() = VideoEncoder::GetDefaultVp9Settings();
>+@@ -232,6 +235,7 @@ VideoCodec VideoCodecInitializer::VideoEncoderConfigToVideoCodec(
>+ 
>+       break;
>+     }
>++#endif
>+     case kVideoCodecH264: {
>+       if (!config.encoder_specific_settings)
>+         *video_codec.H264() = VideoEncoder::GetDefaultH264Settings();
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc
>+index d407228e67c..a6db5198692 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 a65088d3fda..c7cf36e4b4d 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h
>+@@ -81,7 +81,7 @@ 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.
>+@@ -349,7 +349,7 @@ 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::webrtc_checks_impl::LogStreamer<>()
>+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_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_unittest.cc
>+index af512129fbb..7c411926440 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_unittest.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_unittest.cc
>+@@ -196,7 +196,7 @@ 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/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/physicalsocketserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc
>+index ca78499179a..0e9a17a6af5 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc
>+@@ -605,7 +605,7 @@ 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..c94a90de1ba 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>+@@ -21,7 +21,7 @@
>+ // This is not very useful as it does not expand defined symbols if
>+ // called directly. Use its counterpart without the _NO_EXPANSION
>+ // suffix, below.
>+-#define STRINGIZE_NO_EXPANSION(x) #x
>++#define RTC_STRINGIZE_NO_EXPANSION(x) #x
>+ 
>+ // Use this to quote the provided parameter, first expanding it if it
>+ // is a preprocessor symbol.
>+@@ -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) RTC_STRINGIZE_NO_EXPANSION(x)
>+ 
>+ #endif  // RTC_BASE_STRINGIZE_MACROS_H_
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_gcd.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_gcd.cc
>+index c7731dda3ee..f7ba4d1db80 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_gcd.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_gcd.cc
>+@@ -139,7 +139,7 @@ class TaskQueue::Impl : public RefCountInterface {
>+ TaskQueue::Impl::Impl(const char* queue_name,
>+                       TaskQueue* task_queue,
>+                       Priority priority)
>+-    : queue_(dispatch_queue_create(queue_name, DISPATCH_QUEUE_SERIAL)),
>++    : queue_(dispatch_queue_create_with_target(queue_name, DISPATCH_QUEUE_SERIAL, dispatch_get_global_queue(TaskQueuePriorityToGCD(priority), 0))),
>+       context_(new QueueContext(task_queue)) {
>+   RTC_DCHECK(queue_name);
>+   RTC_CHECK(queue_);
>+@@ -148,9 +148,6 @@ TaskQueue::Impl::Impl(const char* queue_name,
>+   // to the queue is released.  This may run after the TaskQueue object has
>+   // been deleted.
>+   dispatch_set_finalizer_f(queue_, &QueueContext::DeleteContext);
>+-
>+-  dispatch_set_target_queue(
>+-      queue_, dispatch_get_global_queue(TaskQueuePriorityToGCD(priority), 0));
>+ }
>+ 
>+ TaskQueue::Impl::~Impl() {
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.h
>+index 4e5019dd4d8..cfa9dd6da96 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.h
>+@@ -38,7 +38,7 @@
>+ #elif (defined(TARGET_OS_IPHONE)  && TARGET_OS_IPHONE)
>+ #define ENABLE_VCP_ENCODER 0
>+ #elif (defined(TARGET_OS_MAC)  && TARGET_OS_MAC)
>+-#define ENABLE_VCP_ENCODER (__MAC_OS_X_VERSION_MAX_ALLOWED >= 101304)
>++#define ENABLE_VCP_ENCODER 0 //(__MAC_OS_X_VERSION_MAX_ALLOWED >= 101304)
>+ #endif
>+ 
>+ #endif
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoToolBoxDecoderFactory.cpp b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoToolBoxDecoderFactory.cpp
>+index 37de8f50070..e7307700080 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoToolBoxDecoderFactory.cpp
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoToolBoxDecoderFactory.cpp
>+@@ -69,7 +69,7 @@ std::unique_ptr<VideoDecoder> VideoToolboxVideoDecoderFactory::CreateVideoDecode
>+ }
>+ 
>+ static inline SdpVideoFormat CreateH264Format(H264::Profile profile, H264::Level level) {
>+-    const rtc::Optional<std::string> profile_string =
>++    const absl::optional<std::string> profile_string =
>+     H264::ProfileLevelIdToString(H264::ProfileLevelId(profile, level));
>+     RTC_CHECK(profile_string);
>+     return SdpVideoFormat(cricket::kH264CodecName,
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoToolBoxEncoderFactory.cpp b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoToolBoxEncoderFactory.cpp
>+index f45ff01780d..296703f35e0 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoToolBoxEncoderFactory.cpp
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoToolBoxEncoderFactory.cpp
>+@@ -69,8 +69,7 @@ void VideoToolboxVideoEncoderFactory::setH264HardwareEncoderAllowed(bool allowed
>+ }
>+ 
>+ static inline SdpVideoFormat CreateH264Format(H264::Profile profile, H264::Level level) {
>+-    const rtc::Optional<std::string> profile_string =
>+-    H264::ProfileLevelIdToString(H264::ProfileLevelId(profile, level));
>++    const absl::optional<std::string> profile_string = H264::ProfileLevelIdToString(H264::ProfileLevelId(profile, level));
>+     RTC_CHECK(profile_string);
>+     return SdpVideoFormat(cricket::kH264CodecName,
>+                           {{cricket::kH264FmtpProfileLevelId, *profile_string},
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.h
>+index 776da18fbc1..1ad27c85260 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.h
>+@@ -25,9 +25,10 @@
>+ 
>+ #pragma once
>+ 
>+-#include "api/video_codecs/video_encoder_factory.h"
>+ #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;
>+ 
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>+index 5e65230f321..240a174534d 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>+@@ -24,13 +24,17 @@
>+  */
>+ 
>+ #include "WebKitUtilities.h"
>++
>+ #include "Common/RTCUIApplicationStatusObserver.h"
>+-#include "VideoToolbox/objc_video_decoder_factory.h"
>+-#include "VideoToolbox/objc_video_encoder_factory.h"
>+ #import "WebRTC/RTCVideoCodecH264.h"
>++
>+ #include "api/video/video_frame.h"
>+-#include <webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h>
>+-#include <webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h>
>++#include "webrtc/sdk/objc/Framework/Native/src/objc_frame_buffer.h"
>++#include "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h"
>++#include "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h"
>++#include "webrtc/sdk/objc/Framework/Native/api/video_decoder_factory.h"
>++#include "webrtc/sdk/objc/Framework/Native/api/video_encoder_factory.h"
>++
>+ 
>+ #if !defined(WEBRTC_IOS)
>+ __attribute__((objc_runtime_name("WK_RTCUIApplicationStatusObserver")))
>+@@ -99,12 +103,12 @@ std::unique_ptr<webrtc::VideoEncoderFactory> createVideoToolboxEncoderFactory()
>+         webrtc::VPModuleInitialize();
>+     });
>+ #endif
>+-    return std::make_unique<webrtc::ObjCVideoEncoderFactory>([[RTCVideoEncoderFactoryH264 alloc] init]);
>++    return ObjCToNativeVideoEncoderFactory([[RTCVideoEncoderFactoryH264 alloc] init]);
>+ }
>+ 
>+ std::unique_ptr<webrtc::VideoDecoderFactory> createVideoToolboxDecoderFactory()
>+ {
>+-    return std::make_unique<webrtc::ObjCVideoDecoderFactory>([[RTCVideoDecoderFactoryH264 alloc] init]);
>++    return ObjCToNativeVideoDecoderFactory([[RTCVideoDecoderFactoryH264 alloc] init]);
>+ }
>+ 
>+ static bool h264HardwareEncoderAllowed = true;
>+@@ -120,17 +124,21 @@ bool isH264HardwareEncoderAllowed()
>+ 
>+ rtc::scoped_refptr<webrtc::VideoFrameBuffer> pixelBufferToFrame(CVPixelBufferRef pixelBuffer)
>+ {
>+-    RTCCVPixelBuffer *frameBuffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer: pixelBuffer];
>+-    return new rtc::RefCountedObject<webrtc::ObjCFrameBuffer>(frameBuffer);
>++    RTCCVPixelBuffer *frameBuffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBuffer];
>++    return new rtc::RefCountedObject<ObjCFrameBuffer>(frameBuffer);
>+ }
>+ 
>+ CVPixelBufferRef pixelBufferFromFrame(const VideoFrame& frame)
>+ {
>+-    auto buffer = frame.video_frame_buffer();
>+-    auto frameBuffer = static_cast<webrtc::ObjCFrameBuffer&>(*buffer).wrapped_frame_buffer();
>+-    if ([frameBuffer isKindOfClass: [RTCCVPixelBuffer class]])
>+-        return [(RTCCVPixelBuffer *)frameBuffer pixelBuffer];
>+-    return nullptr;
>++    if (frame.video_frame_buffer()->type() != VideoFrameBuffer::Type::kNative)
>++        return nullptr;
>++
>++    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/WebKit/decoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/decoder.h
>+index e0a7b03ef22..75b57dad79c 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/decoder.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/decoder.h
>+@@ -37,7 +37,6 @@ class H264VideoToolboxDecoder : public H264Decoder {
>+ 
>+   int Decode(const EncodedImage& input_image,
>+              bool missing_frames,
>+-             const RTPFragmentationHeader* fragmentation,
>+              const CodecSpecificInfo* codec_specific_info,
>+              int64_t render_time_ms) override;
>+ 
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/decoder.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/decoder.mm
>+index 664a99aae56..ca4586bb7b5 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/decoder.mm
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/decoder.mm
>+@@ -99,7 +99,6 @@ int H264VideoToolboxDecoder::InitDecode(const VideoCodec* video_codec,
>+ int H264VideoToolboxDecoder::Decode(
>+     const EncodedImage& input_image,
>+     bool missing_frames,
>+-    const RTPFragmentationHeader* fragmentation,
>+     const CodecSpecificInfo* codec_specific_info,
>+     int64_t render_time_ms) {
>+   RTC_DCHECK(input_image._buffer);
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h
>+index 0c032953be3..c7172288033 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h
>+@@ -12,6 +12,7 @@
>+ 
>+ #import <Foundation/Foundation.h>
>+ 
>++__attribute__((objc_runtime_name("WK_RTCUIApplicationStatusObserver")))
>+ NS_EXTENSION_UNAVAILABLE_IOS("Application status not available in app extensions.")
>+ @interface RTCUIApplicationStatusObserver : NSObject
>+ 
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>+index c167e54757d..d47818f763f 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>+@@ -32,8 +32,46 @@
>+ #include "rtc_base/logging.h"
>+ #include "rtc_base/timeutils.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"
>+ 
>++#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
>++
>+ @interface RTCVideoEncoderH264 ()
>+ 
>+ - (void)frameWasEncoded:(OSStatus)status
>+@@ -167,8 +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) {
>+@@ -284,8 +322,7 @@ @implementation RTCVideoEncoderH264 {
>+   RTCVideoEncoderCallback _callback;
>+   int32_t _width;
>+   int32_t _height;
>+-  VTCompressionSessionRef _compressionSession;
>+-  CVPixelBufferPoolRef _pixelBufferPool;
>++  CompressionSessionRef _compressionSession;
>+   RTCVideoCodecMode _mode;
>+ 
>+   webrtc::H264BitstreamParser _h264BitstreamParser;
>+@@ -308,7 +345,7 @@ - (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo {
>+     RTC_LOG(LS_INFO) << "Using profile " << CFStringToString(_profile);
>+     RTC_CHECK([codecInfo.name isEqualToString:kRTCVideoCodecH264Name]);
>+ 
>+-#if defined(WEBRTC_IOS) && !defined(RTC_APPRTCMOBILE_BROADCAST_EXTENSION)
>++#if defined(WEBRTC_IOS)
>+     [RTCUIApplicationStatusObserver prepareForUse];
>+ #endif
>+   }
>+@@ -329,7 +366,7 @@ - (NSInteger)startEncodeWithSettings:(RTCVideoEncoderSettings *)settings
>+   _mode = settings.mode;
>+ 
>+   // 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
>+@@ -346,7 +383,7 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>+   if (!_callback || !_compressionSession) {
>+     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>+   }
>+-#if defined(WEBRTC_IOS) && !defined(RTC_APPRTCMOBILE_BROADCAST_EXTENSION)
>++#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.
>+@@ -356,7 +393,10 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>+   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;
>+   }
>+ 
>+@@ -373,7 +413,7 @@ - (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;
>+       }
>+@@ -396,7 +436,7 @@ - (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;
>+     }
>+@@ -439,7 +479,7 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>+   // Update the bitrate if needed.
>+   [self setBitrateBps:_bitrateAdjuster->GetAdjustedBitrateBps()];
>+ 
>+-  OSStatus status = VTCompressionSessionEncodeFrame(_compressionSession,
>++  OSStatus status = CompressionSessionEncodeFrame(_compressionSession,
>+                                                     pixelBuffer,
>+                                                     presentationTimeStamp,
>+                                                     kCMTimeInvalid,
>+@@ -452,14 +492,7 @@ - (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;
>+   }
>+@@ -488,44 +521,51 @@ - (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) {
>+@@ -562,22 +602,21 @@ - (int)resetCompressionSessionWithPixelFormat:(OSType)framePixelFormat {
>+     CFRelease(pixelFormat);
>+     pixelFormat = nullptr;
>+   }
>+-  CFMutableDictionaryRef encoder_specs = nullptr;
>++  CFDictionaryRef encoderSpecs = nullptr;
>+ #if defined(WEBRTC_MAC) && !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);
>+ #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,
>+@@ -587,32 +626,105 @@ - (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) << "RTCVideoEncoderH264 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;
>+ }
>+ 
>+@@ -621,6 +733,10 @@ - (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.
>+@@ -638,10 +754,9 @@ - (void)configureCompressionSession {
>+ 
>+ - (void)destroyCompressionSession {
>+   if (_compressionSession) {
>+-    VTCompressionSessionInvalidate(_compressionSession);
>++    CompressionSessionInvalidate(_compressionSession);
>+     CFRelease(_compressionSession);
>+     _compressionSession = nullptr;
>+-    _pixelBufferPool = nullptr;
>+   }
>+ }
>+ 
>+@@ -669,7 +784,7 @@ - (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);
>+@@ -681,7 +796,7 @@ - (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;
>+@@ -698,7 +813,7 @@ - (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) {
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
>+index 15b63d39dc4..8e55852844b 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
>+@@ -28,6 +28,7 @@ extern NSInteger const kRTCAudioSessionErrorConfiguration;
>+ // from AVAudioSession and handle them before calling these delegate methods,
>+ // at which point applications can perform additional processing if required.
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCAudioSessionDelegate")))
>+ @protocol RTCAudioSessionDelegate <NSObject>
>+ 
>+ @optional
>+@@ -103,6 +104,7 @@ RTC_EXPORT
>+  *  case of this is when CallKit activates the audio session for the application
>+  */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCAudioSessionActivationDelegate")))
>+ @protocol RTCAudioSessionActivationDelegate <NSObject>
>+ 
>+ /** Called when the audio session is activated outside of the app by iOS. */
>+@@ -121,6 +123,7 @@ RTC_EXPORT
>+  *  activated only once. See |setActive:error:|.
>+  */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCAudioSession")))
>+ @interface RTCAudioSession : NSObject <RTCAudioSessionActivationDelegate>
>+ 
>+ /** Convenience property to access the AVAudioSession singleton. Callers should
>+@@ -228,6 +231,7 @@ RTC_EXPORT
>+                       error:(NSError **)outError;
>+ @end
>+ 
>++__attribute__((objc_runtime_name("WK_RTCAudioSession")))
>+ @interface RTCAudioSession (Configuration)
>+ 
>+ /** Applies the configuration to the current session. Attempts to set all
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h
>+index 6a02751d294..0097246e7c8 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h
>+@@ -23,6 +23,7 @@ extern const double kRTCAudioSessionLowComplexityIOBufferDuration;
>+ 
>+ // Struct to hold configuration values.
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCAudioSessionConfiguration")))
>+ @interface RTCAudioSessionConfiguration : NSObject
>+ 
>+ @property(nonatomic, strong) NSString *category;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h
>+index 3db31573cb8..7287c31a738 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h
>+@@ -16,6 +16,7 @@
>+ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCAudioSource")))
>+ @interface RTCAudioSource : RTCMediaSource
>+ 
>+ - (instancetype)init NS_UNAVAILABLE;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
>+index b83eede528c..71d797dbd75 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
>+@@ -16,6 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
>+ @class RTCAudioSource;
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCMediaStreamTrack")))
>+ @interface RTCAudioTrack : RTCMediaStreamTrack
>+ 
>+ - (instancetype)init NS_UNAVAILABLE;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
>+index 2995c9c3b10..56ef9281729 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
>+@@ -19,6 +19,7 @@
>+  *  AVCaptureSession.
>+  */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCCameraPreviewView")))
>+ @interface RTCCameraPreviewView : UIView
>+ 
>+ /** The capture session being rendered in the view. Capture session
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
>+index 11ea6a35479..458b20c7ac9 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
>+@@ -69,6 +69,7 @@ typedef NS_ENUM(NSInteger, RTCSdpSemantics) {
>+ 
>+ NS_ASSUME_NONNULL_BEGIN
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCConfiguration")))
>+ @interface RTCConfiguration : NSObject
>+ 
>+ /** An array of Ice Servers available to be used by ICE. */
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
>+index 4859e04bdba..b525a84fc82 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
>+@@ -16,6 +16,7 @@
>+ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCDataBuffer")))
>+ @interface RTCDataBuffer : NSObject
>+ 
>+ /** NSData representation of the underlying buffer. */
>+@@ -60,6 +61,7 @@ typedef NS_ENUM(NSInteger, RTCDataChannelState) {
>+ };
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCDataChannel")))
>+ @interface RTCDataChannel : NSObject
>+ 
>+ /**
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
>+index 65ca8f8d563..567e8838c65 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
>+@@ -16,6 +16,7 @@
>+ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCDataChannelConfiguration")))
>+ @interface RTCDataChannelConfiguration : NSObject
>+ 
>+ /** Set to YES if ordered delivery is required. */
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
>+index 03387471c49..5e149fce0b1 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
>+@@ -26,6 +26,7 @@ typedef NS_ENUM(NSInteger, RTCDispatcherQueueType) {
>+  *  shared dispatch queue.
>+  */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCDispatcher")))
>+ @interface RTCDispatcher : NSObject
>+ 
>+ - (instancetype)init NS_UNAVAILABLE;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
>+index ac8a9104c15..a4e05a70b85 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
>+@@ -34,6 +34,7 @@ NS_ASSUME_NONNULL_BEGIN
>+ // For kRTCFileLoggerTypeApp, the oldest log is overwritten.
>+ // This class is not threadsafe.
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCFileLogger")))
>+ @interface RTCFileLogger : NSObject
>+ 
>+ // The severity level to capture. The default is kRTCFileLoggerSeverityInfo.
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
>+index fb5064e5185..59083dca07c 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
>+@@ -15,6 +15,7 @@
>+ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCIceCandidate")))
>+ @interface RTCIceCandidate : NSObject
>+ 
>+ /**
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h
>+index 727da8acbeb..d1feda40fe7 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h
>+@@ -20,6 +20,7 @@ typedef NS_ENUM(NSUInteger, RTCTlsCertPolicy) {
>+ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCIceServer")))
>+ @interface RTCIceServer : NSObject
>+ 
>+ /** URI(s) for this server represented as NSStrings. */
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
>+index 00508eba17b..1bb27b629c7 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
>+@@ -12,6 +12,7 @@
>+ 
>+ NS_ASSUME_NONNULL_BEGIN
>+ 
>++__attribute__((objc_runtime_name("WK_RTCIntervalRange")))
>+ @interface RTCIntervalRange : NSObject
>+ 
>+ @property(nonatomic, readonly) NSInteger min;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h
>+index 14d996c1cec..bbc4c035f78 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h
>+@@ -16,6 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ /** This does not currently conform to the spec. */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCLegacyStatsReport")))
>+ @interface RTCLegacyStatsReport : NSObject
>+ 
>+ /** Time since 1970-01-01T00:00:00Z in milliseconds. */
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
>+index fce08212e9a..8b358e9c59d 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
>+@@ -39,6 +39,7 @@ RTC_EXTERN NSString *const kRTCMediaConstraintsValueTrue;
>+ RTC_EXTERN NSString *const kRTCMediaConstraintsValueFalse;
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCMediaConstraints")))
>+ @interface RTCMediaConstraints : NSObject
>+ 
>+ - (instancetype)init NS_UNAVAILABLE;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h
>+index 6cfcb7d2ce8..1a4003b5283 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h
>+@@ -22,6 +22,7 @@ typedef NS_ENUM(NSInteger, RTCSourceState) {
>+ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCMediaSource")))
>+ @interface RTCMediaSource : NSObject
>+ 
>+ /** The current state of the RTCMediaSource. */
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h
>+index b97960d436c..9cefde13c5f 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h
>+@@ -19,6 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
>+ @class RTCVideoTrack;
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCMediaStream")))
>+ @interface RTCMediaStream : NSObject
>+ 
>+ /** The audio tracks in this stream. */
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
>+index aee9bdd0a0b..914d8f2e047 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
>+@@ -26,6 +26,7 @@ RTC_EXTERN NSString *const kRTCMediaStreamTrackKindAudio;
>+ RTC_EXTERN NSString *const kRTCMediaStreamTrackKindVideo;
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCMediaStreamTrack")))
>+ @interface RTCMediaStreamTrack : NSObject
>+ 
>+ /**
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h
>+index f01bea9b71b..28b71d1f9e4 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h
>+@@ -15,6 +15,7 @@
>+ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCMetricsSampleInfo")))
>+ @interface RTCMetricsSampleInfo : NSObject
>+ 
>+ /**
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
>+index 94036cc7534..0e930c524e1 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
>+@@ -127,6 +127,7 @@ RTC_EXPORT
>+ @end
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCPeerConnection")))
>+ @interface RTCPeerConnection : NSObject
>+ 
>+ /** The object that will be notifed about events such as state changes and
>+@@ -279,6 +280,7 @@ RTC_EXPORT
>+ 
>+ @end
>+ 
>++__attribute__((objc_runtime_name("WK_RTCPeerConnection")))
>+ @interface RTCPeerConnection (Media)
>+ 
>+ /** Create an RTCRtpSender with the specified kind and media stream ID.
>+@@ -290,6 +292,7 @@ RTC_EXPORT
>+ 
>+ @end
>+ 
>++__attribute__((objc_runtime_name("WK_RTCPeerConnection")))
>+ @interface RTCPeerConnection (DataChannel)
>+ 
>+ /** Create a new data channel with the given label and configuration. */
>+@@ -298,6 +301,7 @@ RTC_EXPORT
>+ 
>+ @end
>+ 
>++__attribute__((objc_runtime_name("WK_RTCPeerConnection")))
>+ @interface RTCPeerConnection (Stats)
>+ 
>+ /** Gather stats for the given RTCMediaStreamTrack. If |mediaStreamTrack| is nil
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
>+index 7f7b7e6c4c4..3fc70e65488 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
>+@@ -28,6 +28,7 @@ NS_ASSUME_NONNULL_BEGIN
>+ @protocol RTCVideoEncoderFactory;
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCPeerConnectionFactory")))
>+ @interface RTCPeerConnectionFactory : NSObject
>+ 
>+ /* Initialize object with default H264 video encoder/decoder factories */
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
>+index bb9aa90507e..341e4289884 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
>+@@ -33,6 +33,7 @@ RTC_EXTERN const NSString *const kRTCH264CodecName;
>+ 
>+ /** Defined in http://w3c.github.io/webrtc-pc/#idl-def-RTCRtpCodecParameters */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCRtpCodecParameters")))
>+ @interface RTCRtpCodecParameters : NSObject
>+ 
>+ /** The RTP payload type. */
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
>+index 5ba0176449d..1230e05b004 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
>+@@ -15,6 +15,7 @@
>+ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ RTC_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/Framework/Headers/WebRTC/RTCRtpParameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
>+index 8f0993489db..ea74644f7e0 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
>+@@ -19,6 +19,7 @@
>+ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCRtpParameters")))
>+ @interface RTCRtpParameters : NSObject
>+ 
>+ /** A unique identifier for the last set of parameters applied. */
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
>+index 6156db9f932..a9b697220a2 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
>+@@ -73,6 +73,7 @@ RTC_EXPORT
>+ @end
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCRtpReceiver")))
>+ @interface RTCRtpReceiver : NSObject <RTCRtpReceiver>
>+ 
>+ - (instancetype)init NS_UNAVAILABLE;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
>+index f6e91250267..01ed2f749dd 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
>+@@ -41,6 +41,7 @@ RTC_EXPORT
>+ @end
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCRtpSender")))
>+ @interface RTCRtpSender : NSObject <RTCRtpSender>
>+ 
>+ - (instancetype)init NS_UNAVAILABLE;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
>+index 7db53da8bea..5ac62d67d3b 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
>+@@ -25,6 +25,7 @@ typedef NS_ENUM(NSInteger, RTCSdpType) {
>+ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCSessionDescription")))
>+ @interface RTCSessionDescription : NSObject
>+ 
>+ /** The type of session description. */
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h
>+index f0a2eaa14c6..60151a1ac15 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h
>+@@ -20,6 +20,7 @@ RTC_EXPORT
>+ @end
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoCapturer")))
>+ @interface RTCVideoCapturer : NSObject
>+ 
>+ @property(nonatomic, weak) id<RTCVideoCapturerDelegate> delegate;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
>+index 5bc6d66d1ee..6fccf44fb93 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
>+@@ -39,6 +39,7 @@ typedef NS_ENUM(NSUInteger, RTCVideoContentType) {
>+ 
>+ /** Represents an encoded frame. Corresponds to webrtc::EncodedImage. */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCEncodedImage")))
>+ @interface RTCEncodedImage : NSObject
>+ 
>+ @property(nonatomic, strong) NSData *buffer;
>+@@ -60,6 +61,7 @@ RTC_EXPORT
>+ 
>+ /** Information for header. Corresponds to webrtc::RTPFragmentationHeader. */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCRtpFragmentationHeader")))
>+ @interface RTCRtpFragmentationHeader : NSObject
>+ 
>+ @property(nonatomic, strong) NSArray<NSNumber *> *fragmentationOffset;
>+@@ -92,6 +94,7 @@ typedef NS_ENUM(NSUInteger, RTCVideoCodecMode) {
>+ 
>+ /** Holds information to identify a codec. Corresponds to webrtc::SdpVideoFormat. */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoCodecInfo")))
>+ @interface RTCVideoCodecInfo : NSObject <NSCoding>
>+ 
>+ - (instancetype)init NS_UNAVAILABLE;
>+@@ -111,6 +114,7 @@ RTC_EXPORT
>+ 
>+ /** Settings for encoder. Corresponds to webrtc::VideoCodec. */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoEncoderSettings")))
>+ @interface RTCVideoEncoderSettings : NSObject
>+ 
>+ @property(nonatomic, strong) NSString *name;
>+@@ -132,6 +136,7 @@ RTC_EXPORT
>+ 
>+ /** QP thresholds for encoder. Corresponds to webrtc::VideoEncoder::QpThresholds. */
>+ RTC_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/Framework/Headers/WebRTC/RTCVideoCodecFactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h
>+index 0ef2c8a655c..e9053025de7 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h
>+@@ -39,6 +39,7 @@ RTC_EXPORT
>+  *  codecs, create custom implementations of RTCVideoEncoderFactory and RTCVideoDecoderFactory.
>+  */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCDefaultVideoEncoderFactory")))
>+ @interface RTCDefaultVideoEncoderFactory : NSObject <RTCVideoEncoderFactory>
>+ 
>+ @property(nonatomic, retain) RTCVideoCodecInfo *preferredCodec;
>+@@ -48,6 +49,7 @@ RTC_EXPORT
>+ @end
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCDefaultVideoDecoderFactory")))
>+ @interface RTCDefaultVideoDecoderFactory : NSObject <RTCVideoDecoderFactory>
>+ @end
>+ 
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
>+index d4dbf88f2b6..84bf9b75028 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
>+@@ -20,6 +20,7 @@ typedef NS_ENUM(NSUInteger, RTCH264PacketizationMode) {
>+ };
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCCodecSpecificInfoH264")))
>+ @interface RTCCodecSpecificInfoH264 : NSObject <RTCCodecSpecificInfo>
>+ 
>+ @property(nonatomic, assign) RTCH264PacketizationMode packetizationMode;
>+@@ -56,6 +57,7 @@ typedef NS_ENUM(NSUInteger, RTCH264Level) {
>+ };
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCH264ProfileLevelId")))
>+ @interface RTCH264ProfileLevelId : NSObject
>+ 
>+ @property(nonatomic, readonly) RTCH264Profile profile;
>+@@ -69,6 +71,7 @@ RTC_EXPORT
>+ 
>+ /** Encoder. */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoEncoderH264")))
>+ @interface RTCVideoEncoderH264 : NSObject <RTCVideoEncoder>
>+ 
>+ - (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo;
>+@@ -77,15 +80,18 @@ RTC_EXPORT
>+ 
>+ /** Decoder. */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoDecoderH264")))
>+ @interface RTCVideoDecoderH264 : NSObject <RTCVideoDecoder>
>+ @end
>+ 
>+ /** Encoder factory. */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoEncoderFactoryH264")))
>+ @interface RTCVideoEncoderFactoryH264 : NSObject <RTCVideoEncoderFactory>
>+ @end
>+ 
>+ /** Decoder factory. */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoDecoderFactoryH264")))
>+ @interface RTCVideoDecoderFactoryH264 : NSObject <RTCVideoDecoderFactory>
>+ @end
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h
>+index 962b6312f7f..3e19ef8909c 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h
>+@@ -14,6 +14,7 @@
>+ #import <WebRTC/RTCVideoCodec.h>
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoDecoderVP8")))
>+ @interface RTCVideoDecoderVP8 : NSObject
>+ 
>+ /* This returns a VP8 decoder that can be returned from a RTCVideoDecoderFactory injected into
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h
>+index faf66ef065b..3fe79bc4a52 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h
>+@@ -14,6 +14,7 @@
>+ #import <WebRTC/RTCVideoCodec.h>
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoDecoderVP9")))
>+ @interface RTCVideoDecoderVP9 : NSObject
>+ 
>+ /* This returns a VP9 decoder that can be returned from a RTCVideoDecoderFactory injected into
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h
>+index e63b24dde75..52155b52551 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h
>+@@ -14,6 +14,7 @@
>+ #import <WebRTC/RTCVideoCodec.h>
>+ 
>+ RTC_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/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h
>+index cef8c82e96f..866e626a05e 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h
>+@@ -14,6 +14,7 @@
>+ #import <WebRTC/RTCVideoCodec.h>
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoEncoderVP9")))
>+ @interface RTCVideoEncoderVP9 : NSObject
>+ 
>+ /* This returns a VP9 encoder that can be returned from a RTCVideoEncoderFactory injected into
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
>+index cfccfa9e940..809363674b4 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
>+@@ -26,6 +26,7 @@ typedef NS_ENUM(NSInteger, RTCVideoRotation) {
>+ 
>+ // RTCVideoFrame is an ObjectiveC version of webrtc::VideoFrame.
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoFrame")))
>+ @interface RTCVideoFrame : NSObject
>+ 
>+ /** Width without rotation applied. */
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h
>+index d590a5f7fc3..3f342536e38 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h
>+@@ -71,6 +71,7 @@ RTC_EXPORT
>+ 
>+ /** RTCVideoFrameBuffer containing a CVPixelBufferRef */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCCVPixelBuffer")))
>+ @interface RTCCVPixelBuffer : NSObject <RTCVideoFrameBuffer>
>+ 
>+ @property(nonatomic, readonly) CVPixelBufferRef pixelBuffer;
>+@@ -105,11 +106,13 @@ RTC_EXPORT
>+ 
>+ /** RTCI420Buffer implements the RTCI420Buffer protocol */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCI420Buffer")))
>+ @interface RTCI420Buffer : NSObject <RTCI420Buffer>
>+ @end
>+ 
>+ /** Mutable version of RTCI420Buffer */
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCMutableI420Buffer")))
>+ @interface RTCMutableI420Buffer : RTCI420Buffer <RTCMutableI420Buffer>
>+ @end
>+ 
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h
>+index a9ebc06c88e..60075e3435a 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h
>+@@ -18,6 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ RTC_EXPORT
>+ 
>++__attribute__((objc_runtime_name("WK_RTCVideoSource")))
>+ @interface RTCVideoSource : RTCMediaSource <RTCVideoCapturerDelegate>
>+ 
>+ - (instancetype)init NS_UNAVAILABLE;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
>+index 899d7c34780..e381c3d1e3e 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
>+@@ -19,6 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
>+ @class RTCVideoSource;
>+ 
>+ RTC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoTrack")))
>+ @interface RTCVideoTrack : RTCMediaStreamTrack
>+ 
>+ /** The video source for this video track. */
>+-- 
>+2.17.1 (Apple Git-112)
>+
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0001-Using-VCP.patch b/Source/ThirdParty/libwebrtc/WebKit/0001-Using-VCP.patch
>deleted file mode 100644
>index 5d08b2a0961..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0001-Using-VCP.patch
>+++ /dev/null
>@@ -1,77 +0,0 @@
>-From f6d216e76cc576b7cac5a980c74d4cf60f28cb17 Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Thu, 22 Mar 2018 15:44:50 -0700
>-Subject: [PATCH] Using VCP
>-
>----
>- .../Classes/VideoToolbox/RTCVideoEncoderH264.mm          | 16 +++++++++-------
>- 1 file changed, 9 insertions(+), 7 deletions(-)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-index d1af6e3c632..138d44573b8 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-@@ -34,6 +34,8 @@
>- #include "sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h"
>- #include "system_wrappers/include/clock.h"
>- #include "third_party/libyuv/include/libyuv/convert_from.h"
>-+
>-+#include "sdk/WebKit/EncoderUtilities.h"
>- #include "sdk/WebKit/WebKitUtilities.h"
>- 
>- @interface RTCVideoEncoderH264 ()
>-@@ -286,7 +288,7 @@ @implementation RTCVideoEncoderH264 {
>-   RTCVideoEncoderCallback _callback;
>-   int32_t _width;
>-   int32_t _height;
>--  VTCompressionSessionRef _compressionSession;
>-+  CompressionSessionRef _compressionSession;
>-   RTCVideoCodecMode _mode;
>- 
>-   webrtc::H264BitstreamParser _h264BitstreamParser;
>-@@ -360,9 +362,9 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>- 
>-   // Get a pixel buffer from the pool and copy frame data over.
>-   CVPixelBufferPoolRef pixelBufferPool =
>--      VTCompressionSessionGetPixelBufferPool(_compressionSession);
>-+      CompressionSessionGetPixelBufferPool(_compressionSession);
>-   if ([self resetCompressionSessionIfNeededForPool:pixelBufferPool withFrame:frame]) {
>--    pixelBufferPool = VTCompressionSessionGetPixelBufferPool(_compressionSession);
>-+    pixelBufferPool = CompressionSessionGetPixelBufferPool(_compressionSession);
>-     isKeyframeRequired = YES;
>-   }
>- 
>-@@ -444,7 +446,7 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>-   // Update the bitrate if needed.
>-   [self setBitrateBps:_bitrateAdjuster->GetAdjustedBitrateBps()];
>- 
>--  OSStatus status = VTCompressionSessionEncodeFrame(_compressionSession,
>-+  OSStatus status = CompressionSessionEncodeFrame(_compressionSession,
>-                                                     pixelBuffer,
>-                                                     presentationTimeStamp,
>-                                                     kCMTimeInvalid,
>-@@ -579,10 +581,10 @@ - (int)resetCompressionSessionWithPixelFormat:(OSType)framePixelFormat {
>-                        webrtc::isH264HardwareEncoderAllowed() ? kCFBooleanTrue : kCFBooleanFalse);
>- #endif
>-   OSStatus status =
>--      VTCompressionSessionCreate(nullptr,  // use default allocator
>-+      CompressionSessionCreate(nullptr,  // use default allocator
>-                                  _width,
>-                                  _height,
>--                                 kCMVideoCodecType_H264,
>-+                                 kCodecTypeH264,
>-                                  encoder_specs,  // use hardware accelerated encoder if available
>-                                  sourceAttributes,
>-                                  nullptr,  // use default compressed data allocator
>-@@ -639,7 +641,7 @@ - (void)configureCompressionSession {
>- 
>- - (void)destroyCompressionSession {
>-   if (_compressionSession) {
>--    VTCompressionSessionInvalidate(_compressionSession);
>-+    CompressionSessionInvalidate(_compressionSession);
>-     CFRelease(_compressionSession);
>-     _compressionSession = nullptr;
>-   }
>--- 
>-2.15.1 (Apple Git-101)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0003-Fixing-VP8-files.patch b/Source/ThirdParty/libwebrtc/WebKit/0003-Fixing-VP8-files.patch
>deleted file mode 100644
>index 06bb0099b02..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0003-Fixing-VP8-files.patch
>+++ /dev/null
>@@ -1,3072 +0,0 @@
>-From be6c0b3edb476e45bf282046e8d1e41c6d4e17cf Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Thu, 14 Dec 2017 13:47:41 -0800
>-Subject: [PATCH 3/8] Fixing VP8 files
>-
>----
>- .../codecs/vp8/default_temporal_layers.cc          | 354 +++++++++---
>- .../codecs/vp8/default_temporal_layers.h           |  59 +-
>- .../codecs/vp8/default_temporal_layers_unittest.cc | 276 +++++++--
>- .../modules/video_coding/codecs/vp8/include/vp8.h  |  16 +-
>- .../codecs/vp8/include/vp8_common_types.h          |  10 +-
>- .../video_coding/codecs/vp8/include/vp8_globals.h  |   8 +-
>- .../video_coding/codecs/vp8/screenshare_layers.cc  | 103 ++--
>- .../video_coding/codecs/vp8/screenshare_layers.h   |  33 +-
>- .../codecs/vp8/screenshare_layers_unittest.cc      |  74 ++-
>- .../codecs/vp8/simulcast_rate_allocator.cc         |   4 +-
>- .../codecs/vp8/simulcast_rate_allocator.h          |  14 +-
>- .../video_coding/codecs/vp8/simulcast_unittest.cc  |  18 +-
>- .../video_coding/codecs/vp8/temporal_layers.h      | 119 +++-
>- .../codecs/vp8/test/vp8_impl_unittest.cc           | 628 +++++++++++----------
>- .../modules/video_coding/codecs/vp8/vp8_impl.cc    | 246 +++++---
>- .../modules/video_coding/codecs/vp8/vp8_impl.h     |  34 +-
>- .../modules/video_coding/codecs/vp8/vp8_noop.cc    |   8 +-
>- 17 files changed, 1364 insertions(+), 640 deletions(-)
>-
>-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 3573aa8f1cb..f070750e7bc 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
>-@@ -1,31 +1,36 @@
>- /* 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.
>--*/
>-+ *
>-+ *  Use of this source code is governed by a BSD-style license
>-+ *  that can be found in the LICENSE file in the root of the source
>-+ *  tree. An additional intellectual property rights grant can 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 "webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h"
>-+#include "modules/video_coding/codecs/vp8/default_temporal_layers.h"
>- 
>- #include <stdlib.h>
>- #include <string.h>
>- 
>- #include <algorithm>
>-+#include <memory>
>-+#include <set>
>- #include <vector>
>- 
>--#include "webrtc/base/checks.h"
>--#include "webrtc/modules/include/module_common_types.h"
>--#include "webrtc/modules/video_coding/include/video_codec_interface.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
>-+#include "modules/include/module_common_types.h"
>-+#include "modules/video_coding/codecs/vp8/include/vp8_common_types.h"
>-+#include "modules/video_coding/include/video_codec_interface.h"
>-+#include "rtc_base/checks.h"
>-+#include "rtc_base/logging.h"
>-+#include "system_wrappers/include/field_trial.h"
>- 
>--#include "vpx/vpx_encoder.h"
>- #include "vpx/vp8cx.h"
>-+#include "vpx/vpx_encoder.h"
>- 
>- namespace webrtc {
>- 
>--TemporalLayers::FrameConfig::FrameConfig() {}
>-+TemporalLayers::FrameConfig::FrameConfig()
>-+    : FrameConfig(kNone, kNone, kNone, false) {}
>- 
>- TemporalLayers::FrameConfig::FrameConfig(TemporalLayers::BufferFlags last,
>-                                          TemporalLayers::BufferFlags golden,
>-@@ -48,6 +53,9 @@ TemporalLayers::FrameConfig::FrameConfig(TemporalLayers::BufferFlags last,
>-       last_buffer_flags(last),
>-       golden_buffer_flags(golden),
>-       arf_buffer_flags(arf),
>-+      encoder_layer_id(0),
>-+      packetizer_temporal_idx(kNoTemporalIdx),
>-+      layer_sync(false),
>-       freeze_entropy(freeze_entropy) {}
>- 
>- namespace {
>-@@ -91,16 +99,19 @@ std::vector<bool> GetTemporalLayerSync(size_t num_layers) {
>-     case 2:
>-       return {false, true, false, false, false, false, false, false};
>-     case 3:
>--      return {false, true, true, false, false, false, false, false};
>-+      if (field_trial::IsEnabled("WebRTC-UseShortVP8TL3Pattern")) {
>-+        return {false, true, true, false};
>-+      } else {
>-+        return {false, true, true, false, false, false, false, false};
>-+      }
>-     case 4:
>--      return {false, true, true,  true, true,  true, false, true,
>--              false, true, false, true, false, true, false, true};
>-+      return {false, true,  true,  false, true,  false, false, false,
>-+              false, false, false, false, false, false, false, false};
>-     default:
>--      RTC_NOTREACHED();
>-       break;
>-   }
>--  RTC_NOTREACHED();
>--  return {false};
>-+  RTC_NOTREACHED() << num_layers;
>-+  return {};
>- }
>- 
>- std::vector<TemporalLayers::FrameConfig> GetTemporalPattern(size_t num_layers) {
>-@@ -129,7 +140,7 @@ std::vector<TemporalLayers::FrameConfig> GetTemporalPattern(size_t num_layers) {
>-       // TL0 also references and updates the 'last' buffer.
>-       // TL1 also references 'last' and references and updates 'golden'.
>-       return {TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>--                                          TemporalLayers::kUpdate,
>-+                                          TemporalLayers::kNone,
>-                                           TemporalLayers::kReference),
>-               TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                           TemporalLayers::kUpdate,
>-@@ -153,35 +164,68 @@ std::vector<TemporalLayers::FrameConfig> GetTemporalPattern(size_t num_layers) {
>-                   TemporalLayers::kReference, TemporalLayers::kReference,
>-                   TemporalLayers::kReference, TemporalLayers::kFreezeEntropy)};
>-     case 3:
>--      // All layers can reference but not update the 'alt' buffer, this means
>--      // that the 'alt' buffer reference is effectively the last keyframe.
>--      // TL0 also references and updates the 'last' buffer.
>--      // TL1 also references 'last' and references and updates 'golden'.
>--      // TL2 references both 'last' and 'golden' but updates no buffer.
>--      return {TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>--                                          TemporalLayers::kUpdate,
>--                                          TemporalLayers::kReference),
>--              TemporalLayers::FrameConfig(
>--                  TemporalLayers::kReference, TemporalLayers::kNone,
>--                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>--              TemporalLayers::FrameConfig(TemporalLayers::kReference,
>--                                          TemporalLayers::kUpdate,
>--                                          TemporalLayers::kReference),
>--              TemporalLayers::FrameConfig(
>--                  TemporalLayers::kReference, TemporalLayers::kReference,
>--                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>--              TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>--                                          TemporalLayers::kNone,
>--                                          TemporalLayers::kReference),
>--              TemporalLayers::FrameConfig(
>--                  TemporalLayers::kReference, TemporalLayers::kReference,
>--                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>--              TemporalLayers::FrameConfig(TemporalLayers::kReference,
>--                                          TemporalLayers::kReferenceAndUpdate,
>--                                          TemporalLayers::kReference),
>--              TemporalLayers::FrameConfig(
>--                  TemporalLayers::kReference, TemporalLayers::kReference,
>--                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy)};
>-+      if (field_trial::IsEnabled("WebRTC-UseShortVP8TL3Pattern")) {
>-+        // This field trial is intended to check if it is worth using a shorter
>-+        // temporal pattern, trading some coding efficiency for less risk of
>-+        // dropped frames.
>-+        // The coding efficiency will decrease somewhat since the higher layer
>-+        // state is more volatile, but it will be offset slightly by updating
>-+        // the altref buffer with TL2 frames, instead of just referencing lower
>-+        // layers.
>-+        // If a frame is dropped in a higher layer, the jitter
>-+        // buffer on the receive side won't be able to decode any higher layer
>-+        // frame until the next sync frame. So we expect a noticeable decrease
>-+        // in frame drops on links with high packet loss.
>-+
>-+        // TL0 references and updates the 'last' buffer.
>-+        // TL1  references 'last' and references and updates 'golden'.
>-+        // TL2 references both 'last' & 'golden' and references and updates
>-+        // 'arf'.
>-+        return {TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>-+                                            TemporalLayers::kNone,
>-+                                            TemporalLayers::kNone),
>-+                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-+                                            TemporalLayers::kNone,
>-+                                            TemporalLayers::kUpdate),
>-+                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-+                                            TemporalLayers::kUpdate,
>-+                                            TemporalLayers::kNone),
>-+                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-+                                            TemporalLayers::kReference,
>-+                                            TemporalLayers::kReference,
>-+                                            TemporalLayers::kFreezeEntropy)};
>-+      } else {
>-+        // All layers can reference but not update the 'alt' buffer, this means
>-+        // that the 'alt' buffer reference is effectively the last keyframe.
>-+        // TL0 also references and updates the 'last' buffer.
>-+        // TL1 also references 'last' and references and updates 'golden'.
>-+        // TL2 references both 'last' and 'golden' but updates no buffer.
>-+        return {TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>-+                                            TemporalLayers::kNone,
>-+                                            TemporalLayers::kReference),
>-+                TemporalLayers::FrameConfig(
>-+                    TemporalLayers::kReference, TemporalLayers::kNone,
>-+                    TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-+                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-+                                            TemporalLayers::kUpdate,
>-+                                            TemporalLayers::kReference),
>-+                TemporalLayers::FrameConfig(
>-+                    TemporalLayers::kReference, TemporalLayers::kReference,
>-+                    TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-+                TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>-+                                            TemporalLayers::kNone,
>-+                                            TemporalLayers::kReference),
>-+                TemporalLayers::FrameConfig(
>-+                    TemporalLayers::kReference, TemporalLayers::kReference,
>-+                    TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-+                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-+                                            TemporalLayers::kReferenceAndUpdate,
>-+                                            TemporalLayers::kReference),
>-+                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-+                                            TemporalLayers::kReference,
>-+                                            TemporalLayers::kReference,
>-+                                            TemporalLayers::kFreezeEntropy)};
>-+      }
>-     case 4:
>-       // TL0 references and updates only the 'last' buffer.
>-       // TL1 references 'last' and updates and references 'golden'.
>-@@ -191,13 +235,13 @@ std::vector<TemporalLayers::FrameConfig> GetTemporalPattern(size_t num_layers) {
>-                                           TemporalLayers::kNone,
>-                                           TemporalLayers::kNone),
>-               TemporalLayers::FrameConfig(
>--                  TemporalLayers::kReference, TemporalLayers::kReference,
>--                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-+                  TemporalLayers::kReference, TemporalLayers::kNone,
>-+                  TemporalLayers::kNone, TemporalLayers::kFreezeEntropy),
>-               TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                           TemporalLayers::kNone,
>-                                           TemporalLayers::kUpdate),
>-               TemporalLayers::FrameConfig(
>--                  TemporalLayers::kReference, TemporalLayers::kReference,
>-+                  TemporalLayers::kReference, TemporalLayers::kNone,
>-                   TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-               TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                           TemporalLayers::kUpdate,
>-@@ -244,6 +288,17 @@ std::vector<TemporalLayers::FrameConfig> GetTemporalPattern(size_t num_layers) {
>-       TemporalLayers::kNone, TemporalLayers::kNone, TemporalLayers::kNone)};
>- }
>- 
>-+// Temporary fix for forced SW fallback.
>-+// For VP8 SW codec, |TemporalLayers| is created and reported to
>-+// SimulcastRateAllocator::OnTemporalLayersCreated but not for VP8 HW.
>-+// Causes an issue when going from forced SW -> HW as |TemporalLayers| is not
>-+// deregistred when deleted by SW codec (tl factory might not exist, owned by
>-+// SimulcastRateAllocator).
>-+bool ExcludeOnTemporalLayersCreated(int num_temporal_layers) {
>-+  return webrtc::field_trial::IsEnabled(
>-+             "WebRTC-VP8-Forced-Fallback-Encoder-v2") &&
>-+         num_temporal_layers == 1;
>-+}
>- }  // namespace
>- 
>- DefaultTemporalLayers::DefaultTemporalLayers(int number_of_temporal_layers,
>-@@ -254,7 +309,6 @@ DefaultTemporalLayers::DefaultTemporalLayers(int number_of_temporal_layers,
>-       temporal_pattern_(GetTemporalPattern(num_layers_)),
>-       tl0_pic_idx_(initial_tl0_pic_idx),
>-       pattern_idx_(255),
>--      timestamp_(0),
>-       last_base_layer_sync_(false) {
>-   RTC_DCHECK_EQ(temporal_pattern_.size(), temporal_layer_sync_.size());
>-   RTC_CHECK_GE(kMaxTemporalStreams, number_of_temporal_layers);
>-@@ -266,12 +320,6 @@ DefaultTemporalLayers::DefaultTemporalLayers(int number_of_temporal_layers,
>-   RTC_DCHECK_LE(temporal_ids_.size(), temporal_pattern_.size());
>- }
>- 
>--int DefaultTemporalLayers::GetTemporalLayerId(
>--    const TemporalLayers::FrameConfig& tl_config) const {
>--  RTC_DCHECK(!tl_config.drop_frame);
>--  return temporal_ids_[tl_config.pattern_idx % temporal_ids_.size()];
>--}
>--
>- uint8_t DefaultTemporalLayers::Tl0PicIdx() const {
>-   return tl0_pic_idx_;
>- }
>-@@ -332,7 +380,10 @@ TemporalLayers::FrameConfig DefaultTemporalLayers::UpdateLayerConfig(
>-   RTC_DCHECK_LT(0, temporal_pattern_.size());
>-   pattern_idx_ = (pattern_idx_ + 1) % temporal_pattern_.size();
>-   TemporalLayers::FrameConfig tl_config = temporal_pattern_[pattern_idx_];
>--  tl_config.pattern_idx = pattern_idx_;
>-+  tl_config.layer_sync =
>-+      temporal_layer_sync_[pattern_idx_ % temporal_layer_sync_.size()];
>-+  tl_config.encoder_layer_id = tl_config.packetizer_temporal_idx =
>-+      temporal_ids_[pattern_idx_ % temporal_ids_.size()];
>-   return tl_config;
>- }
>- 
>-@@ -348,24 +399,19 @@ void DefaultTemporalLayers::PopulateCodecSpecific(
>-     vp8_info->layerSync = false;
>-     vp8_info->tl0PicIdx = kNoTl0PicIdx;
>-   } else {
>-+    vp8_info->temporalIdx = tl_config.packetizer_temporal_idx;
>-+    vp8_info->layerSync = tl_config.layer_sync;
>-     if (frame_is_keyframe) {
>-       vp8_info->temporalIdx = 0;
>-       vp8_info->layerSync = true;
>--    } else {
>--      vp8_info->temporalIdx = GetTemporalLayerId(tl_config);
>--
>--      vp8_info->layerSync = temporal_layer_sync_[tl_config.pattern_idx %
>--                                                 temporal_layer_sync_.size()];
>-     }
>-     if (last_base_layer_sync_ && vp8_info->temporalIdx != 0) {
>-       // Regardless of pattern the frame after a base layer sync will always
>-       // be a layer sync.
>-       vp8_info->layerSync = true;
>-     }
>--    if (vp8_info->temporalIdx == 0 && timestamp != timestamp_) {
>--      timestamp_ = timestamp;
>-+    if (vp8_info->temporalIdx == 0)
>-       tl0_pic_idx_++;
>--    }
>-     last_base_layer_sync_ = frame_is_keyframe;
>-     vp8_info->tl0PicIdx = tl0_pic_idx_;
>-   }
>-@@ -377,13 +423,181 @@ TemporalLayers* TemporalLayersFactory::Create(
>-     uint8_t initial_tl0_pic_idx) const {
>-   TemporalLayers* tl =
>-       new DefaultTemporalLayers(temporal_layers, initial_tl0_pic_idx);
>--  if (listener_)
>-+  if (listener_ && !ExcludeOnTemporalLayersCreated(temporal_layers))
>-     listener_->OnTemporalLayersCreated(simulcast_id, tl);
>-   return tl;
>- }
>- 
>-+std::unique_ptr<TemporalLayersChecker> TemporalLayersFactory::CreateChecker(
>-+    int /*simulcast_id*/,
>-+    int temporal_layers,
>-+    uint8_t initial_tl0_pic_idx) const {
>-+  TemporalLayersChecker* tlc =
>-+      new DefaultTemporalLayersChecker(temporal_layers, initial_tl0_pic_idx);
>-+  return std::unique_ptr<TemporalLayersChecker>(tlc);
>-+}
>-+
>- void TemporalLayersFactory::SetListener(TemporalLayersListener* listener) {
>-   listener_ = listener;
>- }
>- 
>-+// Returns list of temporal dependencies for each frame in the temporal pattern.
>-+// Values are lists of indecies in the pattern.
>-+std::vector<std::set<uint8_t>> GetTemporalDependencies(
>-+    int num_temporal_layers) {
>-+  switch (num_temporal_layers) {
>-+    case 1:
>-+      return {{0}};
>-+    case 2:
>-+      return {{6}, {0}, {0}, {1, 2}, {2}, {3, 4}, {4}, {5, 6}};
>-+    case 3:
>-+      if (field_trial::IsEnabled("WebRTC-UseShortVP8TL3Pattern")) {
>-+        return {{0}, {0}, {0}, {0, 1, 2}};
>-+      } else {
>-+        return {{4}, {0}, {0}, {0, 2}, {0}, {2, 4}, {2, 4}, {4, 6}};
>-+      }
>-+    case 4:
>-+      return {{8},    {0},         {0},         {0, 2},
>-+              {0},    {0, 2, 4},   {0, 2, 4},   {0, 4, 6},
>-+              {0},    {4, 6, 8},   {4, 6, 8},   {4, 8, 10},
>-+              {4, 8}, {8, 10, 12}, {8, 10, 12}, {8, 12, 14}};
>-+    default:
>-+      RTC_NOTREACHED();
>-+      return {};
>-+  }
>-+}
>-+
>-+DefaultTemporalLayersChecker::DefaultTemporalLayersChecker(
>-+    int num_temporal_layers,
>-+    uint8_t initial_tl0_pic_idx)
>-+    : TemporalLayersChecker(num_temporal_layers, initial_tl0_pic_idx),
>-+      num_layers_(std::max(1, num_temporal_layers)),
>-+      temporal_ids_(GetTemporalIds(num_layers_)),
>-+      temporal_dependencies_(GetTemporalDependencies(num_layers_)),
>-+      pattern_idx_(255) {
>-+  int i = 0;
>-+  while (temporal_ids_.size() < temporal_dependencies_.size()) {
>-+    temporal_ids_.push_back(temporal_ids_[i++]);
>-+  }
>-+}
>-+
>-+bool DefaultTemporalLayersChecker::CheckTemporalConfig(
>-+    bool frame_is_keyframe,
>-+    const TemporalLayers::FrameConfig& frame_config) {
>-+  if (!TemporalLayersChecker::CheckTemporalConfig(frame_is_keyframe,
>-+                                                  frame_config)) {
>-+    return false;
>-+  }
>-+  if (frame_config.drop_frame) {
>-+    return true;
>-+  }
>-+  ++pattern_idx_;
>-+  if (pattern_idx_ == temporal_ids_.size()) {
>-+    // All non key-frame buffers should be updated each pattern cycle.
>-+    if (!last_.is_keyframe && !last_.is_updated_this_cycle) {
>-+      RTC_LOG(LS_ERROR) << "Last buffer was not updated during pattern cycle.";
>-+      return false;
>-+    }
>-+    if (!arf_.is_keyframe && !arf_.is_updated_this_cycle) {
>-+      RTC_LOG(LS_ERROR) << "Arf buffer was not updated during pattern cycle.";
>-+      return false;
>-+    }
>-+    if (!golden_.is_keyframe && !golden_.is_updated_this_cycle) {
>-+      RTC_LOG(LS_ERROR)
>-+          << "Golden buffer was not updated during pattern cycle.";
>-+      return false;
>-+    }
>-+    last_.is_updated_this_cycle = false;
>-+    arf_.is_updated_this_cycle = false;
>-+    golden_.is_updated_this_cycle = false;
>-+    pattern_idx_ = 0;
>-+  }
>-+  uint8_t expected_tl_idx = temporal_ids_[pattern_idx_];
>-+  if (frame_config.packetizer_temporal_idx != expected_tl_idx) {
>-+    RTC_LOG(LS_ERROR) << "Frame has an incorrect temporal index. Expected: "
>-+                      << static_cast<int>(expected_tl_idx) << " Actual: "
>-+                      << static_cast<int>(frame_config.packetizer_temporal_idx);
>-+    return false;
>-+  }
>-+
>-+  bool need_sync = temporal_ids_[pattern_idx_] > 0 &&
>-+                   temporal_ids_[pattern_idx_] != kNoTemporalIdx;
>-+  std::vector<int> dependencies;
>-+
>-+  if (frame_config.last_buffer_flags &
>-+      TemporalLayers::BufferFlags::kReference) {
>-+    uint8_t referenced_layer = temporal_ids_[last_.pattern_idx];
>-+    if (referenced_layer > 0) {
>-+      need_sync = false;
>-+    }
>-+    if (!last_.is_keyframe) {
>-+      dependencies.push_back(last_.pattern_idx);
>-+    }
>-+  }
>-+
>-+  if (frame_config.arf_buffer_flags & TemporalLayers::BufferFlags::kReference) {
>-+    uint8_t referenced_layer = temporal_ids_[arf_.pattern_idx];
>-+    if (referenced_layer > 0) {
>-+      need_sync = false;
>-+    }
>-+    if (!arf_.is_keyframe) {
>-+      dependencies.push_back(arf_.pattern_idx);
>-+    }
>-+  }
>-+
>-+  if (frame_config.golden_buffer_flags &
>-+      TemporalLayers::BufferFlags::kReference) {
>-+    uint8_t referenced_layer = temporal_ids_[golden_.pattern_idx];
>-+    if (referenced_layer > 0) {
>-+      need_sync = false;
>-+    }
>-+    if (!golden_.is_keyframe) {
>-+      dependencies.push_back(golden_.pattern_idx);
>-+    }
>-+  }
>-+
>-+  if (need_sync != frame_config.layer_sync) {
>-+    RTC_LOG(LS_ERROR) << "Sync bit is set incorrectly on a frame. Expected: "
>-+                      << need_sync << " Actual: " << frame_config.layer_sync;
>-+    return false;
>-+  }
>-+
>-+  if (!frame_is_keyframe) {
>-+    size_t i;
>-+    for (i = 0; i < dependencies.size(); ++i) {
>-+      if (temporal_dependencies_[pattern_idx_].find(dependencies[i]) ==
>-+          temporal_dependencies_[pattern_idx_].end()) {
>-+        RTC_LOG(LS_ERROR)
>-+            << "Illegal temporal dependency out of defined pattern "
>-+               "from position "
>-+            << static_cast<int>(pattern_idx_) << " to position "
>-+            << static_cast<int>(dependencies[i]);
>-+        return false;
>-+      }
>-+    }
>-+  }
>-+
>-+  if (frame_config.last_buffer_flags & TemporalLayers::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) {
>-+    arf_.is_updated_this_cycle = true;
>-+    arf_.pattern_idx = pattern_idx_;
>-+    arf_.is_keyframe = false;
>-+  }
>-+  if (frame_config.golden_buffer_flags & TemporalLayers::BufferFlags::kUpdate) {
>-+    golden_.is_updated_this_cycle = true;
>-+    golden_.pattern_idx = pattern_idx_;
>-+    golden_.is_keyframe = false;
>-+  }
>-+  if (frame_is_keyframe) {
>-+    last_.is_keyframe = true;
>-+    arf_.is_keyframe = true;
>-+    golden_.is_keyframe = true;
>-+  }
>-+  return true;
>-+}
>-+
>- }  // namespace webrtc
>-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 4681fe9dc84..f1925569209 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
>-@@ -1,22 +1,23 @@
>- /* 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.
>--*/
>-+ *
>-+ *  Use of this source code is governed by a BSD-style license
>-+ *  that can be found in the LICENSE file in the root of the source
>-+ *  tree. An additional intellectual property rights grant can 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 defines classes for doing temporal layers with VP8.
>--*/
>--#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
>--#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
>-+ * This file defines classes for doing temporal layers with VP8.
>-+ */
>-+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
>-+#define MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
>- 
>-+#include <set>
>- #include <vector>
>- 
>--#include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
>-+#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>- 
>--#include "webrtc/base/optional.h"
>-+#include "api/optional.h"
>- 
>- namespace webrtc {
>- 
>-@@ -45,9 +46,6 @@ class DefaultTemporalLayers : public TemporalLayers {
>- 
>-   void FrameEncoded(unsigned int size, int qp) override {}
>- 
>--  int GetTemporalLayerId(
>--      const TemporalLayers::FrameConfig& references) const override;
>--
>-   uint8_t Tl0PicIdx() const override;
>- 
>-  private:
>-@@ -58,10 +56,35 @@ class DefaultTemporalLayers : public TemporalLayers {
>- 
>-   uint8_t tl0_pic_idx_;
>-   uint8_t pattern_idx_;
>--  uint32_t timestamp_;
>-   bool last_base_layer_sync_;
>-   rtc::Optional<std::vector<uint32_t>> new_bitrates_kbps_;
>- };
>- 
>-+class DefaultTemporalLayersChecker : public TemporalLayersChecker {
>-+ public:
>-+  DefaultTemporalLayersChecker(int number_of_temporal_layers,
>-+                               uint8_t initial_tl0_pic_idx);
>-+  bool CheckTemporalConfig(
>-+      bool frame_is_keyframe,
>-+      const TemporalLayers::FrameConfig& frame_config) override;
>-+
>-+ private:
>-+  struct BufferState {
>-+    BufferState()
>-+        : is_updated_this_cycle(false), is_keyframe(true), pattern_idx(0) {}
>-+
>-+    bool is_updated_this_cycle;
>-+    bool is_keyframe;
>-+    uint8_t pattern_idx;
>-+  };
>-+  const size_t num_layers_;
>-+  std::vector<unsigned int> temporal_ids_;
>-+  const std::vector<std::set<uint8_t>> temporal_dependencies_;
>-+  BufferState last_;
>-+  BufferState arf_;
>-+  BufferState golden_;
>-+  uint8_t pattern_idx_;
>-+};
>-+
>- }  // namespace webrtc
>--#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
>-+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
>-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 edab34c16dd..1c1d63f1f8d 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
>-@@ -8,19 +8,20 @@
>-  *  be found in the AUTHORS file in the root of the source tree.
>-  */
>- 
>--#include "webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h"
>-+#include "modules/video_coding/codecs/vp8/default_temporal_layers.h"
>-+#include "modules/video_coding/codecs/vp8/vp8_impl.h"
>-+#include "modules/video_coding/include/video_codec_interface.h"
>-+#include "test/field_trial.h"
>-+#include "test/gtest.h"
>- #include "vpx/vp8cx.h"
>- #include "vpx/vpx_encoder.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/vp8_impl.h"
>--#include "webrtc/modules/video_coding/include/video_codec_interface.h"
>--#include "webrtc/test/gtest.h"
>- 
>- namespace webrtc {
>-+namespace test {
>- 
>- enum {
>-   kTemporalUpdateLast = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
>--                        VP8_EFLAG_NO_REF_GF |
>--                        VP8_EFLAG_NO_REF_ARF,
>-+                        VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_ARF,
>-   kTemporalUpdateGoldenWithoutDependency =
>-       VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_ARF |
>-       VP8_EFLAG_NO_UPD_LAST,
>-@@ -31,16 +32,16 @@ enum {
>-       VP8_EFLAG_NO_UPD_LAST,
>-   kTemporalUpdateAltref = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_LAST,
>-   kTemporalUpdateNone = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
>--                        VP8_EFLAG_NO_UPD_LAST |
>--                        VP8_EFLAG_NO_UPD_ENTROPY,
>--  kTemporalUpdateNoneNoRefAltRef = VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_GF |
>--                                   VP8_EFLAG_NO_UPD_ARF |
>--                                   VP8_EFLAG_NO_UPD_LAST |
>--                                   VP8_EFLAG_NO_UPD_ENTROPY,
>--  kTemporalUpdateNoneNoRefGolden = VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF |
>--                                   VP8_EFLAG_NO_UPD_ARF |
>--                                   VP8_EFLAG_NO_UPD_LAST |
>--                                   VP8_EFLAG_NO_UPD_ENTROPY,
>-+                        VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ENTROPY,
>-+  kTemporalUpdateNoneNoRefAltRef =
>-+      VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
>-+      VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ENTROPY,
>-+  kTemporalUpdateNoneNoRefGolden =
>-+      VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
>-+      VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ENTROPY,
>-+  kTemporalUpdateNoneNoRefGoldenAltRef =
>-+      VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_REF_ARF |
>-+      VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ENTROPY,
>-   kTemporalUpdateGoldenWithoutDependencyRefAltRef =
>-       VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
>-   kTemporalUpdateGoldenRefAltRef = VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
>-@@ -52,13 +53,14 @@ enum {
>- 
>- TEST(TemporalLayersTest, 2Layers) {
>-   DefaultTemporalLayers tl(2, 0);
>-+  DefaultTemporalLayersChecker checker(2, 0);
>-   vpx_codec_enc_cfg_t cfg;
>-   CodecSpecificInfoVP8 vp8_info;
>-   tl.OnRatesUpdated(500, 500, 30);
>-   tl.UpdateConfiguration(&cfg);
>- 
>-   int expected_flags[16] = {
>--      kTemporalUpdateLastAndGoldenRefAltRef,
>-+      kTemporalUpdateLastRefAltRef,
>-       kTemporalUpdateGoldenWithoutDependencyRefAltRef,
>-       kTemporalUpdateLastRefAltRef,
>-       kTemporalUpdateGoldenRefAltRef,
>-@@ -66,7 +68,7 @@ TEST(TemporalLayersTest, 2Layers) {
>-       kTemporalUpdateGoldenRefAltRef,
>-       kTemporalUpdateLastRefAltRef,
>-       kTemporalUpdateNone,
>--      kTemporalUpdateLastAndGoldenRefAltRef,
>-+      kTemporalUpdateLastRefAltRef,
>-       kTemporalUpdateGoldenWithoutDependencyRefAltRef,
>-       kTemporalUpdateLastRefAltRef,
>-       kTemporalUpdateGoldenRefAltRef,
>-@@ -85,23 +87,28 @@ TEST(TemporalLayersTest, 2Layers) {
>-   uint32_t timestamp = 0;
>-   for (int i = 0; i < 16; ++i) {
>-     TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>--    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config));
>--    tl.PopulateCodecSpecific(false, tl_config, &vp8_info, 0);
>-+    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config)) << i;
>-+    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_info, 0);
>-+    EXPECT_TRUE(checker.CheckTemporalConfig(i == 0, tl_config));
>-     EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
>--    EXPECT_EQ(expected_layer_sync[i], vp8_info.layerSync);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.encoder_layer_id);
>-+    EXPECT_EQ(i == 0 || expected_layer_sync[i], vp8_info.layerSync);
>-+    EXPECT_EQ(expected_layer_sync[i], tl_config.layer_sync);
>-     timestamp += 3000;
>-   }
>- }
>- 
>- TEST(TemporalLayersTest, 3Layers) {
>-   DefaultTemporalLayers tl(3, 0);
>-+  DefaultTemporalLayersChecker checker(3, 0);
>-   vpx_codec_enc_cfg_t cfg;
>-   CodecSpecificInfoVP8 vp8_info;
>-   tl.OnRatesUpdated(500, 500, 30);
>-   tl.UpdateConfiguration(&cfg);
>- 
>-   int expected_flags[16] = {
>--      kTemporalUpdateLastAndGoldenRefAltRef,
>-+      kTemporalUpdateLastRefAltRef,
>-       kTemporalUpdateNoneNoRefGolden,
>-       kTemporalUpdateGoldenWithoutDependencyRefAltRef,
>-       kTemporalUpdateNone,
>-@@ -109,7 +116,7 @@ TEST(TemporalLayersTest, 3Layers) {
>-       kTemporalUpdateNone,
>-       kTemporalUpdateGoldenRefAltRef,
>-       kTemporalUpdateNone,
>--      kTemporalUpdateLastAndGoldenRefAltRef,
>-+      kTemporalUpdateLastRefAltRef,
>-       kTemporalUpdateNoneNoRefGolden,
>-       kTemporalUpdateGoldenWithoutDependencyRefAltRef,
>-       kTemporalUpdateNone,
>-@@ -128,25 +135,67 @@ TEST(TemporalLayersTest, 3Layers) {
>-   unsigned int timestamp = 0;
>-   for (int i = 0; i < 16; ++i) {
>-     TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>--    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config));
>--    tl.PopulateCodecSpecific(false, tl_config, &vp8_info, 0);
>-+    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config)) << i;
>-+    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_info, 0);
>-+    EXPECT_TRUE(checker.CheckTemporalConfig(i == 0, tl_config));
>-+    EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.encoder_layer_id);
>-+    EXPECT_EQ(i == 0 || expected_layer_sync[i], vp8_info.layerSync);
>-+    EXPECT_EQ(expected_layer_sync[i], tl_config.layer_sync);
>-+    timestamp += 3000;
>-+  }
>-+}
>-+
>-+TEST(TemporalLayersTest, Alternative3Layers) {
>-+  ScopedFieldTrials field_trial("WebRTC-UseShortVP8TL3Pattern/Enabled/");
>-+  DefaultTemporalLayers tl(3, 0);
>-+  DefaultTemporalLayersChecker checker(3, 0);
>-+  vpx_codec_enc_cfg_t cfg;
>-+  CodecSpecificInfoVP8 vp8_info;
>-+  tl.OnRatesUpdated(500, 500, 30);
>-+  tl.UpdateConfiguration(&cfg);
>-+
>-+  int expected_flags[8] = {kTemporalUpdateLast,
>-+                           kTemporalUpdateAltrefWithoutDependency,
>-+                           kTemporalUpdateGoldenWithoutDependency,
>-+                           kTemporalUpdateNone,
>-+                           kTemporalUpdateLast,
>-+                           kTemporalUpdateAltrefWithoutDependency,
>-+                           kTemporalUpdateGoldenWithoutDependency,
>-+                           kTemporalUpdateNone};
>-+  int expected_temporal_idx[8] = {0, 2, 1, 2, 0, 2, 1, 2};
>-+
>-+  bool expected_layer_sync[8] = {false, true, true, false,
>-+                                 false, true, true, false};
>-+
>-+  unsigned int timestamp = 0;
>-+  for (int i = 0; i < 8; ++i) {
>-+    TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>-+    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config)) << i;
>-+    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_info, 0);
>-+    EXPECT_TRUE(checker.CheckTemporalConfig(i == 0, tl_config));
>-     EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
>--    EXPECT_EQ(expected_layer_sync[i], vp8_info.layerSync);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.encoder_layer_id);
>-+    EXPECT_EQ(i == 0 || expected_layer_sync[i], vp8_info.layerSync);
>-+    EXPECT_EQ(expected_layer_sync[i], tl_config.layer_sync);
>-     timestamp += 3000;
>-   }
>- }
>- 
>- TEST(TemporalLayersTest, 4Layers) {
>-   DefaultTemporalLayers tl(4, 0);
>-+  DefaultTemporalLayersChecker checker(4, 0);
>-   vpx_codec_enc_cfg_t cfg;
>-   CodecSpecificInfoVP8 vp8_info;
>-   tl.OnRatesUpdated(500, 500, 30);
>-   tl.UpdateConfiguration(&cfg);
>-   int expected_flags[16] = {
>-       kTemporalUpdateLast,
>--      kTemporalUpdateNone,
>-+      kTemporalUpdateNoneNoRefGoldenAltRef,
>-       kTemporalUpdateAltrefWithoutDependency,
>--      kTemporalUpdateNone,
>-+      kTemporalUpdateNoneNoRefGolden,
>-       kTemporalUpdateGoldenWithoutDependency,
>-       kTemporalUpdateNone,
>-       kTemporalUpdateAltref,
>-@@ -163,30 +212,35 @@ TEST(TemporalLayersTest, 4Layers) {
>-   int expected_temporal_idx[16] = {0, 3, 2, 3, 1, 3, 2, 3,
>-                                    0, 3, 2, 3, 1, 3, 2, 3};
>- 
>--  bool expected_layer_sync[16] = {false, true, true,  true, true,  true,
>--                                  false, true, false, true, false, true,
>--                                  false, true, false, true};
>-+  bool expected_layer_sync[16] = {false, true,  true,  false, true,  false,
>-+                                  false, false, false, false, false, false,
>-+                                  false, false, false, false};
>- 
>-   uint32_t timestamp = 0;
>-   for (int i = 0; i < 16; ++i) {
>-     TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>--    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config));
>--    tl.PopulateCodecSpecific(false, tl_config, &vp8_info, 0);
>-+    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config)) << i;
>-+    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_info, 0);
>-+    EXPECT_TRUE(checker.CheckTemporalConfig(i == 0, tl_config));
>-     EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
>--    EXPECT_EQ(expected_layer_sync[i], vp8_info.layerSync);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.encoder_layer_id);
>-+    EXPECT_EQ(i == 0 || expected_layer_sync[i], vp8_info.layerSync);
>-+    EXPECT_EQ(expected_layer_sync[i], tl_config.layer_sync);
>-     timestamp += 3000;
>-   }
>- }
>- 
>- TEST(TemporalLayersTest, KeyFrame) {
>-   DefaultTemporalLayers tl(3, 0);
>-+  DefaultTemporalLayersChecker checker(3, 0);
>-   vpx_codec_enc_cfg_t cfg;
>-   CodecSpecificInfoVP8 vp8_info;
>-   tl.OnRatesUpdated(500, 500, 30);
>-   tl.UpdateConfiguration(&cfg);
>- 
>-   int expected_flags[8] = {
>--      kTemporalUpdateLastAndGoldenRefAltRef,
>-+      kTemporalUpdateLastRefAltRef,
>-       kTemporalUpdateNoneNoRefGolden,
>-       kTemporalUpdateGoldenWithoutDependencyRefAltRef,
>-       kTemporalUpdateNone,
>-@@ -195,21 +249,161 @@ TEST(TemporalLayersTest, KeyFrame) {
>-       kTemporalUpdateGoldenRefAltRef,
>-       kTemporalUpdateNone,
>-   };
>--  int expected_temporal_idx[8] = {0, 0, 0, 0, 0, 0, 0, 2};
>-+  int expected_temporal_idx[8] = {0, 2, 1, 2, 0, 2, 1, 2};
>-+  bool expected_layer_sync[8] = {false, true,  true,  false,
>-+                                 false, false, false, false};
>- 
>-   uint32_t timestamp = 0;
>-   for (int i = 0; i < 7; ++i) {
>-     TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>--    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config));
>-+    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config)) << i;
>-     tl.PopulateCodecSpecific(true, tl_config, &vp8_info, 0);
>--    EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
>--    EXPECT_EQ(true, vp8_info.layerSync);
>-+    EXPECT_TRUE(checker.CheckTemporalConfig(true, tl_config));
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.encoder_layer_id);
>-+    EXPECT_EQ(0, vp8_info.temporalIdx)
>-+        << "Key frame should always be packetized as layer 0";
>-+    EXPECT_EQ(expected_layer_sync[i], tl_config.layer_sync);
>-+    EXPECT_TRUE(vp8_info.layerSync) << "Key frame should be marked layer sync.";
>-     timestamp += 3000;
>-   }
>-   TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>-   EXPECT_EQ(expected_flags[7], VP8EncoderImpl::EncodeFlags(tl_config));
>-   tl.PopulateCodecSpecific(false, tl_config, &vp8_info, 0);
>--  EXPECT_EQ(expected_temporal_idx[7], vp8_info.temporalIdx);
>--  EXPECT_EQ(true, vp8_info.layerSync);
>-+  EXPECT_TRUE(checker.CheckTemporalConfig(false, tl_config));
>-+  EXPECT_NE(0, vp8_info.temporalIdx)
>-+      << "To test something useful, this frame should not use layer 0.";
>-+  EXPECT_EQ(expected_temporal_idx[7], vp8_info.temporalIdx)
>-+      << "Non-keyframe, should use frame temporal index.";
>-+  EXPECT_EQ(expected_temporal_idx[7], tl_config.packetizer_temporal_idx);
>-+  EXPECT_EQ(expected_temporal_idx[7], tl_config.encoder_layer_id);
>-+  EXPECT_FALSE(tl_config.layer_sync);
>-+  EXPECT_TRUE(vp8_info.layerSync) << "Frame after keyframe should always be "
>-+                                     "marked layer sync since it only depends "
>-+                                     "on the base layer.";
>-+}
>-+
>-+class TemporalLayersReferenceTest : public ::testing::TestWithParam<int> {
>-+ public:
>-+  TemporalLayersReferenceTest()
>-+      : timestamp_(1),
>-+        last_sync_timestamp_(timestamp_),
>-+        tl0_reference_(nullptr) {}
>-+  virtual ~TemporalLayersReferenceTest() {}
>-+
>-+ protected:
>-+  static const int kMaxPatternLength = 32;
>-+
>-+  struct BufferState {
>-+    BufferState() : BufferState(-1, 0, false) {}
>-+    BufferState(int temporal_idx, uint32_t timestamp, bool sync)
>-+        : temporal_idx(temporal_idx), timestamp(timestamp), sync(sync) {}
>-+    int temporal_idx;
>-+    uint32_t timestamp;
>-+    bool sync;
>-+  };
>-+
>-+  bool UpdateSyncRefState(const TemporalLayers::BufferFlags& flags,
>-+                          BufferState* buffer_state) {
>-+    if (flags & TemporalLayers::kReference) {
>-+      if (buffer_state->temporal_idx == -1)
>-+        return true;  // References key-frame.
>-+      if (buffer_state->temporal_idx == 0) {
>-+        // No more than one reference to TL0 frame.
>-+        EXPECT_EQ(nullptr, tl0_reference_);
>-+        tl0_reference_ = buffer_state;
>-+        return true;
>-+      }
>-+      return false;  // References higher layer.
>-+    }
>-+    return true;  // No reference, does not affect sync frame status.
>-+  }
>-+
>-+  void ValidateReference(const TemporalLayers::BufferFlags& flags,
>-+                         const BufferState& buffer_state,
>-+                         int temporal_layer) {
>-+    if (flags & TemporalLayers::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_);
>-+      }
>-+      // No reference to buffer in higher layer.
>-+      EXPECT_LE(buffer_state.temporal_idx, temporal_layer);
>-+    }
>-+  }
>-+
>-+  uint32_t timestamp_ = 1;
>-+  uint32_t last_sync_timestamp_ = timestamp_;
>-+  BufferState* tl0_reference_;
>-+
>-+  BufferState last_state;
>-+  BufferState golden_state;
>-+  BufferState altref_state;
>-+};
>-+
>-+INSTANTIATE_TEST_CASE_P(DefaultTemporalLayersTest,
>-+                        TemporalLayersReferenceTest,
>-+                        ::testing::Range(1, kMaxTemporalStreams + 1));
>-+
>-+TEST_P(TemporalLayersReferenceTest, ValidFrameConfigs) {
>-+  const int num_layers = GetParam();
>-+  DefaultTemporalLayers tl(num_layers, 0);
>-+  vpx_codec_enc_cfg_t cfg;
>-+  tl.OnRatesUpdated(500, 500, 30);
>-+  tl.UpdateConfiguration(&cfg);
>-+
>-+  // Run through the pattern and store the frame dependencies, plus keep track
>-+  // of the buffer state; which buffers references which temporal layers (if
>-+  // (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);
>-+  for (int i = 0; i < kMaxPatternLength; ++i) {
>-+    TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp_++);
>-+    EXPECT_FALSE(tl_config.drop_frame);
>-+    tl_configs.push_back(tl_config);
>-+    int temporal_idx = tl_config.encoder_layer_id;
>-+    // For the default layers, always keep encoder and rtp layers in sync.
>-+    EXPECT_EQ(tl_config.packetizer_temporal_idx, temporal_idx);
>-+
>-+    // Determine if this frame is in a higher layer but references only TL0
>-+    // or untouched buffers, if so verify it is marked as a layer sync.
>-+    bool is_sync_frame = true;
>-+    tl0_reference_ = nullptr;
>-+    if (temporal_idx <= 0) {
>-+      is_sync_frame = false;  // TL0 by definition not a sync frame.
>-+    } else if (!UpdateSyncRefState(tl_config.last_buffer_flags, &last_state)) {
>-+      is_sync_frame = false;
>-+    } else if (!UpdateSyncRefState(tl_config.golden_buffer_flags,
>-+                                   &golden_state)) {
>-+      is_sync_frame = false;
>-+    } else if (!UpdateSyncRefState(tl_config.arf_buffer_flags, &altref_state)) {
>-+      is_sync_frame = false;
>-+    }
>-+    if (is_sync_frame) {
>-+      // Cache timestamp for last found sync frame, so that we can verify no
>-+      // references back past this frame.
>-+      ASSERT_TRUE(tl0_reference_);
>-+      last_sync_timestamp_ = tl0_reference_->timestamp;
>-+    }
>-+    EXPECT_EQ(tl_config.layer_sync, is_sync_frame);
>-+
>-+    // Validate no reference from lower to high temporal layer, or backwards
>-+    // past last reference frame.
>-+    ValidateReference(tl_config.last_buffer_flags, last_state, temporal_idx);
>-+    ValidateReference(tl_config.golden_buffer_flags, golden_state,
>-+                      temporal_idx);
>-+    ValidateReference(tl_config.arf_buffer_flags, altref_state, temporal_idx);
>-+
>-+    // Update the current layer state.
>-+    BufferState state = {temporal_idx, timestamp_, is_sync_frame};
>-+    if (tl_config.last_buffer_flags & TemporalLayers::kUpdate)
>-+      last_state = state;
>-+    if (tl_config.golden_buffer_flags & TemporalLayers::kUpdate)
>-+      golden_state = state;
>-+    if (tl_config.arf_buffer_flags & TemporalLayers::kUpdate)
>-+      altref_state = state;
>-+  }
>- }
>-+}  // namespace test
>- }  // namespace webrtc
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
>-index c9e45ee5b65..06003613adf 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
>-@@ -10,28 +10,30 @@
>-  *  WEBRTC VP8 wrapper interface
>-  */
>- 
>--#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
>--#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
>-+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
>-+#define MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
>- 
>--#include "webrtc/modules/video_coding/include/video_codec_interface.h"
>-+#include <memory>
>-+
>-+#include "modules/video_coding/include/video_codec_interface.h"
>- 
>- namespace webrtc {
>- 
>- class VP8Encoder : public VideoEncoder {
>-  public:
>-   static bool IsSupported();
>--  static VP8Encoder* Create();
>-+  static std::unique_ptr<VP8Encoder> Create();
>- 
>-   virtual ~VP8Encoder() {}
>- };  // end of VP8Encoder class
>- 
>- class VP8Decoder : public VideoDecoder {
>-  public:
>--  static bool IsSupported();
>--  static VP8Decoder* Create();
>-+    static bool IsSupported();
>-+    static std::unique_ptr<VP8Decoder> Create();
>- 
>-   virtual ~VP8Decoder() {}
>- };  // end of VP8Decoder class
>- }  // namespace webrtc
>- 
>--#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
>-+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h
>-index 7a27e4429a9..dff70ac3328 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h
>-@@ -8,17 +8,17 @@
>-  *  be found in the AUTHORS file in the root of the source tree.
>-  */
>- 
>--#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
>--#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
>-+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
>-+#define MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
>- 
>--#include "webrtc/common_types.h"
>-+#include "common_types.h"  // NOLINT(build/include)
>- 
>- namespace webrtc {
>- 
>- // Ratio allocation between temporal streams:
>- // Values as required for the VP8 codec (accumulating).
>- static const float
>--    kVp8LayerRateAlloction[kMaxTemporalStreams][kMaxTemporalStreams] = {
>-+    kVp8LayerRateAlloction[kMaxSimulcastStreams][kMaxTemporalStreams] = {
>-         {1.0f, 1.0f, 1.0f, 1.0f},  // 1 layer
>-         {0.6f, 1.0f, 1.0f, 1.0f},  // 2 layers {60%, 40%}
>-         {0.4f, 0.6f, 1.0f, 1.0f},  // 3 layers {40%, 20%, 40%}
>-@@ -26,4 +26,4 @@ static const float
>- };
>- 
>- }  // namespace webrtc
>--#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
>-+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h
>-index 938e1991269..1fab5f45a6f 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h
>-@@ -11,10 +11,10 @@
>- // This file contains codec dependent definitions that are needed in
>- // order to compile the WebRTC codebase, even if this codec is not used.
>- 
>--#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_GLOBALS_H_
>--#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_GLOBALS_H_
>-+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_GLOBALS_H_
>-+#define MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_GLOBALS_H_
>- 
>--#include "webrtc/modules/video_coding/codecs/interface/common_constants.h"
>-+#include "modules/video_coding/codecs/interface/common_constants.h"
>- 
>- namespace webrtc {
>- 
>-@@ -46,4 +46,4 @@ struct RTPVideoHeaderVP8 {
>- 
>- }  // namespace webrtc
>- 
>--#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_GLOBALS_H_
>-+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_GLOBALS_H_
>-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 a30679a36cf..782dc772fca 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
>-@@ -1,30 +1,32 @@
>- /* 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.
>--*/
>-+ *
>-+ *  Use of this source code is governed by a BSD-style license
>-+ *  that can be found in the LICENSE file in the root of the source
>-+ *  tree. An additional intellectual property rights grant can 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 "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h"
>-+#include "modules/video_coding/codecs/vp8/screenshare_layers.h"
>- 
>- #include <stdlib.h>
>- 
>- #include <algorithm>
>-+#include <memory>
>- 
>--#include "webrtc/base/checks.h"
>--#include "vpx/vpx_encoder.h"
>-+#include "modules/video_coding/include/video_codec_interface.h"
>-+#include "rtc_base/checks.h"
>-+#include "rtc_base/logging.h"
>-+#include "system_wrappers/include/clock.h"
>-+#include "system_wrappers/include/metrics.h"
>- #include "vpx/vp8cx.h"
>--#include "webrtc/modules/video_coding/include/video_codec_interface.h"
>--#include "webrtc/system_wrappers/include/clock.h"
>--#include "webrtc/system_wrappers/include/metrics.h"
>-+#include "vpx/vpx_encoder.h"
>- 
>- namespace webrtc {
>- 
>- static const int kOneSecond90Khz = 90000;
>--static const int kMinTimeBetweenSyncs = kOneSecond90Khz * 5;
>--static const int kMaxTimeBetweenSyncs = kOneSecond90Khz * 10;
>-+static const int kMinTimeBetweenSyncs = kOneSecond90Khz * 2;
>-+static const int kMaxTimeBetweenSyncs = kOneSecond90Khz * 4;
>- static const int kQpDeltaThresholdForSync = 8;
>- static const int kMinBitrateKbpsForQpBoost = 500;
>- 
>-@@ -54,6 +56,23 @@ webrtc::TemporalLayers* ScreenshareTemporalLayersFactory::Create(
>-   return tl;
>- }
>- 
>-+std::unique_ptr<webrtc::TemporalLayersChecker>
>-+ScreenshareTemporalLayersFactory::CreateChecker(
>-+    int simulcast_id,
>-+    int temporal_layers,
>-+    uint8_t initial_tl0_pic_idx) const {
>-+  webrtc::TemporalLayersChecker* tlc;
>-+  if (simulcast_id == 0) {
>-+    tlc =
>-+        new webrtc::TemporalLayersChecker(temporal_layers, initial_tl0_pic_idx);
>-+  } else {
>-+    TemporalLayersFactory rt_tl_factory;
>-+    return rt_tl_factory.CreateChecker(simulcast_id, temporal_layers,
>-+                                       initial_tl0_pic_idx);
>-+  }
>-+  return std::unique_ptr<webrtc::TemporalLayersChecker>(tlc);
>-+}
>-+
>- ScreenshareLayers::ScreenshareLayers(int num_temporal_layers,
>-                                      uint8_t initial_tl0_pic_idx,
>-                                      Clock* clock)
>-@@ -79,12 +98,6 @@ ScreenshareLayers::~ScreenshareLayers() {
>-   UpdateHistograms();
>- }
>- 
>--int ScreenshareLayers::GetTemporalLayerId(
>--    const TemporalLayers::FrameConfig& tl_config) const {
>--  // Codec does not use temporal layers for screenshare.
>--  return 0;
>--}
>--
>- uint8_t ScreenshareLayers::Tl0PicIdx() const {
>-   return tl0_pic_idx_;
>- }
>-@@ -96,7 +109,6 @@ TemporalLayers::FrameConfig ScreenshareLayers::UpdateLayerConfig(
>-     // TODO(pbos): Consider updating only last, and not all buffers.
>-     TemporalLayers::FrameConfig tl_config(
>-         kReferenceAndUpdate, kReferenceAndUpdate, kReferenceAndUpdate);
>--    tl_config.pattern_idx = static_cast<int>(TemporalLayerState::kTl1);
>-     return tl_config;
>-   }
>- 
>-@@ -152,11 +164,17 @@ TemporalLayers::FrameConfig ScreenshareLayers::UpdateLayerConfig(
>-       last_emitted_tl0_timestamp_ = unwrapped_timestamp;
>-       break;
>-     case 1:
>--      if (TimeToSync(unwrapped_timestamp)) {
>--        last_sync_timestamp_ = unwrapped_timestamp;
>--        layer_state = TemporalLayerState::kTl1Sync;
>-+      if (layers_[1].state != TemporalLayer::State::kDropped) {
>-+        if (TimeToSync(unwrapped_timestamp)) {
>-+          last_sync_timestamp_ = unwrapped_timestamp;
>-+          layer_state = TemporalLayerState::kTl1Sync;
>-+        } else {
>-+          layer_state = TemporalLayerState::kTl1;
>-+        }
>-       } else {
>--        layer_state = TemporalLayerState::kTl1;
>-+        layer_state = last_sync_timestamp_ == unwrapped_timestamp
>-+                          ? TemporalLayerState::kTl1Sync
>-+                          : TemporalLayerState::kTl1;
>-       }
>-       break;
>-     case -1:
>-@@ -178,21 +196,24 @@ TemporalLayers::FrameConfig ScreenshareLayers::UpdateLayerConfig(
>-       // TL0 only references and updates 'last'.
>-       tl_config =
>-           TemporalLayers::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.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.packetizer_temporal_idx = 1;
>-       break;
>-   }
>- 
>--  tl_config.pattern_idx = static_cast<int>(layer_state);
>-+  tl_config.layer_sync = layer_state == TemporalLayerState::kTl1Sync;
>-   return tl_config;
>- }
>- 
>-@@ -269,36 +290,24 @@ void ScreenshareLayers::PopulateCodecSpecific(
>-     const TemporalLayers::FrameConfig& tl_config,
>-     CodecSpecificInfoVP8* vp8_info,
>-     uint32_t timestamp) {
>--  int64_t unwrapped_timestamp = time_wrap_handler_.Unwrap(timestamp);
>-   if (number_of_temporal_layers_ == 1) {
>-     vp8_info->temporalIdx = kNoTemporalIdx;
>-     vp8_info->layerSync = false;
>-     vp8_info->tl0PicIdx = kNoTl0PicIdx;
>-   } else {
>--    TemporalLayerState layer_state =
>--        static_cast<TemporalLayerState>(tl_config.pattern_idx);
>--    switch (layer_state) {
>--      case TemporalLayerState::kDrop:
>--        RTC_NOTREACHED();
>--        break;
>--      case TemporalLayerState::kTl0:
>--        vp8_info->temporalIdx = 0;
>--        break;
>--      case TemporalLayerState::kTl1:
>--      case TemporalLayerState::kTl1Sync:
>--        vp8_info->temporalIdx = 1;
>--        break;
>--    }
>-+    int64_t unwrapped_timestamp = time_wrap_handler_.Unwrap(timestamp);
>-+    vp8_info->temporalIdx = tl_config.packetizer_temporal_idx;
>-+    vp8_info->layerSync = tl_config.layer_sync;
>-     if (frame_is_keyframe) {
>-       vp8_info->temporalIdx = 0;
>-       last_sync_timestamp_ = unwrapped_timestamp;
>-+      vp8_info->layerSync = true;
>-     } else if (last_base_layer_sync_ && vp8_info->temporalIdx != 0) {
>-       // Regardless of pattern the frame after a base layer sync will always
>-       // be a layer sync.
>-       last_sync_timestamp_ = unwrapped_timestamp;
>-+      vp8_info->layerSync = true;
>-     }
>--    vp8_info->layerSync = last_sync_timestamp_ != -1 &&
>--                          last_sync_timestamp_ == unwrapped_timestamp;
>-     if (vp8_info->temporalIdx == 0) {
>-       tl0_pic_idx_++;
>-     }
>-@@ -443,8 +452,9 @@ void ScreenshareLayers::UpdateHistograms() {
>-     int total_frames = stats_.num_tl0_frames_ + stats_.num_tl1_frames_;
>-     RTC_HISTOGRAM_COUNTS_10000(
>-         "WebRTC.Video.Screenshare.FramesPerDrop",
>--        (stats_.num_dropped_frames_ == 0 ? 0 : total_frames /
>--                                                   stats_.num_dropped_frames_));
>-+        (stats_.num_dropped_frames_ == 0
>-+             ? 0
>-+             : total_frames / stats_.num_dropped_frames_));
>-     RTC_HISTOGRAM_COUNTS_10000(
>-         "WebRTC.Video.Screenshare.FramesPerOvershoot",
>-         (stats_.num_overshoots_ == 0 ? 0
>-@@ -465,5 +475,4 @@ void ScreenshareLayers::UpdateHistograms() {
>-     }
>-   }
>- }
>--
>- }  // namespace webrtc
>-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 873846e8b22..81db90abe6b 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
>-@@ -1,21 +1,21 @@
>- /* 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 WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_SCREENSHARE_LAYERS_H_
>--#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_SCREENSHARE_LAYERS_H_
>-+ *
>-+ *  Use of this source code is governed by a BSD-style license
>-+ *  that can be found in the LICENSE file in the root of the source
>-+ *  tree. An additional intellectual property rights grant can 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_SCREENSHARE_LAYERS_H_
>-+#define MODULES_VIDEO_CODING_CODECS_VP8_SCREENSHARE_LAYERS_H_
>- 
>- #include <vector>
>- 
>--#include "webrtc/base/rate_statistics.h"
>--#include "webrtc/base/timeutils.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
>--#include "webrtc/modules/video_coding/utility/frame_dropper.h"
>--#include "webrtc/typedefs.h"
>-+#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>-+#include "modules/video_coding/utility/frame_dropper.h"
>-+#include "rtc_base/rate_statistics.h"
>-+#include "rtc_base/timeutils.h"
>-+#include "typedefs.h"  // NOLINT(build/include)
>- 
>- namespace webrtc {
>- 
>-@@ -54,9 +54,6 @@ class ScreenshareLayers : public TemporalLayers {
>- 
>-   void FrameEncoded(unsigned int size, int qp) override;
>- 
>--  int GetTemporalLayerId(
>--      const TemporalLayers::FrameConfig& tl_config) const override;
>--
>-   uint8_t Tl0PicIdx() const override;
>- 
>-  private:
>-@@ -127,4 +124,4 @@ class ScreenshareLayers : public TemporalLayers {
>- };
>- }  // namespace webrtc
>- 
>--#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_SCREENSHARE_LAYERS_H_
>-+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_SCREENSHARE_LAYERS_H_
>-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 9cac0b0a420..754235ab0b3 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
>-@@ -13,14 +13,14 @@
>- 
>- #include "vpx/vp8cx.h"
>- #include "vpx/vpx_encoder.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/vp8_impl.h"
>--#include "webrtc/modules/video_coding/include/video_codec_interface.h"
>--#include "webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h"
>--#include "webrtc/system_wrappers/include/clock.h"
>--#include "webrtc/system_wrappers/include/metrics.h"
>--#include "webrtc/system_wrappers/include/metrics_default.h"
>--#include "webrtc/test/gtest.h"
>-+#include "modules/video_coding/codecs/vp8/screenshare_layers.h"
>-+#include "modules/video_coding/codecs/vp8/vp8_impl.h"
>-+#include "modules/video_coding/include/video_codec_interface.h"
>-+#include "modules/video_coding/utility/mock/mock_frame_dropper.h"
>-+#include "system_wrappers/include/clock.h"
>-+#include "system_wrappers/include/metrics.h"
>-+#include "system_wrappers/include/metrics_default.h"
>-+#include "test/gtest.h"
>- 
>- using ::testing::_;
>- using ::testing::ElementsAre;
>-@@ -35,8 +35,8 @@ const int kDefaultQp = 54;
>- const int kDefaultTl0BitrateKbps = 200;
>- const int kDefaultTl1BitrateKbps = 2000;
>- const int kFrameRate = 5;
>--const int kSyncPeriodSeconds = 5;
>--const int kMaxSyncPeriodSeconds = 10;
>-+const int kSyncPeriodSeconds = 2;
>-+const int kMaxSyncPeriodSeconds = 4;
>- 
>- // Expected flags for corresponding temporal layers.
>- const int kTl0Flags = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
>-@@ -71,6 +71,10 @@ class ScreenshareLayerTest : public ::testing::Test {
>- 
>-   int ConfigureFrame(bool key_frame) {
>-     tl_config_ = layers_->UpdateLayerConfig(timestamp_);
>-+    EXPECT_EQ(0, tl_config_.encoder_layer_id)
>-+        << "ScreenshareLayers always encodes using the bitrate allocator for "
>-+           "layer 0, but may reference different buffers and packetize "
>-+           "differently.";
>-     if (tl_config_.drop_frame) {
>-       return -1;
>-     }
>-@@ -126,18 +130,29 @@ class ScreenshareLayerTest : public ::testing::Test {
>- 
>-   // Adds frames until we get one in the specified temporal layer. The last
>-   // FrameEncoded() call will be omitted and needs to be done by the caller.
>--  void SkipUntilTl(int layer) {
>--    for (int i = 0; i < 5; ++i) {
>--      ConfigureFrame(false);
>-+  // Returns the flags for the last frame.
>-+  int SkipUntilTl(int layer) {
>-+    return SkipUntilTlAndSync(layer, rtc::nullopt);
>-+  }
>-+
>-+  // Same as SkipUntilTl, but also waits until the sync bit condition is met.
>-+  int SkipUntilTlAndSync(int layer, rtc::Optional<bool> sync) {
>-+    int flags = 0;
>-+    const int kMaxFramesToSkip =
>-+        1 + (sync.value_or(false) ? kMaxSyncPeriodSeconds : 1) * kFrameRate;
>-+    for (int i = 0; i < kMaxFramesToSkip; ++i) {
>-+      flags = ConfigureFrame(false);
>-       timestamp_ += kTimestampDelta5Fps;
>--      if (vp8_info_.temporalIdx != layer) {
>-+      if (vp8_info_.temporalIdx != layer ||
>-+          (sync && *sync != vp8_info_.layerSync)) {
>-         layers_->FrameEncoded(frame_size_, kDefaultQp);
>-       } else {
>--        // Found frame form sought layer.
>--        return;
>-+        // Found frame from sought after layer.
>-+        return flags;
>-       }
>-     }
>-     ADD_FAILURE() << "Did not get a frame of TL" << layer << " in time.";
>-+    return -1;
>-   }
>- 
>-   int min_qp_;
>-@@ -562,4 +577,31 @@ TEST_F(ScreenshareLayerTest, RespectsConfiguredFramerate) {
>-   // Allow for some rounding errors in the measurements.
>-   EXPECT_NEAR(num_discarded_frames, num_input_frames / 2, 2);
>- }
>-+
>-+TEST_F(ScreenshareLayerTest, 2LayersSyncAtOvershootDrop) {
>-+  // Run grace period so we have existing frames in both TL0 and Tl1.
>-+  EXPECT_TRUE(RunGracePeriod());
>-+
>-+  // Move ahead until we have a sync frame in TL1.
>-+  EXPECT_EQ(kTl1SyncFlags, SkipUntilTlAndSync(1, true));
>-+  ASSERT_TRUE(vp8_info_.layerSync);
>-+
>-+  // Simulate overshoot of this frame.
>-+  layers_->FrameEncoded(0, -1);
>-+
>-+  // Reencode, frame config, flags and codec specific info should remain the
>-+  // same as for the dropped frame.
>-+  timestamp_ -= kTimestampDelta5Fps;  // Undo last timestamp increment.
>-+  TemporalLayers::FrameConfig new_tl_config =
>-+      layers_->UpdateLayerConfig(timestamp_);
>-+  EXPECT_EQ(tl_config_, new_tl_config);
>-+
>-+  config_updated_ = layers_->UpdateConfiguration(&cfg_);
>-+  EXPECT_EQ(kTl1SyncFlags, VP8EncoderImpl::EncodeFlags(tl_config_));
>-+
>-+  CodecSpecificInfoVP8 new_vp8_info;
>-+  layers_->PopulateCodecSpecific(false, tl_config_, &new_vp8_info, timestamp_);
>-+  EXPECT_TRUE(new_vp8_info.layerSync);
>-+}
>-+
>- }  // namespace webrtc
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.cc
>-index 39c43c9b6d0..b2b3334815f 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.cc
>-@@ -8,14 +8,14 @@
>-  *  be found in the AUTHORS file in the root of the source tree.
>-  */
>- 
>--#include "webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
>-+#include "modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
>- 
>- #include <algorithm>
>- #include <memory>
>- #include <vector>
>- #include <utility>
>- 
>--#include "webrtc/base/checks.h"
>-+#include "rtc_base/checks.h"
>- 
>- namespace webrtc {
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.h
>-index e558290e6b4..929abba4132 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.h
>-@@ -8,18 +8,18 @@
>-  *  be found in the AUTHORS file in the root of the source tree.
>-  */
>- 
>--#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_RATE_ALLOCATOR_H_
>--#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_RATE_ALLOCATOR_H_
>-+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_RATE_ALLOCATOR_H_
>-+#define MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_RATE_ALLOCATOR_H_
>- 
>- #include <stdint.h>
>- 
>- #include <map>
>- #include <memory>
>- 
>--#include "webrtc/api/video_codecs/video_encoder.h"
>--#include "webrtc/base/constructormagic.h"
>--#include "webrtc/common_video/include/video_bitrate_allocator.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
>-+#include "api/video_codecs/video_encoder.h"
>-+#include "common_video/include/video_bitrate_allocator.h"
>-+#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>-+#include "rtc_base/constructormagic.h"
>- 
>- namespace webrtc {
>- 
>-@@ -48,4 +48,4 @@ class SimulcastRateAllocator : public VideoBitrateAllocator,
>- 
>- }  // namespace webrtc
>- 
>--#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_RATE_ALLOCATOR_H_
>-+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_RATE_ALLOCATOR_H_
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc
>-index 9d919cdd5bd..b1dd794f291 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc
>-@@ -8,19 +8,19 @@
>-  *  be found in the AUTHORS file in the root of the source tree.
>-  */
>- 
>--#include "webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h"
>-+#include "modules/video_coding/codecs/vp8/simulcast_test_utility.h"
>- 
>- namespace webrtc {
>- namespace testing {
>- 
>- class TestVp8Impl : public TestVp8Simulcast {
>-- public:
>--  TestVp8Impl()
>--      : TestVp8Simulcast(VP8Encoder::Create(), VP8Decoder::Create()) {}
>--
>-  protected:
>--  virtual void SetUp() { TestVp8Simulcast::SetUp(); }
>--  virtual void TearDown() { TestVp8Simulcast::TearDown(); }
>-+  std::unique_ptr<VP8Encoder> CreateEncoder() override {
>-+    return VP8Encoder::Create();
>-+  }
>-+  std::unique_ptr<VP8Decoder> CreateDecoder() override {
>-+    return VP8Decoder::Create();
>-+  }
>- };
>- 
>- TEST_F(TestVp8Impl, TestKeyFrameRequestsOnAllStreams) {
>-@@ -71,10 +71,6 @@ TEST_F(TestVp8Impl, TestSaptioTemporalLayers333PatternEncoder) {
>-   TestVp8Simulcast::TestSaptioTemporalLayers333PatternEncoder();
>- }
>- 
>--TEST_F(TestVp8Impl, TestSpatioTemporalLayers321PatternEncoder) {
>--  TestVp8Simulcast::TestSpatioTemporalLayers321PatternEncoder();
>--}
>--
>- TEST_F(TestVp8Impl, TestStrideEncodeDecode) {
>-   TestVp8Simulcast::TestStrideEncodeDecode();
>- }
>-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 f0561076a0e..dad718cd0f3 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
>-@@ -1,21 +1,21 @@
>- /* 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.
>--*/
>-+ *
>-+ *  Use of this source code is governed by a BSD-style license
>-+ *  that can be found in the LICENSE file in the root of the source
>-+ *  tree. An additional intellectual property rights grant can 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 defines the interface for doing temporal layers with VP8.
>--*/
>--#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
>--#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
>-+ * This file defines the interface for doing temporal layers with VP8.
>-+ */
>-+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
>-+#define MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
>- 
>- #include <vector>
>-+#include <memory>
>- 
>--#include "webrtc/typedefs.h"
>--
>-+#include "typedefs.h"  // NOLINT(build/include)
>- struct vpx_codec_enc_cfg;
>- typedef struct vpx_codec_enc_cfg vpx_codec_enc_cfg_t;
>- 
>-@@ -47,13 +47,34 @@ class TemporalLayers {
>-     BufferFlags golden_buffer_flags;
>-     BufferFlags arf_buffer_flags;
>- 
>--    // TODO(pbos): Consider breaking these out of here and returning only a
>--    // pattern index that needs to be returned to fill CodecSpecificInfoVP8 or
>--    // EncodeFlags.
>-+    // 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;
>- 
>--    int pattern_idx;
>-+    bool 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;
>-+    }
>-+    bool operator!=(const FrameConfig& o) const { return !(*this == o); }
>- 
>-    private:
>-     FrameConfig(BufferFlags last,
>-@@ -91,18 +112,27 @@ class TemporalLayers {
>-   // Returns the current tl0_pic_idx, so it can be reused in future
>-   // instantiations.
>-   virtual uint8_t Tl0PicIdx() const = 0;
>--  virtual int GetTemporalLayerId(
>--      const TemporalLayers::FrameConfig& tl_config) const = 0;
>- };
>- 
>- class TemporalLayersListener;
>-+class TemporalLayersChecker;
>-+
>- class TemporalLayersFactory {
>-  public:
>-   TemporalLayersFactory() : listener_(nullptr) {}
>-   virtual ~TemporalLayersFactory() {}
>--  virtual TemporalLayers* Create(int /* simulcast_id */,
>--                                 int /* temporal_layers */,
>--                                 uint8_t /* initial_tl0_pic_idx */) const { return nullptr; }
>-+  virtual TemporalLayers* Create(int simulcast_id,
>-+                                 int temporal_layers,
>-+                                 uint8_t initial_tl0_pic_idx) const;
>-+
>-+  // Creates helper class which performs online checks of a correctness of
>-+  // temporal layers dependencies returned by TemporalLayers class created in
>-+  // the same factory.
>-+  virtual std::unique_ptr<TemporalLayersChecker> CreateChecker(
>-+      int simulcast_id,
>-+      int temporal_layers,
>-+      uint8_t initial_tl0_pic_idx) const;
>-+
>-   void SetListener(TemporalLayersListener* listener);
>- 
>-  protected:
>-@@ -117,6 +147,14 @@ class ScreenshareTemporalLayersFactory : public webrtc::TemporalLayersFactory {
>-   webrtc::TemporalLayers* Create(int simulcast_id,
>-                                  int num_temporal_layers,
>-                                  uint8_t initial_tl0_pic_idx) const override;
>-+
>-+  // Creates helper class which performs online checks of a correctness of
>-+  // temporal layers dependencies returned by TemporalLayers class created in
>-+  // the same factory.
>-+  std::unique_ptr<webrtc::TemporalLayersChecker> CreateChecker(
>-+      int simulcast_id,
>-+      int temporal_layers,
>-+      uint8_t initial_tl0_pic_idx) const override;
>- };
>- 
>- class TemporalLayersListener {
>-@@ -128,5 +166,40 @@ class TemporalLayersListener {
>-                                        TemporalLayers* layers) = 0;
>- };
>- 
>-+// Used only inside RTC_DCHECK(). It checks correctness of temporal layers
>-+// dependencies and sync bits. The only method of this class is called after
>-+// each UpdateLayersConfig() of a corresponding TemporalLayers class.
>-+class TemporalLayersChecker {
>-+ public:
>-+  TemporalLayersChecker(int num_temporal_layers, uint8_t initial_tl0_pic_idx);
>-+  virtual ~TemporalLayersChecker() {}
>-+
>-+  virtual bool CheckTemporalConfig(
>-+      bool frame_is_keyframe,
>-+      const TemporalLayers::FrameConfig& frame_config);
>-+
>-+ private:
>-+  struct BufferState {
>-+    BufferState() : is_keyframe(true), temporal_layer(0), sequence_number(0) {}
>-+    bool is_keyframe;
>-+    uint8_t temporal_layer;
>-+    uint32_t sequence_number;
>-+  };
>-+  bool CheckAndUpdateBufferState(BufferState* state,
>-+                                 bool* need_sync,
>-+                                 bool frame_is_keyframe,
>-+                                 uint8_t temporal_layer,
>-+                                 webrtc::TemporalLayers::BufferFlags flags,
>-+                                 uint32_t sequence_number,
>-+                                 uint32_t* lowest_sequence_referenced);
>-+  BufferState last_;
>-+  BufferState arf_;
>-+  BufferState golden_;
>-+  int num_temporal_layers_;
>-+  uint32_t sequence_number_;
>-+  uint32_t last_sync_sequence_number_;
>-+  uint32_t last_tl0_sequence_number_;
>-+};
>-+
>- }  // namespace webrtc
>--#endif  // WEBRTC_MODULES_VIDEO_CODING_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/test/vp8_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
>-index a3db4fdad69..5d6abec76d8 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
>-@@ -12,99 +12,88 @@
>- 
>- #include <memory>
>- 
>--#include "webrtc/api/video/i420_buffer.h"
>--#include "webrtc/base/checks.h"
>--#include "webrtc/base/optional.h"
>--#include "webrtc/base/timeutils.h"
>--#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
>--#include "webrtc/modules/video_coding/codecs/test/video_codec_test.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
>--#include "webrtc/test/frame_utils.h"
>--#include "webrtc/test/gtest.h"
>--#include "webrtc/test/testsupport/fileutils.h"
>-+#include "api/optional.h"
>-+#include "api/video/i420_buffer.h"
>-+#include "common_video/libyuv/include/webrtc_libyuv.h"
>-+#include "modules/video_coding/codecs/test/video_codec_test.h"
>-+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>-+#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>-+#include "rtc_base/checks.h"
>-+#include "rtc_base/timeutils.h"
>-+#include "test/field_trial.h"
>-+#include "test/frame_utils.h"
>-+#include "test/gtest.h"
>-+#include "test/testsupport/fileutils.h"
>-+#include "test/video_codec_settings.h"
>- 
>- namespace webrtc {
>- 
>- namespace {
>-+constexpr uint32_t kInitialTimestampRtp = 123;
>-+constexpr int64_t kTestNtpTimeMs = 456;
>-+constexpr int64_t kInitialTimestampMs = 789;
>-+constexpr uint32_t kTimestampIncrement = 3000;
>-+constexpr int kNumCores = 1;
>-+constexpr size_t kMaxPayloadSize = 1440;
>-+constexpr int kDefaultMinPixelsPerFrame = 320 * 180;
>-+constexpr int kWidth = 172;
>-+constexpr int kHeight = 144;
>- 
>- void Calc16ByteAlignedStride(int width, int* stride_y, int* stride_uv) {
>-   *stride_y = 16 * ((width + 15) / 16);
>-   *stride_uv = 16 * ((width + 31) / 32);
>- }
>--
>--enum { kMaxWaitEncTimeMs = 100 };
>--enum { kMaxWaitDecTimeMs = 25 };
>--
>--constexpr uint32_t kTestTimestamp = 123;
>--constexpr int64_t kTestNtpTimeMs = 456;
>--constexpr uint32_t kTimestampIncrementPerFrame = 3000;
>--
>- }  // namespace
>- 
>--// TODO(mikhal): Replace these with mocks.
>--class Vp8UnitTestEncodeCompleteCallback : public webrtc::EncodedImageCallback {
>-+class EncodedImageCallbackTestImpl : public webrtc::EncodedImageCallback {
>-  public:
>--  Vp8UnitTestEncodeCompleteCallback(EncodedImage* frame,
>--                                    CodecSpecificInfo* codec_specific_info,
>--                                    unsigned int decoderSpecificSize,
>--                                    void* decoderSpecificInfo)
>--      : encoded_frame_(frame),
>--        codec_specific_info_(codec_specific_info),
>--        encode_complete_(false) {}
>--
>--  Result OnEncodedImage(const EncodedImage& encoded_frame_,
>-+  Result OnEncodedImage(const EncodedImage& encoded_frame,
>-                         const CodecSpecificInfo* codec_specific_info,
>--                        const RTPFragmentationHeader* fragmentation) override;
>--  bool EncodeComplete();
>-+                        const RTPFragmentationHeader* fragmentation) override {
>-+    EXPECT_GT(encoded_frame._length, 0u);
>-+    VerifyQpParser(encoded_frame);
>- 
>-- private:
>--  EncodedImage* const encoded_frame_;
>--  CodecSpecificInfo* const codec_specific_info_;
>--  std::unique_ptr<uint8_t[]> frame_buffer_;
>--  bool encode_complete_;
>--};
>-+    if (encoded_frame_._size != encoded_frame._size) {
>-+      delete[] encoded_frame_._buffer;
>-+      frame_buffer_.reset(new uint8_t[encoded_frame._size]);
>-+    }
>-+    RTC_DCHECK(frame_buffer_);
>-+    memcpy(frame_buffer_.get(), encoded_frame._buffer, encoded_frame._length);
>-+    encoded_frame_ = encoded_frame;
>-+    encoded_frame_._buffer = frame_buffer_.get();
>-+
>-+    // Skip |codec_name|, to avoid allocating.
>-+    EXPECT_STREQ("libvpx", codec_specific_info->codec_name);
>-+    EXPECT_EQ(kVideoCodecVP8, codec_specific_info->codecType);
>-+    EXPECT_EQ(0u, codec_specific_info->codecSpecific.VP8.simulcastIdx);
>-+    codec_specific_info_.codecType = codec_specific_info->codecType;
>-+    codec_specific_info_.codecSpecific = codec_specific_info->codecSpecific;
>-+    complete_ = true;
>-+    return Result(Result::OK, 0);
>-+  }
>- 
>--webrtc::EncodedImageCallback::Result
>--Vp8UnitTestEncodeCompleteCallback::OnEncodedImage(
>--    const EncodedImage& encoded_frame,
>--    const CodecSpecificInfo* codec_specific_info,
>--    const RTPFragmentationHeader* fragmentation) {
>--  if (encoded_frame_->_size < encoded_frame._length) {
>--    delete[] encoded_frame_->_buffer;
>--    frame_buffer_.reset(new uint8_t[encoded_frame._length]);
>--    encoded_frame_->_buffer = frame_buffer_.get();
>--    encoded_frame_->_size = encoded_frame._length;
>-+  void VerifyQpParser(const EncodedImage& encoded_frame) const {
>-+    int qp;
>-+    ASSERT_TRUE(vp8::GetQp(encoded_frame._buffer, encoded_frame._length, &qp));
>-+    EXPECT_EQ(encoded_frame.qp_, qp) << "Encoder QP != parsed bitstream QP.";
>-   }
>--  memcpy(encoded_frame_->_buffer, encoded_frame._buffer, encoded_frame._length);
>--  encoded_frame_->_length = encoded_frame._length;
>--  encoded_frame_->_encodedWidth = encoded_frame._encodedWidth;
>--  encoded_frame_->_encodedHeight = encoded_frame._encodedHeight;
>--  encoded_frame_->_timeStamp = encoded_frame._timeStamp;
>--  encoded_frame_->_frameType = encoded_frame._frameType;
>--  encoded_frame_->_completeFrame = encoded_frame._completeFrame;
>--  encoded_frame_->rotation_ = encoded_frame.rotation_;
>--  encoded_frame_->qp_ = encoded_frame.qp_;
>--  codec_specific_info_->codecType = codec_specific_info->codecType;
>--  // Skip |codec_name|, to avoid allocating.
>--  codec_specific_info_->codecSpecific = codec_specific_info->codecSpecific;
>--  encode_complete_ = true;
>--  return Result(Result::OK, 0);
>--}
>- 
>--bool Vp8UnitTestEncodeCompleteCallback::EncodeComplete() {
>--  if (encode_complete_) {
>--    encode_complete_ = false;
>--    return true;
>-+  bool EncodeComplete() {
>-+    if (complete_) {
>-+      complete_ = false;
>-+      return true;
>-+    }
>-+    return false;
>-   }
>--  return false;
>--}
>- 
>--class Vp8UnitTestDecodeCompleteCallback : public webrtc::DecodedImageCallback {
>-+  EncodedImage encoded_frame_;
>-+  CodecSpecificInfo codec_specific_info_;
>-+  std::unique_ptr<uint8_t[]> frame_buffer_;
>-+  bool complete_ = false;
>-+};
>-+
>-+class DecodedImageCallbackTestImpl : public webrtc::DecodedImageCallback {
>-  public:
>--  explicit Vp8UnitTestDecodeCompleteCallback(rtc::Optional<VideoFrame>* frame,
>--                                             rtc::Optional<uint8_t>* qp)
>--      : decoded_frame_(frame), decoded_qp_(qp), decode_complete(false) {}
>-   int32_t Decoded(VideoFrame& frame) override {
>-     RTC_NOTREACHED();
>-     return -1;
>-@@ -115,163 +104,158 @@ class Vp8UnitTestDecodeCompleteCallback : public webrtc::DecodedImageCallback {
>-   }
>-   void Decoded(VideoFrame& frame,
>-                rtc::Optional<int32_t> decode_time_ms,
>--               rtc::Optional<uint8_t> qp) override;
>--  bool DecodeComplete();
>--
>-- private:
>--  rtc::Optional<VideoFrame>* decoded_frame_;
>--  rtc::Optional<uint8_t>* decoded_qp_;
>--  bool decode_complete;
>--};
>-+               rtc::Optional<uint8_t> qp) override {
>-+    EXPECT_GT(frame.width(), 0);
>-+    EXPECT_GT(frame.height(), 0);
>-+    EXPECT_TRUE(qp);
>-+    frame_ = frame;
>-+    qp_ = qp;
>-+    complete_ = true;
>-+  }
>- 
>--bool Vp8UnitTestDecodeCompleteCallback::DecodeComplete() {
>--  if (decode_complete) {
>--    decode_complete = false;
>--    return true;
>-+  bool DecodeComplete() {
>-+    if (complete_) {
>-+      complete_ = false;
>-+      return true;
>-+    }
>-+    return false;
>-   }
>--  return false;
>--}
>- 
>--void Vp8UnitTestDecodeCompleteCallback::Decoded(
>--    VideoFrame& frame,
>--    rtc::Optional<int32_t> decode_time_ms,
>--    rtc::Optional<uint8_t> qp) {
>--  *decoded_frame_ = rtc::Optional<VideoFrame>(frame);
>--  *decoded_qp_ = qp;
>--  decode_complete = true;
>--}
>-+  rtc::Optional<VideoFrame> frame_;
>-+  rtc::Optional<uint8_t> qp_;
>-+  bool complete_ = false;
>-+};
>- 
>- class TestVp8Impl : public ::testing::Test {
>-+ public:
>-+  TestVp8Impl() : TestVp8Impl("") {}
>-+  explicit TestVp8Impl(const std::string& field_trials)
>-+      : override_field_trials_(field_trials),
>-+        encoder_(VP8Encoder::Create()),
>-+        decoder_(VP8Decoder::Create()) {}
>-+  virtual ~TestVp8Impl() {}
>-+
>-  protected:
>-   virtual void SetUp() {
>--    encoder_.reset(VP8Encoder::Create());
>--    decoder_.reset(VP8Decoder::Create());
>--    memset(&codec_settings_, 0, sizeof(codec_settings_));
>--    encode_complete_callback_.reset(new Vp8UnitTestEncodeCompleteCallback(
>--        &encoded_frame_, &codec_specific_info_, 0, nullptr));
>--    decode_complete_callback_.reset(
>--        new Vp8UnitTestDecodeCompleteCallback(&decoded_frame_, &decoded_qp_));
>--    encoder_->RegisterEncodeCompleteCallback(encode_complete_callback_.get());
>--    decoder_->RegisterDecodeCompleteCallback(decode_complete_callback_.get());
>-+    encoder_->RegisterEncodeCompleteCallback(&encoded_cb_);
>-+    decoder_->RegisterDecodeCompleteCallback(&decoded_cb_);
>-+    SetupCodecSettings();
>-+    SetupInputFrame();
>-+  }
>-+
>-+  void SetupInputFrame() {
>-     // Using a QCIF image (aligned stride (u,v planes) > width).
>-     // Processing only one frame.
>--    source_file_ = fopen(test::ResourcePath("paris_qcif", "yuv").c_str(), "rb");
>--    ASSERT_TRUE(source_file_ != nullptr);
>-+    FILE* file = fopen(test::ResourcePath("paris_qcif", "yuv").c_str(), "rb");
>-+    ASSERT_TRUE(file != nullptr);
>-     rtc::scoped_refptr<I420BufferInterface> compact_buffer(
>--        test::ReadI420Buffer(kWidth, kHeight, source_file_));
>-+        test::ReadI420Buffer(kWidth, kHeight, file));
>-     ASSERT_TRUE(compact_buffer);
>--    codec_settings_.width = kWidth;
>--    codec_settings_.height = kHeight;
>--    const int kFramerate = 30;
>--    codec_settings_.maxFramerate = kFramerate;
>-+
>-     // Setting aligned stride values.
>-     int stride_uv;
>-     int stride_y;
>--    Calc16ByteAlignedStride(codec_settings_.width, &stride_y, &stride_uv);
>-+    Calc16ByteAlignedStride(kWidth, &stride_y, &stride_uv);
>-     EXPECT_EQ(stride_y, 176);
>-     EXPECT_EQ(stride_uv, 96);
>--
>-     rtc::scoped_refptr<I420Buffer> stride_buffer(
>-         I420Buffer::Create(kWidth, kHeight, stride_y, stride_uv, stride_uv));
>- 
>-     // No scaling in our case, just a copy, to add stride to the image.
>-     stride_buffer->ScaleFrom(*compact_buffer);
>- 
>--    input_frame_.reset(
>--        new VideoFrame(stride_buffer, kVideoRotation_0, 0));
>--    input_frame_->set_timestamp(kTestTimestamp);
>-+    input_frame_.reset(new VideoFrame(stride_buffer, kInitialTimestampRtp,
>-+                                      kInitialTimestampMs, kVideoRotation_0));
>-+    fclose(file);
>-   }
>- 
>--  void SetUpEncodeDecode() {
>--    codec_settings_.startBitrate = 300;
>-+  void SetupCodecSettings() {
>-+    webrtc::test::CodecSettings(kVideoCodecVP8, &codec_settings_);
>-     codec_settings_.maxBitrate = 4000;
>--    codec_settings_.qpMax = 56;
>-+    codec_settings_.width = kWidth;
>-+    codec_settings_.height = kHeight;
>-     codec_settings_.VP8()->denoisingOn = true;
>-+    codec_settings_.VP8()->frameDroppingOn = false;
>-+    codec_settings_.VP8()->automaticResizeOn = false;
>-+    codec_settings_.VP8()->complexity = kComplexityNormal;
>-     codec_settings_.VP8()->tl_factory = &tl_factory_;
>--    codec_settings_.VP8()->numberOfTemporalLayers = 1;
>--
>--    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--              encoder_->InitEncode(&codec_settings_, 1, 1440));
>--    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->InitDecode(&codec_settings_, 1));
>-   }
>- 
>--  size_t WaitForEncodedFrame() const {
>--    int64_t startTime = rtc::TimeMillis();
>--    while (rtc::TimeMillis() - startTime < kMaxWaitEncTimeMs) {
>--      if (encode_complete_callback_->EncodeComplete()) {
>--        return encoded_frame_._length;
>--      }
>--    }
>--    return 0;
>-+  void InitEncodeDecode() {
>-+    EXPECT_EQ(
>-+        WEBRTC_VIDEO_CODEC_OK,
>-+        encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-+              decoder_->InitDecode(&codec_settings_, kNumCores));
>-   }
>- 
>--  size_t WaitForDecodedFrame() const {
>--    int64_t startTime = rtc::TimeMillis();
>--    while (rtc::TimeMillis() - startTime < kMaxWaitDecTimeMs) {
>--      if (decode_complete_callback_->DecodeComplete()) {
>--        return CalcBufferSize(VideoType::kI420, decoded_frame_->width(),
>--                              decoded_frame_->height());
>--      }
>--    }
>--    return 0;
>-+  void EncodeFrame() {
>-+    EXPECT_FALSE(encoded_cb_.EncodeComplete());
>-+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-+              encoder_->Encode(*input_frame_, nullptr, nullptr));
>-+    EXPECT_TRUE(encoded_cb_.EncodeComplete());
>-   }
>- 
>-   void ExpectFrameWith(int16_t picture_id,
>-                        int tl0_pic_idx,
>-                        uint8_t temporal_idx) {
>--    ASSERT_TRUE(WaitForEncodedFrame());
>--    EXPECT_EQ(picture_id, codec_specific_info_.codecSpecific.VP8.pictureId);
>--    EXPECT_EQ(tl0_pic_idx, codec_specific_info_.codecSpecific.VP8.tl0PicIdx);
>--    EXPECT_EQ(temporal_idx, codec_specific_info_.codecSpecific.VP8.temporalIdx);
>-+    EXPECT_EQ(picture_id % (1 << 15),
>-+              encoded_cb_.codec_specific_info_.codecSpecific.VP8.pictureId);
>-+    EXPECT_EQ(tl0_pic_idx % (1 << 8),
>-+              encoded_cb_.codec_specific_info_.codecSpecific.VP8.tl0PicIdx);
>-+    EXPECT_EQ(temporal_idx,
>-+              encoded_cb_.codec_specific_info_.codecSpecific.VP8.temporalIdx);
>-   }
>- 
>--  const int kWidth = 172;
>--  const int kHeight = 144;
>--
>--  std::unique_ptr<Vp8UnitTestEncodeCompleteCallback> encode_complete_callback_;
>--  std::unique_ptr<Vp8UnitTestDecodeCompleteCallback> decode_complete_callback_;
>--  std::unique_ptr<uint8_t[]> source_buffer_;
>--  FILE* source_file_;
>-+  test::ScopedFieldTrials override_field_trials_;
>-+  EncodedImageCallbackTestImpl encoded_cb_;
>-+  DecodedImageCallbackTestImpl decoded_cb_;
>-   std::unique_ptr<VideoFrame> input_frame_;
>--  std::unique_ptr<VideoEncoder> encoder_;
>--  std::unique_ptr<VideoDecoder> decoder_;
>--  EncodedImage encoded_frame_;
>--  CodecSpecificInfo codec_specific_info_;
>--  rtc::Optional<VideoFrame> decoded_frame_;
>--  rtc::Optional<uint8_t> decoded_qp_;
>-+  const std::unique_ptr<VideoEncoder> encoder_;
>-+  const std::unique_ptr<VideoDecoder> decoder_;
>-   VideoCodec codec_settings_;
>-   TemporalLayersFactory tl_factory_;
>- };
>- 
>--TEST_F(TestVp8Impl, EncoderParameterTest) {
>--  strncpy(codec_settings_.plName, "VP8", 31);
>--  codec_settings_.plType = 126;
>--  codec_settings_.maxBitrate = 0;
>--  codec_settings_.minBitrate = 0;
>--  codec_settings_.width = 1440;
>--  codec_settings_.height = 1080;
>--  codec_settings_.maxFramerate = 30;
>--  codec_settings_.startBitrate = 300;
>--  codec_settings_.qpMax = 56;
>--  codec_settings_.VP8()->complexity = kComplexityNormal;
>--  codec_settings_.VP8()->numberOfTemporalLayers = 1;
>--  codec_settings_.VP8()->tl_factory = &tl_factory_;
>--  // Calls before InitEncode().
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>--  int bit_rate = 300;
>-+TEST_F(TestVp8Impl, SetRateAllocation) {
>-+  const int kBitrateBps = 300000;
>-   BitrateAllocation bitrate_allocation;
>--  bitrate_allocation.SetBitrate(0, 0, bit_rate * 1000);
>-+  bitrate_allocation.SetBitrate(0, 0, kBitrateBps);
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_UNINITIALIZED,
>-             encoder_->SetRateAllocation(bitrate_allocation,
>-                                         codec_settings_.maxFramerate));
>-+  InitEncodeDecode();
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-+            encoder_->SetRateAllocation(bitrate_allocation,
>-+                                        codec_settings_.maxFramerate));
>-+}
>- 
>-+TEST_F(TestVp8Impl, EncodeFrameAndRelease) {
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->InitEncode(&codec_settings_, 1, 1440));
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-+  EncodeFrame();
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_UNINITIALIZED,
>-+            encoder_->Encode(*input_frame_, nullptr, nullptr));
>-+}
>- 
>--  // Decoder parameter tests.
>--  // Calls before InitDecode().
>-+TEST_F(TestVp8Impl, InitDecode) {
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Release());
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->InitDecode(&codec_settings_, 1));
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-+            decoder_->InitDecode(&codec_settings_, kNumCores));
>-+}
>-+
>-+TEST_F(TestVp8Impl, OnEncodedImageReportsInfo) {
>-+  InitEncodeDecode();
>-+  EncodeFrame();
>-+  EXPECT_EQ(kInitialTimestampRtp, encoded_cb_.encoded_frame_._timeStamp);
>-+  EXPECT_EQ(kInitialTimestampMs, encoded_cb_.encoded_frame_.capture_time_ms_);
>-+  EXPECT_EQ(kWidth, static_cast<int>(encoded_cb_.encoded_frame_._encodedWidth));
>-+  EXPECT_EQ(kHeight,
>-+            static_cast<int>(encoded_cb_.encoded_frame_._encodedHeight));
>-+  EXPECT_EQ(-1,  // Disabled for single stream.
>-+            encoded_cb_.encoded_frame_.adapt_reason_.bw_resolutions_disabled);
>- }
>- 
>- // We only test the encoder here, since the decoded frame rotation is set based
>-@@ -279,46 +263,84 @@ TEST_F(TestVp8Impl, EncoderParameterTest) {
>- // TODO(brandtr): Consider passing through the rotation flag through the decoder
>- // in the same way as done in the encoder.
>- TEST_F(TestVp8Impl, EncodedRotationEqualsInputRotation) {
>--  SetUpEncodeDecode();
>--
>-+  InitEncodeDecode();
>-   input_frame_->set_rotation(kVideoRotation_0);
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  WaitForEncodedFrame();
>--  EXPECT_EQ(kVideoRotation_0, encoded_frame_.rotation_);
>-+  EncodeFrame();
>-+  EXPECT_EQ(kVideoRotation_0, encoded_cb_.encoded_frame_.rotation_);
>- 
>-   input_frame_->set_rotation(kVideoRotation_90);
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  WaitForEncodedFrame();
>--  EXPECT_EQ(kVideoRotation_90, encoded_frame_.rotation_);
>-+  EncodeFrame();
>-+  EXPECT_EQ(kVideoRotation_90, encoded_cb_.encoded_frame_.rotation_);
>- }
>- 
>- TEST_F(TestVp8Impl, DecodedQpEqualsEncodedQp) {
>--  SetUpEncodeDecode();
>--  encoder_->Encode(*input_frame_, nullptr, nullptr);
>--  EXPECT_GT(WaitForEncodedFrame(), 0u);
>-+  InitEncodeDecode();
>-+  EncodeFrame();
>-   // First frame should be a key frame.
>--  encoded_frame_._frameType = kVideoFrameKey;
>-+  encoded_cb_.encoded_frame_._frameType = kVideoFrameKey;
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            decoder_->Decode(encoded_frame_, false, nullptr));
>--  EXPECT_GT(WaitForDecodedFrame(), 0u);
>--  ASSERT_TRUE(decoded_frame_);
>--  EXPECT_GT(I420PSNR(input_frame_.get(), &*decoded_frame_), 36);
>--  ASSERT_TRUE(decoded_qp_);
>--  EXPECT_EQ(encoded_frame_.qp_, *decoded_qp_);
>-+            decoder_->Decode(encoded_cb_.encoded_frame_, false, nullptr));
>-+  EXPECT_TRUE(decoded_cb_.DecodeComplete());
>-+  EXPECT_GT(I420PSNR(input_frame_.get(), &*decoded_cb_.frame_), 36);
>-+  EXPECT_EQ(encoded_cb_.encoded_frame_.qp_, *decoded_cb_.qp_);
>- }
>- 
>--TEST_F(TestVp8Impl, ParserQpEqualsEncodedQp) {
>--  SetUpEncodeDecode();
>-+TEST_F(TestVp8Impl, ChecksSimulcastSettings) {
>-+  codec_settings_.numberOfSimulcastStreams = 2;
>-+  // Reslutions are not scaled by 2, temporal layers do not match.
>-+  codec_settings_.simulcastStream[0] = {kWidth, kHeight, 2, 4000,
>-+                                        3000,   2000,    80};
>-+  codec_settings_.simulcastStream[1] = {kWidth, kHeight, 3, 4000,
>-+                                        3000,   2000,    80};
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED,
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-+  codec_settings_.numberOfSimulcastStreams = 3;
>-+  // Reslutions are not scaled by 2.
>-+  codec_settings_.simulcastStream[0] = {kWidth / 2, kHeight / 2, 1, 4000,
>-+                                        3000,       2000,        80};
>-+  codec_settings_.simulcastStream[1] = {kWidth / 2, kHeight / 2, 1, 4000,
>-+                                        3000,       2000,        80};
>-+  codec_settings_.simulcastStream[2] = {kWidth, kHeight, 1, 4000,
>-+                                        3000,   2000,    80};
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED,
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-+  // Reslutions are not scaled by 2.
>-+  codec_settings_.simulcastStream[0] = {kWidth, kHeight, 1, 4000,
>-+                                        3000,   2000,    80};
>-+  codec_settings_.simulcastStream[1] = {kWidth, kHeight, 1, 4000,
>-+                                        3000,   2000,    80};
>-+  codec_settings_.simulcastStream[2] = {kWidth, kHeight, 1, 4000,
>-+                                        3000,   2000,    80};
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED,
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-+  // Temporal layers do not match.
>-+  codec_settings_.simulcastStream[0] = {kWidth / 4, kHeight / 4, 1, 4000,
>-+                                        3000,       2000,        80};
>-+  codec_settings_.simulcastStream[1] = {kWidth / 2, kHeight / 2, 2, 4000,
>-+                                        3000,       2000,        80};
>-+  codec_settings_.simulcastStream[2] = {kWidth, kHeight, 3, 4000,
>-+                                        3000,   2000,    80};
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED,
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-+  // Resolutions do not match codec config.
>-+  codec_settings_.simulcastStream[0] = {
>-+      kWidth / 4 + 1, kHeight / 4 + 1, 1, 4000, 3000, 2000, 80};
>-+  codec_settings_.simulcastStream[1] = {
>-+      kWidth / 2 + 2, kHeight / 2 + 2, 1, 4000, 3000, 2000, 80};
>-+  codec_settings_.simulcastStream[2] = {kWidth + 4, kHeight + 4, 1, 4000,
>-+                                        3000,       2000,        80};
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED,
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-+  // Everything fine: scaling by 2, top resolution matches video, temporal
>-+  // settings are the same for all layers.
>-+  codec_settings_.simulcastStream[0] = {kWidth / 4, kHeight / 4, 1, 4000,
>-+                                        3000,       2000,        80};
>-+  codec_settings_.simulcastStream[1] = {kWidth / 2, kHeight / 2, 1, 4000,
>-+                                        3000,       2000,        80};
>-+  codec_settings_.simulcastStream[2] = {kWidth, kHeight, 1, 4000,
>-+                                        3000,   2000,    80};
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  EXPECT_GT(WaitForEncodedFrame(), 0u);
>--
>--  int qp = 0;
>--  ASSERT_TRUE(vp8::GetQp(encoded_frame_._buffer, encoded_frame_._length, &qp));
>--
>--  EXPECT_EQ(encoded_frame_.qp_, qp);
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>- }
>- 
>- #if defined(WEBRTC_ANDROID)
>-@@ -327,20 +349,18 @@ TEST_F(TestVp8Impl, ParserQpEqualsEncodedQp) {
>- #define MAYBE_AlignedStrideEncodeDecode AlignedStrideEncodeDecode
>- #endif
>- TEST_F(TestVp8Impl, MAYBE_AlignedStrideEncodeDecode) {
>--  SetUpEncodeDecode();
>--  encoder_->Encode(*input_frame_, nullptr, nullptr);
>--  EXPECT_GT(WaitForEncodedFrame(), 0u);
>-+  InitEncodeDecode();
>-+  EncodeFrame();
>-   // First frame should be a key frame.
>--  encoded_frame_._frameType = kVideoFrameKey;
>--  encoded_frame_.ntp_time_ms_ = kTestNtpTimeMs;
>-+  encoded_cb_.encoded_frame_._frameType = kVideoFrameKey;
>-+  encoded_cb_.encoded_frame_.ntp_time_ms_ = kTestNtpTimeMs;
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            decoder_->Decode(encoded_frame_, false, nullptr));
>--  EXPECT_GT(WaitForDecodedFrame(), 0u);
>--  ASSERT_TRUE(decoded_frame_);
>-+            decoder_->Decode(encoded_cb_.encoded_frame_, false, nullptr));
>-+  EXPECT_TRUE(decoded_cb_.DecodeComplete());
>-   // Compute PSNR on all planes (faster than SSIM).
>--  EXPECT_GT(I420PSNR(input_frame_.get(), &*decoded_frame_), 36);
>--  EXPECT_EQ(kTestTimestamp, decoded_frame_->timestamp());
>--  EXPECT_EQ(kTestNtpTimeMs, decoded_frame_->ntp_time_ms());
>-+  EXPECT_GT(I420PSNR(input_frame_.get(), &*decoded_cb_.frame_), 36);
>-+  EXPECT_EQ(kInitialTimestampRtp, decoded_cb_.frame_->timestamp());
>-+  EXPECT_EQ(kTestNtpTimeMs, decoded_cb_.frame_->ntp_time_ms());
>- }
>- 
>- #if defined(WEBRTC_ANDROID)
>-@@ -349,100 +369,138 @@ TEST_F(TestVp8Impl, MAYBE_AlignedStrideEncodeDecode) {
>- #define MAYBE_DecodeWithACompleteKeyFrame DecodeWithACompleteKeyFrame
>- #endif
>- TEST_F(TestVp8Impl, MAYBE_DecodeWithACompleteKeyFrame) {
>--  SetUpEncodeDecode();
>--  encoder_->Encode(*input_frame_, nullptr, nullptr);
>--  EXPECT_GT(WaitForEncodedFrame(), 0u);
>-+  InitEncodeDecode();
>-+  EncodeFrame();
>-   // Setting complete to false -> should return an error.
>--  encoded_frame_._completeFrame = false;
>-+  encoded_cb_.encoded_frame_._completeFrame = false;
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERROR,
>--            decoder_->Decode(encoded_frame_, false, nullptr));
>-+            decoder_->Decode(encoded_cb_.encoded_frame_, false, nullptr));
>-   // Setting complete back to true. Forcing a delta frame.
>--  encoded_frame_._frameType = kVideoFrameDelta;
>--  encoded_frame_._completeFrame = true;
>-+  encoded_cb_.encoded_frame_._frameType = kVideoFrameDelta;
>-+  encoded_cb_.encoded_frame_._completeFrame = true;
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERROR,
>--            decoder_->Decode(encoded_frame_, false, nullptr));
>-+            decoder_->Decode(encoded_cb_.encoded_frame_, false, nullptr));
>-   // Now setting a key frame.
>--  encoded_frame_._frameType = kVideoFrameKey;
>-+  encoded_cb_.encoded_frame_._frameType = kVideoFrameKey;
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            decoder_->Decode(encoded_frame_, false, nullptr));
>--  ASSERT_TRUE(decoded_frame_);
>--  EXPECT_GT(I420PSNR(input_frame_.get(), &*decoded_frame_), 36);
>-+            decoder_->Decode(encoded_cb_.encoded_frame_, false, nullptr));
>-+  ASSERT_TRUE(decoded_cb_.frame_);
>-+  EXPECT_GT(I420PSNR(input_frame_.get(), &*decoded_cb_.frame_), 36);
>- }
>- 
>--TEST_F(TestVp8Impl, EncoderRetainsRtpStateAfterRelease) {
>--  SetUpEncodeDecode();
>--  // Override default settings.
>-+TEST_F(TestVp8Impl, EncoderWith2TemporalLayersRetainsRtpStateAfterRelease) {
>-   codec_settings_.VP8()->numberOfTemporalLayers = 2;
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->InitEncode(&codec_settings_, 1, 1440));
>-+  InitEncodeDecode();
>- 
>-   // Temporal layer 0.
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  ASSERT_TRUE(WaitForEncodedFrame());
>--  EXPECT_EQ(0, codec_specific_info_.codecSpecific.VP8.temporalIdx);
>--  int16_t picture_id = codec_specific_info_.codecSpecific.VP8.pictureId;
>--  int tl0_pic_idx = codec_specific_info_.codecSpecific.VP8.tl0PicIdx;
>--
>-+  EncodeFrame();
>-+  EXPECT_EQ(0, encoded_cb_.codec_specific_info_.codecSpecific.VP8.temporalIdx);
>-+  int16_t picture_id =
>-+      encoded_cb_.codec_specific_info_.codecSpecific.VP8.pictureId;
>-+  int tl0_pic_idx =
>-+      encoded_cb_.codec_specific_info_.codecSpecific.VP8.tl0PicIdx;
>-   // Temporal layer 1.
>--  input_frame_->set_timestamp(input_frame_->timestamp() +
>--                              kTimestampIncrementPerFrame);
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  ExpectFrameWith((picture_id + 1) % (1 << 15), tl0_pic_idx, 1);
>--
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 1, tl0_pic_idx + 0, 1);
>-   // Temporal layer 0.
>--  input_frame_->set_timestamp(input_frame_->timestamp() +
>--                              kTimestampIncrementPerFrame);
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  ExpectFrameWith((picture_id + 2) % (1 << 15), (tl0_pic_idx + 1) % (1 << 8),
>--                  0);
>--
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 2, tl0_pic_idx + 1, 0);
>-   // Temporal layer 1.
>--  input_frame_->set_timestamp(input_frame_->timestamp() +
>--                              kTimestampIncrementPerFrame);
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  ExpectFrameWith((picture_id + 3) % (1 << 15), (tl0_pic_idx + 1) % (1 << 8),
>--                  1);
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 3, tl0_pic_idx + 1, 1);
>- 
>-   // Reinit.
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->InitEncode(&codec_settings_, 1, 1440));
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>- 
>-   // Temporal layer 0.
>--  input_frame_->set_timestamp(input_frame_->timestamp() +
>--                              kTimestampIncrementPerFrame);
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  ExpectFrameWith((picture_id + 4) % (1 << 15), (tl0_pic_idx + 2) % (1 << 8),
>--                  0);
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 4, tl0_pic_idx + 2, 0);
>-+  // Temporal layer 1.
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 5, tl0_pic_idx + 2, 1);
>-+  // Temporal layer 0.
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 6, tl0_pic_idx + 3, 0);
>-+  // Temporal layer 1.
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 7, tl0_pic_idx + 3, 1);
>-+}
>- 
>-+TEST_F(TestVp8Impl, EncoderWith3TemporalLayersRetainsRtpStateAfterRelease) {
>-+  codec_settings_.VP8()->numberOfTemporalLayers = 3;
>-+  InitEncodeDecode();
>-+
>-+  // Temporal layer 0.
>-+  EncodeFrame();
>-+  EXPECT_EQ(0, encoded_cb_.codec_specific_info_.codecSpecific.VP8.temporalIdx);
>-+  int16_t picture_id =
>-+      encoded_cb_.codec_specific_info_.codecSpecific.VP8.pictureId;
>-+  int tl0_pic_idx =
>-+      encoded_cb_.codec_specific_info_.codecSpecific.VP8.tl0PicIdx;
>-+  // Temporal layer 2.
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 1, tl0_pic_idx + 0, 2);
>-   // Temporal layer 1.
>--  input_frame_->set_timestamp(input_frame_->timestamp() +
>--                              kTimestampIncrementPerFrame);
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 2, tl0_pic_idx + 0, 1);
>-+  // Temporal layer 2.
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 3, tl0_pic_idx + 0, 2);
>-+
>-+  // Reinit.
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  ExpectFrameWith((picture_id + 5) % (1 << 15), (tl0_pic_idx + 2) % (1 << 8),
>--                  1);
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>- 
>-   // Temporal layer 0.
>--  input_frame_->set_timestamp(input_frame_->timestamp() +
>--                              kTimestampIncrementPerFrame);
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 4, tl0_pic_idx + 1, 0);
>-+  // Temporal layer 2.
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 5, tl0_pic_idx + 1, 2);
>-+  // Temporal layer 1.
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 6, tl0_pic_idx + 1, 1);
>-+  // Temporal layer 2.
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 7, tl0_pic_idx + 1, 2);
>-+}
>-+
>-+TEST_F(TestVp8Impl, ScalingDisabledIfAutomaticResizeOff) {
>-+  codec_settings_.VP8()->frameDroppingOn = true;
>-+  codec_settings_.VP8()->automaticResizeOn = false;
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  ExpectFrameWith((picture_id + 6) % (1 << 15), (tl0_pic_idx + 3) % (1 << 8),
>--                  0);
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>- 
>--  // Temporal layer 1.
>--  input_frame_->set_timestamp(input_frame_->timestamp() +
>--                              kTimestampIncrementPerFrame);
>-+  VideoEncoder::ScalingSettings settings = encoder_->GetScalingSettings();
>-+  EXPECT_FALSE(settings.enabled);
>-+}
>-+
>-+TEST_F(TestVp8Impl, ScalingEnabledIfAutomaticResizeOn) {
>-+  codec_settings_.VP8()->frameDroppingOn = true;
>-+  codec_settings_.VP8()->automaticResizeOn = true;
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  ExpectFrameWith((picture_id + 7) % (1 << 15), (tl0_pic_idx + 3) % (1 << 8),
>--                  1);
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-+
>-+  VideoEncoder::ScalingSettings settings = encoder_->GetScalingSettings();
>-+  EXPECT_TRUE(settings.enabled);
>-+  EXPECT_EQ(kDefaultMinPixelsPerFrame, settings.min_pixels_per_frame);
>- }
>- 
>- }  // namespace webrtc
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
>-index 5038e8edf49..9e89696d465 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
>-@@ -8,7 +8,7 @@
>-  *  be found in the AUTHORS file in the root of the source tree.
>-  */
>- 
>--#include "webrtc/modules/video_coding/codecs/vp8/vp8_impl.h"
>-+#include "modules/video_coding/codecs/vp8/vp8_impl.h"
>- 
>- #include <stdlib.h>
>- #include <string.h>
>-@@ -17,29 +17,31 @@
>- #include <string>
>- 
>- // NOTE(ajm): Path provided by gyp.
>--#include "libyuv/scale.h"    // NOLINT
>- #include "libyuv/convert.h"  // NOLINT
>-+#include "libyuv/scale.h"    // NOLINT
>- 
>--#include "webrtc/base/checks.h"
>--#include "webrtc/base/random.h"
>--#include "webrtc/base/timeutils.h"
>--#include "webrtc/base/trace_event.h"
>--#include "webrtc/common_types.h"
>--#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
>--#include "webrtc/modules/include/module_common_types.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
>--#include "webrtc/modules/video_coding/include/video_codec_interface.h"
>--#include "webrtc/system_wrappers/include/clock.h"
>--#include "webrtc/system_wrappers/include/field_trial.h"
>--#include "webrtc/system_wrappers/include/metrics.h"
>-+#include "common_types.h"  // NOLINT(build/include)
>-+#include "common_video/libyuv/include/webrtc_libyuv.h"
>-+#include "modules/include/module_common_types.h"
>-+#include "modules/video_coding/codecs/vp8/include/vp8_common_types.h"
>-+#include "modules/video_coding/codecs/vp8/screenshare_layers.h"
>-+#include "modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
>-+#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>-+#include "modules/video_coding/include/video_codec_interface.h"
>-+#include "rtc_base/checks.h"
>-+#include "rtc_base/numerics/exp_filter.h"
>-+#include "rtc_base/ptr_util.h"
>-+#include "rtc_base/random.h"
>-+#include "rtc_base/timeutils.h"
>-+#include "rtc_base/trace_event.h"
>-+#include "system_wrappers/include/clock.h"
>-+#include "system_wrappers/include/field_trial.h"
>-+#include "system_wrappers/include/metrics.h"
>- 
>- namespace webrtc {
>- namespace {
>- 
>--const char kVp8PostProcArmFieldTrial[] = "WebRTC-VP8-Postproc-Arm";
>-+const char kVp8PostProcArmFieldTrial[] = "WebRTC-VP8-Postproc-Config-Arm";
>- const char kVp8GfBoostFieldTrial[] = "WebRTC-VP8-GfBoost";
>- 
>- const int kTokenPartitions = VP8_ONE_TOKENPARTITION;
>-@@ -97,6 +99,21 @@ bool ValidSimulcastResolutions(const VideoCodec& codec, int num_streams) {
>-       return false;
>-     }
>-   }
>-+  for (int i = 1; i < num_streams; ++i) {
>-+    if (codec.simulcastStream[i].width !=
>-+        codec.simulcastStream[i - 1].width * 2) {
>-+      return false;
>-+    }
>-+  }
>-+  return true;
>-+}
>-+
>-+bool ValidSimulcastTemporalLayers(const VideoCodec& codec, int num_streams) {
>-+  for (int i = 0; i < num_streams - 1; ++i) {
>-+    if (codec.simulcastStream[i].numberOfTemporalLayers !=
>-+        codec.simulcastStream[i + 1].numberOfTemporalLayers)
>-+      return false;
>-+  }
>-   return true;
>- }
>- 
>-@@ -122,14 +139,36 @@ bool GetGfBoostPercentageFromFieldTrialGroup(int* boost_percentage) {
>- 
>-   return true;
>- }
>-+
>-+void GetPostProcParamsFromFieldTrialGroup(
>-+    VP8DecoderImpl::DeblockParams* deblock_params) {
>-+  std::string group =
>-+      webrtc::field_trial::FindFullName(kVp8PostProcArmFieldTrial);
>-+  if (group.empty())
>-+    return;
>-+
>-+  VP8DecoderImpl::DeblockParams params;
>-+  if (sscanf(group.c_str(), "Enabled-%d,%d,%d", &params.max_level,
>-+             &params.min_qp, &params.degrade_qp) != 3)
>-+    return;
>-+
>-+  if (params.max_level < 0 || params.max_level > 16)
>-+    return;
>-+
>-+  if (params.min_qp < 0 || params.degrade_qp <= params.min_qp)
>-+    return;
>-+
>-+  *deblock_params = params;
>-+}
>-+
>- }  // namespace
>- 
>--VP8Encoder* VP8Encoder::Create() {
>--  return new VP8EncoderImpl();
>-+std::unique_ptr<VP8Encoder> VP8Encoder::Create() {
>-+  return rtc::MakeUnique<VP8EncoderImpl>();
>- }
>- 
>--VP8Decoder* VP8Decoder::Create() {
>--  return new VP8DecoderImpl();
>-+std::unique_ptr<VP8Decoder> VP8Decoder::Create() {
>-+  return rtc::MakeUnique<VP8DecoderImpl>();
>- }
>- 
>- vpx_enc_frame_flags_t VP8EncoderImpl::EncodeFlags(
>-@@ -173,6 +212,7 @@ VP8EncoderImpl::VP8EncoderImpl()
>-     tl0_pic_idx_.push_back(random.Rand<uint8_t>());
>-   }
>-   temporal_layers_.reserve(kMaxSimulcastStreams);
>-+  temporal_layers_checkers_.reserve(kMaxSimulcastStreams);
>-   raw_images_.reserve(kMaxSimulcastStreams);
>-   encoded_images_.reserve(kMaxSimulcastStreams);
>-   send_stream_.reserve(kMaxSimulcastStreams);
>-@@ -212,6 +252,7 @@ int VP8EncoderImpl::Release() {
>-     tl0_pic_idx_[i] = temporal_layers_[i]->Tl0PicIdx();
>-   }
>-   temporal_layers_.clear();
>-+  temporal_layers_checkers_.clear();
>-   inited_ = false;
>-   return ret_val;
>- }
>-@@ -281,8 +322,7 @@ const char* VP8EncoderImpl::ImplementationName() const {
>-   return "libvpx";
>- }
>- 
>--void VP8EncoderImpl::SetStreamState(bool send_stream,
>--                                            int stream_idx) {
>-+void VP8EncoderImpl::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.
>-     key_frame_request_[stream_idx] = true;
>-@@ -298,6 +338,8 @@ void VP8EncoderImpl::SetupTemporalLayers(int num_streams,
>-   if (num_streams == 1) {
>-     temporal_layers_.emplace_back(
>-         tl_factory->Create(0, num_temporal_layers, tl0_pic_idx_[0]));
>-+    temporal_layers_checkers_.emplace_back(
>-+        tl_factory->CreateChecker(0, num_temporal_layers, tl0_pic_idx_[0]));
>-   } else {
>-     for (int i = 0; i < num_streams; ++i) {
>-       RTC_CHECK_GT(num_temporal_layers, 0);
>-@@ -305,6 +347,8 @@ void VP8EncoderImpl::SetupTemporalLayers(int num_streams,
>-                             codec.simulcastStream[i].numberOfTemporalLayers);
>-       temporal_layers_.emplace_back(
>-           tl_factory->Create(i, layers, tl0_pic_idx_[i]));
>-+      temporal_layers_checkers_.emplace_back(
>-+          tl_factory->CreateChecker(i, layers, tl0_pic_idx_[i]));
>-     }
>-   }
>- }
>-@@ -339,8 +383,10 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
>-   int number_of_streams = NumberOfStreams(*inst);
>-   bool doing_simulcast = (number_of_streams > 1);
>- 
>--  if (doing_simulcast && !ValidSimulcastResolutions(*inst, number_of_streams)) {
>--    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>-+  if (doing_simulcast &&
>-+      (!ValidSimulcastResolutions(*inst, number_of_streams) ||
>-+       !ValidSimulcastTemporalLayers(*inst, number_of_streams))) {
>-+    return WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED;
>-   }
>- 
>-   int num_temporal_layers =
>-@@ -524,7 +570,8 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
>- }
>- 
>- int VP8EncoderImpl::SetCpuSpeed(int width, int height) {
>--#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || defined(ANDROID)
>-+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) \
>-+  || defined(WEBRTC_ANDROID)
>-   // On mobile platform, use a lower speed setting for lower resolutions for
>-   // CPUs with 4 or more cores.
>-   RTC_DCHECK_GT(number_of_cores_, 0);
>-@@ -549,7 +596,7 @@ int VP8EncoderImpl::SetCpuSpeed(int width, int height) {
>- }
>- 
>- int VP8EncoderImpl::NumberOfThreads(int width, int height, int cpus) {
>--#if defined(ANDROID)
>-+#if defined(WEBRTC_ANDROID)
>-   if (width * height >= 320 * 180) {
>-     if (cpus >= 4) {
>-       // 3 threads for CPUs with 4 and more cores since most of times only 4
>-@@ -603,7 +650,8 @@ int VP8EncoderImpl::InitAndSetControlSettings() {
>-   // when encoding lower resolution streams. Would it work with the
>-   // multi-res encoding feature?
>-   denoiserState denoiser_state = kDenoiserOnYOnly;
>--#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || defined(ANDROID)
>-+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) \
>-+  || defined(WEBRTC_ANDROID)
>-   denoiser_state = kDenoiserOnYOnly;
>- #else
>-   denoiser_state = kDenoiserOnAdaptive;
>-@@ -703,17 +751,6 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
>-         raw_images_[i].stride[VPX_PLANE_V], raw_images_[i].d_w,
>-         raw_images_[i].d_h, libyuv::kFilterBilinear);
>-   }
>--  vpx_enc_frame_flags_t flags[kMaxSimulcastStreams];
>--  TemporalLayers::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) {
>--      // Drop this frame.
>--      return WEBRTC_VIDEO_CODEC_OK;
>--    }
>--    flags[i] = EncodeFlags(tl_configs[i]);
>--  }
>-   bool send_key_frame = false;
>-   for (size_t i = 0; i < key_frame_request_.size() && i < send_stream_.size();
>-        ++i) {
>-@@ -731,6 +768,18 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
>-       }
>-     }
>-   }
>-+  vpx_enc_frame_flags_t flags[kMaxSimulcastStreams];
>-+  TemporalLayers::FrameConfig tl_configs[kMaxSimulcastStreams];
>-+  for (size_t i = 0; i < encoders_.size(); ++i) {
>-+    tl_configs[i] = temporal_layers_[i]->UpdateLayerConfig(frame.timestamp());
>-+    RTC_DCHECK(temporal_layers_checkers_[i]->CheckTemporalConfig(
>-+        send_key_frame, tl_configs[i]));
>-+    if (tl_configs[i].drop_frame) {
>-+      // Drop this frame.
>-+      return WEBRTC_VIDEO_CODEC_OK;
>-+    }
>-+    flags[i] = EncodeFlags(tl_configs[i]);
>-+  }
>-   if (send_key_frame) {
>-     // Adapt the size of the key frame when in screenshare with 1 temporal
>-     // layer.
>-@@ -765,9 +814,8 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
>-     }
>- 
>-     vpx_codec_control(&encoders_[i], VP8E_SET_FRAME_FLAGS, flags[stream_idx]);
>--    vpx_codec_control(
>--        &encoders_[i], VP8E_SET_TEMPORAL_LAYER_ID,
>--        temporal_layers_[stream_idx]->GetTemporalLayerId(tl_configs[i]));
>-+    vpx_codec_control(&encoders_[i], VP8E_SET_TEMPORAL_LAYER_ID,
>-+                      tl_configs[i].encoder_layer_id);
>-   }
>-   // TODO(holmer): Ideally the duration should be the timestamp diff of this
>-   // frame and the next frame to be encoded, which we don't have. Instead we
>-@@ -777,20 +825,31 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
>-   assert(codec_.maxFramerate > 0);
>-   uint32_t duration = 90000 / codec_.maxFramerate;
>- 
>--  // Note we must pass 0 for |flags| field in encode call below since they are
>--  // set above in |vpx_codec_control| function for each encoder/spatial layer.
>--  int error = vpx_codec_encode(&encoders_[0], &raw_images_[0], timestamp_,
>--                               duration, 0, VPX_DL_REALTIME);
>--  // Reset specific intra frame thresholds, following the key frame.
>--  if (send_key_frame) {
>--    vpx_codec_control(&(encoders_[0]), VP8E_SET_MAX_INTRA_BITRATE_PCT,
>--                      rc_max_intra_target_);
>-+  int error = WEBRTC_VIDEO_CODEC_OK;
>-+  int num_tries = 0;
>-+  // If the first try returns WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT
>-+  // the frame must be reencoded with the same parameters again because
>-+  // target bitrate is exceeded and encoder state has been reset.
>-+  while (num_tries == 0 ||
>-+         (num_tries == 1 &&
>-+          error == WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT)) {
>-+    ++num_tries;
>-+    // Note we must pass 0 for |flags| field in encode call below since they are
>-+    // set above in |vpx_codec_control| function for each encoder/spatial layer.
>-+    error = vpx_codec_encode(&encoders_[0], &raw_images_[0], timestamp_,
>-+                             duration, 0, VPX_DL_REALTIME);
>-+    // Reset specific intra frame thresholds, following the key frame.
>-+    if (send_key_frame) {
>-+      vpx_codec_control(&(encoders_[0]), VP8E_SET_MAX_INTRA_BITRATE_PCT,
>-+                        rc_max_intra_target_);
>-+    }
>-+    if (error)
>-+      return WEBRTC_VIDEO_CODEC_ERROR;
>-+    timestamp_ += duration;
>-+    // Examines frame timestamps only.
>-+    error = GetEncodedPartitions(tl_configs, frame);
>-   }
>--  if (error)
>--    return WEBRTC_VIDEO_CODEC_ERROR;
>--  timestamp_ += duration;
>--  // Examines frame timestamps only.
>--  return GetEncodedPartitions(tl_configs, frame);
>-+  return error;
>- }
>- 
>- void VP8EncoderImpl::PopulateCodecSpecific(
>-@@ -878,7 +937,7 @@ int VP8EncoderImpl::GetEncodedPartitions(
>-     encoded_images_[encoder_idx].content_type_ =
>-         (codec_.mode == kScreensharing) ? VideoContentType::SCREENSHARE
>-                                         : VideoContentType::UNSPECIFIED;
>--    encoded_images_[encoder_idx].timing_.is_timing_frame = false;
>-+    encoded_images_[encoder_idx].timing_.flags = TimingFrameFlags::kInvalid;
>- 
>-     int qp = -1;
>-     vpx_codec_control(&encoders_[encoder_idx], VP8E_GET_LAST_QUANTIZER_64, &qp);
>-@@ -926,9 +985,33 @@ int VP8EncoderImpl::RegisterEncodeCompleteCallback(
>-   return WEBRTC_VIDEO_CODEC_OK;
>- }
>- 
>-+class VP8DecoderImpl::QpSmoother {
>-+ public:
>-+  QpSmoother() : last_sample_ms_(rtc::TimeMillis()), smoother_(kAlpha) {}
>-+
>-+  int GetAvg() const {
>-+    float value = smoother_.filtered();
>-+    return (value == rtc::ExpFilter::kValueUndefined) ? 0
>-+                                                      : static_cast<int>(value);
>-+  }
>-+
>-+  void Add(float sample) {
>-+    int64_t now_ms = rtc::TimeMillis();
>-+    smoother_.Apply(static_cast<float>(now_ms - last_sample_ms_), sample);
>-+    last_sample_ms_ = now_ms;
>-+  }
>-+
>-+  void Reset() { smoother_.Reset(kAlpha); }
>-+
>-+ private:
>-+  const float kAlpha = 0.95f;
>-+  int64_t last_sample_ms_;
>-+  rtc::ExpFilter smoother_;
>-+};
>-+
>- VP8DecoderImpl::VP8DecoderImpl()
>--    : use_postproc_arm_(webrtc::field_trial::FindFullName(
>--                            kVp8PostProcArmFieldTrial) == "Enabled"),
>-+    : use_postproc_arm_(
>-+          webrtc::field_trial::IsEnabled(kVp8PostProcArmFieldTrial)),
>-       buffer_pool_(false, 300 /* max_number_of_buffers*/),
>-       decode_complete_callback_(NULL),
>-       inited_(false),
>-@@ -936,7 +1019,11 @@ VP8DecoderImpl::VP8DecoderImpl()
>-       propagation_cnt_(-1),
>-       last_frame_width_(0),
>-       last_frame_height_(0),
>--      key_frame_required_(true) {}
>-+      key_frame_required_(true),
>-+      qp_smoother_(use_postproc_arm_ ? new QpSmoother() : nullptr) {
>-+  if (use_postproc_arm_)
>-+    GetPostProcParamsFromFieldTrialGroup(&deblock_);
>-+}
>- 
>- VP8DecoderImpl::~VP8DecoderImpl() {
>-   inited_ = true;  // in order to do the actual release
>-@@ -957,7 +1044,8 @@ int VP8DecoderImpl::InitDecode(const VideoCodec* inst, int number_of_cores) {
>-   cfg.threads = 1;
>-   cfg.h = cfg.w = 0;  // set after decode
>- 
>--#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || defined(ANDROID)
>-+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) \
>-+  || defined(WEBRTC_ANDROID)
>-   vpx_codec_flags_t flags = use_postproc_arm_ ? VPX_CODEC_USE_POSTPROC : 0;
>- #else
>-   vpx_codec_flags_t flags = VPX_CODEC_USE_POSTPROC;
>-@@ -996,16 +1084,28 @@ int VP8DecoderImpl::Decode(const EncodedImage& input_image,
>-   }
>- 
>- // Post process configurations.
>--#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || defined(ANDROID)
>-+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) \
>-+  || defined(WEBRTC_ANDROID)
>-   if (use_postproc_arm_) {
>-     vp8_postproc_cfg_t ppcfg;
>-     ppcfg.post_proc_flag = VP8_MFQE;
>--    // For low resolutions, use stronger deblocking filter and enable the
>--    // deblock and demacroblocker.
>-+    // For low resolutions, use stronger deblocking filter.
>-     int last_width_x_height = last_frame_width_ * last_frame_height_;
>-     if (last_width_x_height > 0 && last_width_x_height <= 320 * 240) {
>--      ppcfg.deblocking_level = 6;  // Only affects VP8_DEMACROBLOCK.
>--      ppcfg.post_proc_flag |= VP8_DEBLOCK | VP8_DEMACROBLOCK;
>-+      // Enable the deblock and demacroblocker based on qp thresholds.
>-+      RTC_DCHECK(qp_smoother_);
>-+      int qp = qp_smoother_->GetAvg();
>-+      if (qp > deblock_.min_qp) {
>-+        int level = deblock_.max_level;
>-+        if (qp < deblock_.degrade_qp) {
>-+          // Use lower level.
>-+          level = deblock_.max_level * (qp - deblock_.min_qp) /
>-+                  (deblock_.degrade_qp - deblock_.min_qp);
>-+        }
>-+        // Deblocking level only affects VP8_DEMACROBLOCK.
>-+        ppcfg.deblocking_level = std::max(level, 1);
>-+        ppcfg.post_proc_flag |= VP8_DEBLOCK | VP8_DEMACROBLOCK;
>-+      }
>-     }
>-     vpx_codec_control(decoder_, VP8_SET_POSTPROC, &ppcfg);
>-   }
>-@@ -1105,6 +1205,13 @@ int VP8DecoderImpl::ReturnFrame(const vpx_image_t* img,
>-     // Decoder OK and NULL image => No show frame
>-     return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
>-   }
>-+  if (qp_smoother_) {
>-+    if (last_frame_width_ != static_cast<int>(img->d_w) ||
>-+        last_frame_height_ != static_cast<int>(img->d_h)) {
>-+      qp_smoother_->Reset();
>-+    }
>-+    qp_smoother_->Add(qp);
>-+  }
>-   last_frame_width_ = img->d_w;
>-   last_frame_height_ = img->d_h;
>-   // Allocate memory for decoded image.
>-@@ -1122,13 +1229,12 @@ int VP8DecoderImpl::ReturnFrame(const vpx_image_t* img,
>-                    img->planes[VPX_PLANE_V], img->stride[VPX_PLANE_V],
>-                    buffer->MutableDataY(), buffer->StrideY(),
>-                    buffer->MutableDataU(), buffer->StrideU(),
>--                   buffer->MutableDataV(), buffer->StrideV(),
>--                   img->d_w, img->d_h);
>-+                   buffer->MutableDataV(), buffer->StrideV(), img->d_w,
>-+                   img->d_h);
>- 
>-   VideoFrame decoded_image(buffer, timestamp, 0, kVideoRotation_0);
>-   decoded_image.set_ntp_time_ms(ntp_time_ms);
>--  decode_complete_callback_->Decoded(decoded_image, rtc::Optional<int32_t>(),
>--                                     rtc::Optional<uint8_t>(qp));
>-+  decode_complete_callback_->Decoded(decoded_image, rtc::nullopt, qp);
>- 
>-   return WEBRTC_VIDEO_CODEC_OK;
>- }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
>-index 376dde12808..e5d5d81e35c 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
>-@@ -10,26 +10,26 @@
>-  * WEBRTC VP8 wrapper interface
>-  */
>- 
>--#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
>--#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
>-+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
>-+#define MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
>- 
>- #include <memory>
>- #include <vector>
>- 
>- // NOTE: This include order must remain to avoid compile errors, even though
>- //       it breaks the style guide.
>--#include "vpx/vpx_encoder.h"
>--#include "vpx/vpx_decoder.h"
>- #include "vpx/vp8cx.h"
>- #include "vpx/vp8dx.h"
>-+#include "vpx/vpx_decoder.h"
>-+#include "vpx/vpx_encoder.h"
>- 
>--#include "webrtc/api/video/video_frame.h"
>--#include "webrtc/common_video/include/i420_buffer_pool.h"
>--#include "webrtc/common_video/include/video_frame.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
>--#include "webrtc/modules/video_coding/include/video_codec_interface.h"
>--#include "webrtc/modules/video_coding/utility/quality_scaler.h"
>-+#include "api/video/video_frame.h"
>-+#include "common_video/include/i420_buffer_pool.h"
>-+#include "common_video/include/video_frame.h"
>-+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>-+#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>-+#include "modules/video_coding/include/video_codec_interface.h"
>-+#include "modules/video_coding/utility/quality_scaler.h"
>- 
>- namespace webrtc {
>- 
>-@@ -104,6 +104,7 @@ class VP8EncoderImpl : public VP8Encoder {
>-   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<uint16_t> picture_id_;
>-   std::vector<uint8_t> tl0_pic_idx_;
>-   std::vector<bool> key_frame_request_;
>-@@ -135,7 +136,14 @@ class VP8DecoderImpl : public VP8Decoder {
>- 
>-   const char* ImplementationName() const override;
>- 
>-+  struct DeblockParams {
>-+    int max_level = 6;   // Deblocking strength: [0, 16].
>-+    int degrade_qp = 1;  // If QP value is below, start lowering |max_level|.
>-+    int min_qp = 0;      // If QP value is below, turn off deblocking.
>-+  };
>-+
>-  private:
>-+  class QpSmoother;
>-   int ReturnFrame(const vpx_image_t* img,
>-                   uint32_t timeStamp,
>-                   int64_t ntp_time_ms,
>-@@ -151,7 +159,9 @@ class VP8DecoderImpl : public VP8Decoder {
>-   int last_frame_width_;
>-   int last_frame_height_;
>-   bool key_frame_required_;
>-+  DeblockParams deblock_;
>-+  const std::unique_ptr<QpSmoother> qp_smoother_;
>- };
>- }  // namespace webrtc
>- 
>--#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
>-+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_noop.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_noop.cc
>-index a61c3655190..3267344204c 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_noop.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_noop.cc
>-@@ -13,8 +13,8 @@
>- #error
>- #endif  // !defined(RTC_DISABLE_VP8)
>- 
>--#include "webrtc/base/checks.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
>-+#include "rtc_base/checks.h"
>-+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>- 
>- namespace webrtc {
>- 
>-@@ -22,7 +22,7 @@ bool VP8Encoder::IsSupported() {
>-   return false;
>- }
>- 
>--VP8Encoder* VP8Encoder::Create() {
>-+std::unique_ptr<VP8Encoder> VP8Encoder::Create() {
>-   RTC_NOTREACHED();
>-   return nullptr;
>- }
>-@@ -31,7 +31,7 @@ bool VP8Decoder::IsSupported() {
>-   return false;
>- }
>- 
>--VP8Decoder* VP8Decoder::Create() {
>-+std::unique_ptr<VP8Decoder> VP8Decoder::Create() {
>-   RTC_NOTREACHED();
>-   return nullptr;
>- }
>--- 
>-2.13.5 (Apple Git-94)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0004-Removing-parameter-names-from-files-included-from-We.patch b/Source/ThirdParty/libwebrtc/WebKit/0004-Removing-parameter-names-from-files-included-from-We.patch
>deleted file mode 100644
>index 359833fc7ee..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0004-Removing-parameter-names-from-files-included-from-We.patch
>+++ /dev/null
>@@ -1,682 +0,0 @@
>-From f848b7acea46bb74d5a41b9da5b2e48b31203a30 Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Thu, 14 Dec 2017 13:50:22 -0800
>-Subject: [PATCH 4/8] Removing parameter names from files included from WebKit
>-
>----
>- .../libwebrtc/Source/webrtc/api/array_view.h       |  4 +-
>- .../Source/webrtc/api/datachannelinterface.h       |  2 +-
>- .../ThirdParty/libwebrtc/Source/webrtc/api/jsep.h  |  2 +-
>- .../Source/webrtc/api/mediastreaminterface.h       | 20 ++---
>- .../Source/webrtc/api/peerconnectioninterface.h    | 87 +++++++++++-----------
>- .../libwebrtc/Source/webrtc/api/umametrics.h       |  6 +-
>- .../Source/webrtc/api/video_codecs/video_decoder.h |  8 +-
>- .../Source/webrtc/api/video_codecs/video_encoder.h |  2 +-
>- .../webrtc/common_video/include/frame_callback.h   |  2 +-
>- .../webrtc/logging/rtc_event_log/rtc_event_log.h   |  8 +-
>- .../Source/webrtc/media/base/mediachannel.h        |  6 +-
>- .../modules/audio_device/include/audio_device.h    | 16 ++--
>- .../Source/webrtc/p2p/base/icetransportinternal.h  |  2 +-
>- .../libwebrtc/Source/webrtc/p2p/base/port.h        |  2 +-
>- .../libwebrtc/Source/webrtc/p2p/base/stun.h        |  2 +-
>- .../libwebrtc/Source/webrtc/p2p/base/stunrequest.h |  6 +-
>- .../Source/webrtc/rtc_base/messagehandler.h        |  4 +-
>- .../libwebrtc/Source/webrtc/rtc_base/network.h     |  2 +-
>- .../rtc_base/numerics/safe_conversions_impl.h      |  2 +-
>- .../Source/webrtc/rtc_base/socketserver.h          |  2 +-
>- .../libwebrtc/Source/webrtc/rtc_base/stream.h      |  4 +-
>- 21 files changed, 94 insertions(+), 95 deletions(-)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view.h
>-index d951d0f02d3..b27bd29a8e2 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view.h
>-@@ -94,7 +94,7 @@ class ArrayViewBase {
>-   static_assert(Size > 0, "ArrayView size must be variable or non-negative");
>- 
>-  public:
>--  ArrayViewBase(T* data, size_t size) : data_(data) {}
>-+  ArrayViewBase(T* data, size_t) : data_(data) {}
>- 
>-   static constexpr size_t size() { return Size; }
>-   static constexpr bool empty() { return false; }
>-@@ -111,7 +111,7 @@ class ArrayViewBase {
>- template <typename T>
>- class ArrayViewBase<T, 0> {
>-  public:
>--  explicit ArrayViewBase(T* data, size_t size) {}
>-+  explicit ArrayViewBase(T*, size_t) {}
>- 
>-   static constexpr size_t size() { return 0; }
>-   static constexpr bool empty() { return true; }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h
>-index 4ab7efb7407..4b5ce9a7333 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h
>-@@ -90,7 +90,7 @@ class DataChannelObserver {
>-   //  A data buffer was successfully received.
>-   virtual void OnMessage(const DataBuffer& buffer) = 0;
>-   // The data channel's buffered_amount has changed.
>--  virtual void OnBufferedAmountChange(uint64_t previous_amount) {}
>-+  virtual void OnBufferedAmountChange(uint64_t) {}
>- 
>-  protected:
>-   virtual ~DataChannelObserver() {}
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h
>-index acfffd4d0a7..59064d85758 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h
>-@@ -126,7 +126,7 @@ class SessionDescriptionInterface {
>-   //
>-   // Returns the number of candidates removed.
>-   virtual size_t RemoveCandidates(
>--      const std::vector<cricket::Candidate>& candidates) { return 0; }
>-+      const std::vector<cricket::Candidate>&) { return 0; }
>- 
>-   // Returns the number of m= sections in the session description.
>-   virtual size_t number_of_mediasections() const = 0;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h
>-index 80595e09abe..aae5a032937 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h
>-@@ -167,14 +167,14 @@ class VideoTrackInterface
>- 
>-   // Register a video sink for this track. Used to connect the track to the
>-   // underlying video engine.
>--  void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
>--                       const rtc::VideoSinkWants& wants) override {}
>--  void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override {}
>-+  void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>*,
>-+                       const rtc::VideoSinkWants&) override {}
>-+  void RemoveSink(rtc::VideoSinkInterface<VideoFrame>*) override {}
>- 
>-   virtual VideoTrackSourceInterface* GetSource() const = 0;
>- 
>-   virtual ContentHint content_hint() const { return ContentHint::kNone; }
>--  virtual void set_content_hint(ContentHint hint) {}
>-+  virtual void set_content_hint(ContentHint) {}
>- 
>-  protected:
>-   virtual ~VideoTrackInterface() {}
>-@@ -211,15 +211,15 @@ class AudioSourceInterface : public MediaSourceInterface {
>-   // Sets the volume of the source. |volume| is in  the range of [0, 10].
>-   // TODO(tommi): This method should be on the track and ideally volume should
>-   // be applied in the track in a way that does not affect clones of the track.
>--  virtual void SetVolume(double volume) {}
>-+  virtual void SetVolume(double) {}
>- 
>-   // Registers/unregisters observers to the audio source.
>--  virtual void RegisterAudioObserver(AudioObserver* observer) {}
>--  virtual void UnregisterAudioObserver(AudioObserver* observer) {}
>-+  virtual void RegisterAudioObserver(AudioObserver*) {}
>-+  virtual void UnregisterAudioObserver(AudioObserver*) {}
>- 
>-   // TODO(tommi): Make pure virtual.
>--  virtual void AddSink(AudioTrackSinkInterface* sink) {}
>--  virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
>-+  virtual void AddSink(AudioTrackSinkInterface*) {}
>-+  virtual void RemoveSink(AudioTrackSinkInterface*) {}
>- };
>- 
>- // Interface of the audio processor used by the audio track to collect
>-@@ -288,7 +288,7 @@ class AudioTrackInterface : public MediaStreamTrackInterface {
>-   // Return true on success, otherwise false.
>-   // TODO(deadbeef): Change the interface to int GetSignalLevel() and pure
>-   // virtual after it's implemented in chromium.
>--  virtual bool GetSignalLevel(int* level) { return false; }
>-+  virtual bool GetSignalLevel(int*) { return false; }
>- 
>-   // Get the audio processor used by the audio track. Return null if the track
>-   // does not have any processor.
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
>-index a1e7d7ee2fb..76f7b28945c 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
>-@@ -629,12 +629,12 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // Errors:
>-   // - INVALID_PARAMETER: |track| is null.
>-   virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
>--  AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track) {
>-+  AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface>) {
>-     return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
>-   }
>-   virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
>--  AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track,
>--                 const RtpTransceiverInit& init) {
>-+  AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface>,
>-+                 const RtpTransceiverInit&) {
>-     return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
>-   }
>- 
>-@@ -644,12 +644,12 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // - INVALID_PARAMETER: |media_type| is not MEDIA_TYPE_AUDIO or
>-   //                      MEDIA_TYPE_VIDEO.
>-   virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
>--  AddTransceiver(cricket::MediaType media_type) {
>-+  AddTransceiver(cricket::MediaType) {
>-     return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
>-   }
>-   virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
>--  AddTransceiver(cricket::MediaType media_type,
>--                 const RtpTransceiverInit& init) {
>-+  AddTransceiver(cricket::MediaType,
>-+                 const RtpTransceiverInit&) {
>-     return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
>-   }
>- 
>-@@ -675,8 +675,8 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // |stream_id| is used to populate the msid attribute; if empty, one will
>-   // be generated automatically.
>-   virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
>--      const std::string& kind,
>--      const std::string& stream_id) {
>-+      const std::string&,
>-+      const std::string&) {
>-     return rtc::scoped_refptr<RtpSenderInterface>();
>-   }
>- 
>-@@ -719,7 +719,7 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // TODO(hbos): Default implementation that does nothing only exists as to not
>-   // break third party projects. As soon as they have been updated this should
>-   // be changed to "= 0;".
>--  virtual void GetStats(RTCStatsCollectorCallback* callback) {}
>-+  virtual void GetStats(RTCStatsCollectorCallback*) {}
>- 
>-   // Create a data channel with the provided config, or default config if none
>-   // is provided. Note that an offer/answer negotiation is still necessary
>-@@ -760,23 +760,23 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>- 
>-   // Create a new offer.
>-   // The CreateSessionDescriptionObserver callback will be called when done.
>--  virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
>--                           const MediaConstraintsInterface* constraints) {}
>-+  virtual void CreateOffer(CreateSessionDescriptionObserver*,
>-+                           const MediaConstraintsInterface*) {}
>- 
>-   // TODO(jiayl): remove the default impl and the old interface when chromium
>-   // code is updated.
>--  virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
>--                           const RTCOfferAnswerOptions& options) {}
>-+  virtual void CreateOffer(CreateSessionDescriptionObserver*,
>-+                           const RTCOfferAnswerOptions&) {}
>- 
>-   // Create an answer to an offer.
>-   // The CreateSessionDescriptionObserver callback will be called when done.
>--  virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
>--                            const RTCOfferAnswerOptions& options) {}
>-+  virtual void CreateAnswer(CreateSessionDescriptionObserver*,
>-+                            const RTCOfferAnswerOptions&) {}
>-   // Deprecated - use version above.
>-   // TODO(hta): Remove and remove default implementations when all callers
>-   // are updated.
>--  virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
>--                            const MediaConstraintsInterface* constraints) {}
>-+  virtual void CreateAnswer(CreateSessionDescriptionObserver*,
>-+                            const MediaConstraintsInterface*) {}
>- 
>-   // Sets the local session description.
>-   // The PeerConnection takes the ownership of |desc| even if it fails.
>-@@ -789,19 +789,19 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // The PeerConnection takes the ownership of |desc| even if it fails.
>-   // The |observer| callback will be called when done.
>-   // TODO(hbos): Remove when Chrome implements the new signature.
>--  virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
>--                                    SessionDescriptionInterface* desc) {}
>-+  virtual void SetRemoteDescription(SetSessionDescriptionObserver*,
>-+                                    SessionDescriptionInterface*) {}
>-   // TODO(hbos): Make pure virtual when Chrome has updated its signature.
>-   virtual void SetRemoteDescription(
>--      std::unique_ptr<SessionDescriptionInterface> desc,
>--      rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {}
>-+      std::unique_ptr<SessionDescriptionInterface>,
>-+      rtc::scoped_refptr<SetRemoteDescriptionObserverInterface>) {}
>-   // Deprecated; Replaced by SetConfiguration.
>-   // TODO(deadbeef): Remove once Chrome is moved over to SetConfiguration.
>--  virtual bool UpdateIce(const IceServers& configuration,
>--                         const MediaConstraintsInterface* constraints) {
>-+  virtual bool UpdateIce(const IceServers&,
>-+                         const MediaConstraintsInterface*) {
>-     return false;
>-   }
>--  virtual bool UpdateIce(const IceServers& configuration) { return false; }
>-+  virtual bool UpdateIce(const IceServers&) { return false; }
>- 
>-   // TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
>-   // PeerConnectionInterface implement it.
>-@@ -833,14 +833,14 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
>-   // PeerConnectionInterface implement it.
>-   virtual bool SetConfiguration(
>--      const PeerConnectionInterface::RTCConfiguration& config,
>--      RTCError* error) {
>-+      const PeerConnectionInterface::RTCConfiguration&,
>-+      RTCError*) {
>-     return false;
>-   }
>-   // Version without error output param for backwards compatibility.
>-   // TODO(deadbeef): Remove once chromium is updated.
>-   virtual bool SetConfiguration(
>--      const PeerConnectionInterface::RTCConfiguration& config) {
>-+      const PeerConnectionInterface::RTCConfiguration&) {
>-     return false;
>-   }
>- 
>-@@ -854,7 +854,7 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // continual gathering, to avoid an ever-growing list of candidates as
>-   // networks come and go.
>-   virtual bool RemoveIceCandidates(
>--      const std::vector<cricket::Candidate>& candidates) {
>-+      const std::vector<cricket::Candidate>&) {
>-     return false;
>-   }
>- 
>-@@ -884,8 +884,7 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // ownership is passed with std::unique_ptr
>-   // TODO(alexnarest): Make this pure virtual when tests will be updated
>-   virtual void SetBitrateAllocationStrategy(
>--      std::unique_ptr<rtc::BitrateAllocationStrategy>
>--          bitrate_allocation_strategy) {}
>-+      std::unique_ptr<rtc::BitrateAllocationStrategy>) {}
>- 
>-   // Enable/disable playout of received audio streams. Enabled by default. Note
>-   // that even if playout is enabled, streams will only be played out if the
>-@@ -894,13 +893,13 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // for audio data every 10ms to ensure that audio processing happens and the
>-   // audio statistics are updated.
>-   // TODO(henrika): deprecate and remove this.
>--  virtual void SetAudioPlayout(bool playout) {}
>-+  virtual void SetAudioPlayout(bool) {}
>- 
>-   // Enable/disable recording of transmitted audio streams. Enabled by default.
>-   // Note that even if recording is enabled, streams will only be recorded if
>-   // the appropriate SDP is also applied.
>-   // TODO(henrika): deprecate and remove this.
>--  virtual void SetAudioRecording(bool recording) {}
>-+  virtual void SetAudioRecording(bool) {}
>- 
>-   // Returns the current SignalingState.
>-   virtual SignalingState signaling_state() = 0;
>-@@ -919,8 +918,8 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // automatically after 10 minutes have passed, or when the StopRtcEventLog
>-   // function is called.
>-   // TODO(eladalon): Deprecate and remove this.
>--  virtual bool StartRtcEventLog(rtc::PlatformFile file,
>--                                int64_t max_size_bytes) {
>-+  virtual bool StartRtcEventLog(rtc::PlatformFile,
>-+                                int64_t) {
>-     return false;
>-   }
>- 
>-@@ -928,8 +927,8 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // |output| and passes it on to Call, which will take the ownership. If the
>-   // operation fails the output will be closed and deallocated. The event log
>-   // will send serialized events to the output object every |output_period_ms|.
>--  virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
>--                                int64_t output_period_ms) {
>-+  virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput>,
>-+                                int64_t) {
>-     return false;
>-   }
>- 
>-@@ -1002,16 +1001,16 @@ class PeerConnectionObserver {
>-   // TODO(honghaiz): Make this a pure virtual method when all its subclasses
>-   // implement it.
>-   virtual void OnIceCandidatesRemoved(
>--      const std::vector<cricket::Candidate>& candidates) {}
>-+      const std::vector<cricket::Candidate>&) {}
>- 
>-   // Called when the ICE connection receiving status changes.
>--  virtual void OnIceConnectionReceivingChange(bool receiving) {}
>-+  virtual void OnIceConnectionReceivingChange(bool) {}
>- 
>-   // This is called when a receiver and its track is created.
>-   // TODO(zhihuang): Make this pure virtual when all subclasses implement it.
>-   virtual void OnAddTrack(
>--      rtc::scoped_refptr<RtpReceiverInterface> receiver,
>--      const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {}
>-+      rtc::scoped_refptr<RtpReceiverInterface>,
>-+      const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&) {}
>- 
>-   // TODO(hbos,deadbeef): Add |OnAssociatedStreamsUpdated| with |receiver| and
>-   // |streams| as arguments. This should be called when an existing receiver its
>-@@ -1030,7 +1029,7 @@ class PeerConnectionObserver {
>-   // no longer removed, deprecate and remove this callback.
>-   // TODO(hbos,deadbeef): Make pure virtual when all subclasses implement it.
>-   virtual void OnRemoveTrack(
>--      rtc::scoped_refptr<RtpReceiverInterface> receiver) {}
>-+      rtc::scoped_refptr<RtpReceiverInterface>) {}
>- 
>-  protected:
>-   // Dtor protected as objects shouldn't be deleted via this interface.
>-@@ -1133,7 +1132,7 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
>-   // TODO(deadbeef): Make pure virtual once downstream mock PC factory classes
>-   // are updated.
>-   virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
>--      std::unique_ptr<cricket::VideoCapturer> capturer) {
>-+      std::unique_ptr<cricket::VideoCapturer>) {
>-     return nullptr;
>-   }
>- 
>-@@ -1144,8 +1143,8 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
>-   // |constraints| is only used for the invocation of this method, and can
>-   // safely be destroyed afterwards.
>-   virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
>--      std::unique_ptr<cricket::VideoCapturer> capturer,
>--      const MediaConstraintsInterface* constraints) {
>-+      std::unique_ptr<cricket::VideoCapturer>,
>-+      const MediaConstraintsInterface*) {
>-     return nullptr;
>-   }
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/umametrics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/umametrics.h
>-index c5125985233..0d66a5e51fe 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/umametrics.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/umametrics.h
>-@@ -116,9 +116,9 @@ class MetricsObserverInterface : public rtc::RefCountInterface {
>-   // |type| is the type of the enum counter to be incremented. |counter|
>-   // is the particular counter in that type. |counter_max| is the next sequence
>-   // number after the highest counter.
>--  virtual void IncrementEnumCounter(PeerConnectionEnumCounterType type,
>--                                    int counter,
>--                                    int counter_max) {}
>-+  virtual void IncrementEnumCounter(PeerConnectionEnumCounterType,
>-+                                    int,
>-+                                    int) {}
>- 
>-   // This is used to handle sparse counters like SSL cipher suites.
>-   // TODO(guoweis): Remove the implementation once the dependency's interface
>-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 5897901346c..7d3a7f59781 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
>-@@ -36,7 +36,7 @@ class DecodedImageCallback {
>-   // decode time excluding waiting time for any previous pending frame to
>-   // return. This is necessary for breaking positive feedback in the delay
>-   // estimation when the decoder has a single output buffer.
>--  virtual int32_t Decoded(VideoFrame& decodedImage, int64_t decode_time_ms) {
>-+  virtual int32_t Decoded(VideoFrame& decodedImage, int64_t /* decode_time_ms */) {
>-     // The default implementation ignores custom decode time value.
>-     return Decoded(decodedImage);
>-   }
>-@@ -44,16 +44,16 @@ class DecodedImageCallback {
>-   // updated.
>-   virtual void Decoded(VideoFrame& decodedImage,
>-                        rtc::Optional<int32_t> decode_time_ms,
>--                       rtc::Optional<uint8_t> qp) {
>-+                       rtc::Optional<uint8_t> /* qp */) {
>-     Decoded(decodedImage,
>-             decode_time_ms ? static_cast<int32_t>(*decode_time_ms) : -1);
>-   }
>- 
>--  virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId) {
>-+  virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t /* pictureId */) {
>-     return -1;
>-   }
>- 
>--  virtual int32_t ReceivedDecodedFrame(const uint64_t pictureId) { return -1; }
>-+  virtual int32_t ReceivedDecodedFrame(const uint64_t /* pictureId */) { return -1; }
>- };
>- 
>- class VideoDecoder {
>-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 eb401afded9..64e33069099 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
>-@@ -74,7 +74,7 @@ class EncodedImageCallback {
>-   // Deprecated. TODO(ilnik): Remove this in few weeks.
>-   virtual void OnDroppedFrame() {}
>- 
>--  virtual void OnDroppedFrame(DropReason reason) {}
>-+  virtual void OnDroppedFrame(DropReason) {}
>- };
>- 
>- class VideoEncoder {
>-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
>-index 21019652870..0ab06a9a026 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/frame_callback.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/frame_callback.h
>-@@ -49,7 +49,7 @@ struct EncodedFrame {
>- class EncodedFrameObserver {
>-  public:
>-   virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) = 0;
>--  virtual void OnEncodeTiming(int64_t capture_ntp_ms, int encode_duration_ms) {}
>-+  virtual void OnEncodeTiming(int64_t, int) {}
>- 
>-  protected:
>-   virtual ~EncodedFrameObserver() {}
>-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 89b4fc1a086..89ace2c9e95 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
>-@@ -41,7 +41,7 @@ class RtcEventLog {
>-   // TODO(nisse): webrtc::Clock is deprecated. Delete this method and
>-   // above forward declaration of Clock when
>-   // webrtc/system_wrappers/include/clock.h is deleted.
>--  static std::unique_ptr<RtcEventLog> Create(const Clock* clock,
>-+  static std::unique_ptr<RtcEventLog> Create(const Clock*,
>-                                              EncodingType encoding_type) {
>-     return Create(encoding_type);
>-   }
>-@@ -65,12 +65,12 @@ class RtcEventLog {
>- // No-op implementation is used if flag is not set, or in tests.
>- class RtcEventLogNullImpl : public RtcEventLog {
>-  public:
>--  bool StartLogging(std::unique_ptr<RtcEventLogOutput> output,
>--                    int64_t output_period_ms) override {
>-+  bool StartLogging(std::unique_ptr<RtcEventLogOutput>,
>-+                    int64_t) override {
>-     return false;
>-   }
>-   void StopLogging() override {}
>--  void Log(std::unique_ptr<RtcEvent> event) override {}
>-+  void Log(std::unique_ptr<RtcEvent>) override {}
>- };
>- 
>- }  // namespace webrtc
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h
>-index 4826b2dc4a7..1a409615467 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h
>-@@ -1228,13 +1228,13 @@ class DataMediaChannel : public MediaChannel {
>-   virtual bool SetRecvParameters(const DataRecvParameters& params) = 0;
>- 
>-   // TODO(pthatcher): Implement this.
>--  virtual bool GetStats(DataMediaInfo* info) { return true; }
>-+  virtual bool GetStats(DataMediaInfo*) { return true; }
>- 
>-   virtual bool SetSend(bool send) = 0;
>-   virtual bool SetReceive(bool receive) = 0;
>- 
>--  virtual void OnNetworkRouteChanged(const std::string& transport_name,
>--                                     const rtc::NetworkRoute& network_route) {}
>-+  virtual void OnNetworkRouteChanged(const std::string&,
>-+                                     const rtc::NetworkRoute&) {}
>- 
>-   virtual bool SendData(
>-       const SendDataParams& params,
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device.h
>-index c096a8398cd..754c4e16ef9 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device.h
>-@@ -143,29 +143,29 @@ class AudioDeviceModule : public rtc::RefCountInterface {
>-   virtual int32_t SetStereoRecording(bool enable) = 0;
>-   virtual int32_t StereoRecording(bool* enabled) const = 0;
>-   // TODO(bugs.webrtc.org/7306): deprecated.
>--  virtual int32_t SetRecordingChannel(const ChannelType channel) { return -1; }
>--  virtual int32_t RecordingChannel(ChannelType* channel) const { return -1; }
>-+  virtual int32_t SetRecordingChannel(const ChannelType) { return -1; }
>-+  virtual int32_t RecordingChannel(ChannelType*) const { return -1; }
>- 
>-   // Playout delay
>-   virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;
>- 
>-   // TODO(bugs.webrtc.org/7306): deprecated (to be removed).
>--  virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) {
>-+  virtual int32_t SetRecordingSampleRate(const uint32_t) {
>-     return -1;
>-   }
>--  virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const {
>-+  virtual int32_t RecordingSampleRate(uint32_t*) const {
>-     return -1;
>-   }
>--  virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) {
>-+  virtual int32_t SetPlayoutSampleRate(const uint32_t) {
>-     return -1;
>-   }
>--  virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const {
>-+  virtual int32_t PlayoutSampleRate(uint32_t*) const {
>-     return -1;
>-   }
>- 
>-   // TODO(bugs.webrtc.org/7306): deprecated (to be removed).
>--  virtual int32_t SetLoudspeakerStatus(bool enable) { return -1; }
>--  virtual int32_t GetLoudspeakerStatus(bool* enabled) const { return -1; }
>-+  virtual int32_t SetLoudspeakerStatus(bool) { return -1; }
>-+  virtual int32_t GetLoudspeakerStatus(bool*) const { return -1; }
>- 
>-   // Only supported on Android.
>-   virtual bool BuiltInAECIsAvailable() const = 0;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.h
>-index 0a71162eea1..16f6b699a9b 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.h
>-@@ -62,7 +62,7 @@ class IceTransportInternal : public rtc::PacketTransportInternal {
>- 
>-   // TODO(zhihuang): Remove this once it's no longer called in
>-   // remoting/protocol/libjingle_transport_factory.cc
>--  virtual void SetIceProtocolType(IceProtocolType type) {}
>-+  virtual void SetIceProtocolType(IceProtocolType) {}
>- 
>-   virtual void SetIceCredentials(const std::string& ice_ufrag,
>-                                  const std::string& ice_pwd);
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h
>-index 2957fd8b742..380fd896933 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h
>-@@ -380,7 +380,7 @@ class Port : public PortInterface, public rtc::MessageHandler,
>-   }
>- 
>-   // Extra work to be done in subclasses when a connection is destroyed.
>--  virtual void HandleConnectionDestroyed(Connection* conn) {}
>-+  virtual void HandleConnectionDestroyed(Connection*) {}
>- 
>-  private:
>-   void Construct();
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h
>-index 6ff07b5766e..6ecd40d1779 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h
>-@@ -218,7 +218,7 @@ class StunAttribute {
>-   virtual StunAttributeValueType value_type() const = 0;
>- 
>-   // Only XorAddressAttribute needs this so far.
>--  virtual void SetOwner(StunMessage* owner) {}
>-+  virtual void SetOwner(StunMessage*) {}
>- 
>-   // Reads the body (not the type or length) for this type of attribute from
>-   // the given buffer.  Return value is true if successful.
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h
>-index 7bb04664089..9bca20ae20b 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h
>-@@ -118,11 +118,11 @@ class StunRequest : public rtc::MessageHandler {
>- 
>-   // Fills in a request object to be sent.  Note that request's transaction ID
>-   // will already be set and cannot be changed.
>--  virtual void Prepare(StunMessage* request) {}
>-+  virtual void Prepare(StunMessage*) {}
>- 
>-   // Called when the message receives a response or times out.
>--  virtual void OnResponse(StunMessage* response) {}
>--  virtual void OnErrorResponse(StunMessage* response) {}
>-+  virtual void OnResponse(StunMessage*) {}
>-+  virtual void OnErrorResponse(StunMessage*) {}
>-   virtual void OnTimeout() {}
>-   // Called when the message is sent.
>-   virtual void OnSent();
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.h
>-index ff953f7898d..dec3b8b86a0 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.h
>-@@ -40,7 +40,7 @@ class FunctorMessageHandler : public MessageHandler {
>-  public:
>-   explicit FunctorMessageHandler(FunctorT&& functor)
>-       : functor_(std::forward<FunctorT>(functor)) {}
>--  virtual void OnMessage(Message* msg) {
>-+  virtual void OnMessage(Message*) {
>-     result_ = functor_();
>-   }
>-   const ReturnT& result() const { return result_; }
>-@@ -60,7 +60,7 @@ class FunctorMessageHandler<void, FunctorT> : public MessageHandler {
>-  public:
>-   explicit FunctorMessageHandler(const FunctorT& functor)
>-       : functor_(functor) {}
>--  virtual void OnMessage(Message* msg) {
>-+  virtual void OnMessage(Message*) {
>-     functor_();
>-   }
>-   void result() const {}
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.h
>-index 5755699fcdf..a1b5c4fa68a 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.h
>-@@ -117,7 +117,7 @@ class NetworkManager : public DefaultLocalAddressProvider {
>-   // can optionally be called after GetNetworks.
>-   //
>-   // TODO(guoweis): remove this body when chromium implements this.
>--  virtual void GetAnyAddressNetworks(NetworkList* networks) {}
>-+  virtual void GetAnyAddressNetworks(NetworkList*) {}
>- 
>-   // Dumps the current list of networks in the network manager.
>-   virtual void DumpNetworks() {}
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions_impl.h
>-index 9b4f1c6483d..0174a83d2f5 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions_impl.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions_impl.h
>-@@ -101,7 +101,7 @@ struct RangeCheckImpl {};
>- // Dst range always contains the result: nothing to check.
>- template <typename Dst, typename Src, DstSign IsDstSigned, SrcSign IsSrcSigned>
>- struct RangeCheckImpl<Dst, Src, IsDstSigned, IsSrcSigned, CONTAINS_RANGE> {
>--  static RangeCheckResult Check(Src value) { return TYPE_VALID; }
>-+  static RangeCheckResult Check(Src) { return TYPE_VALID; }
>- };
>- 
>- // Signed to signed narrowing.
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketserver.h
>-index e29884e6b3c..38543693d37 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketserver.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketserver.h
>-@@ -35,7 +35,7 @@ class SocketServer : public SocketFactory {
>-   // When the socket server is installed into a Thread, this function is
>-   // called to allow the socket server to use the thread's message queue for
>-   // any messaging that it might need to perform.
>--  virtual void SetMessageQueue(MessageQueue* queue) {}
>-+  virtual void SetMessageQueue(MessageQueue*) {}
>- 
>-   // Sleeps until:
>-   //  1) cms milliseconds have elapsed (unless cms == kForever)
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.h
>-index 171a09f1e00..c959c228ea8 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.h
>-@@ -132,7 +132,7 @@ class StreamInterface : public MessageHandler {
>-   // processed.  Read and ConsumeReadData invalidate the buffer returned by
>-   // GetReadData.
>-   virtual const void* GetReadData(size_t* data_len);
>--  virtual void ConsumeReadData(size_t used) {}
>-+  virtual void ConsumeReadData(size_t) {}
>- 
>-   // 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
>-@@ -146,7 +146,7 @@ class StreamInterface : public MessageHandler {
>-   // 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) {}
>-+  virtual void ConsumeWriteBuffer(size_t) {}
>- 
>-   // 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
>--- 
>-2.13.5 (Apple Git-94)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0005-Fix-RTC_FATAL.patch b/Source/ThirdParty/libwebrtc/WebKit/0005-Fix-RTC_FATAL.patch
>deleted file mode 100644
>index 43aa5a00dcd..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0005-Fix-RTC_FATAL.patch
>+++ /dev/null
>@@ -1,537 +0,0 @@
>-From 62e7f44af723f2f7f28038a9f08355a29cb0dc24 Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Thu, 14 Dec 2017 13:56:05 -0800
>-Subject: [PATCH 5/8] Fix RTC_FATAL
>-
>----
>- .../api/audio_codecs/ilbc/audio_encoder_ilbc.cc    |  2 +-
>- .../libwebrtc/Source/webrtc/api/mediatypes.cc      |  4 +--
>- .../webrtc/common_video/video_frame_buffer.cc      |  2 +-
>- .../audio_coding/acm2/audio_coding_module.cc       |  2 +-
>- .../modules/audio_coding/acm2/rent_a_codec.cc      |  4 +--
>- .../audio_coding/codecs/cng/audio_encoder_cng.cc   |  2 +-
>- .../audio_coding/codecs/ilbc/audio_encoder_ilbc.cc |  4 +--
>- .../audio_coding/neteq/tools/neteq_rtpplay.cc      |  2 +-
>- .../modules/audio_coding/neteq/tools/neteq_test.cc |  4 +--
>- .../audio_coding/neteq/tools/rtp_file_source.cc    |  2 +-
>- .../audio_device/android/audio_device_template.h   | 30 +++++++++++-----------
>- .../audio_device/android/audio_record_jni.cc       |  2 +-
>- .../quality_assessment/fake_polqa.cc               |  2 +-
>- .../audio_processing/test/wav_based_simulator.cc   |  2 +-
>- .../libwebrtc/Source/webrtc/ortc/ortcfactory.cc    |  4 +--
>- .../Source/webrtc/ortc/rtpparametersconversion.cc  |  2 +-
>- .../libwebrtc/Source/webrtc/rtc_base/checks.h      |  6 ++---
>- .../libwebrtc/Source/webrtc/rtc_base/flags.cc      |  6 ++---
>- .../webrtc/rtc_base/numerics/safe_conversions.h    |  4 +--
>- .../Source/webrtc/rtc_base/opensslidentity.cc      |  8 +++---
>- .../Source/webrtc/test/rtp_file_reader.cc          |  4 +--
>- .../Source/webrtc/voice_engine/utility.cc          |  4 +--
>- 22 files changed, 51 insertions(+), 51 deletions(-)
>-
>-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 a7c68ffcf02..6db88402e39 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/common_video/video_frame_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc
>-index 88c8ca4a5d4..c05f72e6d17 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
>-@@ -253,7 +253,7 @@ 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/audio_coding_module.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
>-index 18325f41590..f9c9690e98e 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
>-@@ -1145,7 +1145,7 @@ int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
>-       app = AudioEncoder::Application::kAudio;
>-       break;
>-     default:
>--      FATAL();
>-+      RTC_FATAL();
>-       return 0;
>-   }
>-   return encoder_stack_->SetApplication(app) ? 0 : -1;
>-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 78db38dad60..4a1746e3da9 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
>-@@ -215,7 +215,7 @@ 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)));
>- }
>-@@ -230,7 +230,7 @@ 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 78148ab3504..5a043ecbccb 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 6ddc078f095..e108c412ceb 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();
>-   }
>- }
>- 
>-@@ -148,7 +148,7 @@ size_t AudioEncoderIlbcImpl::RequiredOutputSizeBytes() const {
>-     case 3:   return 50;
>-     case 4:   return 2 * 38;
>-     case 6:   return 2 * 50;
>--    default:  FATAL();
>-+    default:  RTC_FATAL();
>-   }
>- }
>- 
>-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 2657bfaeb5b..4137f8e0c4f 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
>-@@ -171,7 +171,7 @@ std::string CodecName(NetEqDecoder codec) {
>-     case NetEqDecoder::kDecoderCNGswb48kHz:
>-       return "comfort noise (48 kHz)";
>-     default:
>--      FATAL();
>-+      RTC_FATAL();
>-       return "undefined";
>-   }
>- }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc
>-index 39d95495779..295111a97d9 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc
>-@@ -21,12 +21,12 @@ void DefaultNetEqTestErrorCallback::OnInsertPacketError(
>-     const NetEqInput::PacketData& packet) {
>-   std::cerr << "InsertPacket returned an error." << std::endl;
>-   std::cerr << "Packet data: " << packet.ToString() << std::endl;
>--  FATAL();
>-+  RTC_FATAL();
>- }
>- 
>- void DefaultNetEqTestErrorCallback::OnGetAudioError() {
>-   std::cerr << "GetAudio returned an error." << std::endl;
>--  FATAL();
>-+  RTC_FATAL();
>- }
>- 
>- NetEqTest::NetEqTest(const NetEq::Config& config,
>-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 c9ae5f2e714..d1b1008d704 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
>-@@ -93,7 +93,7 @@ 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/modules/audio_device/android/audio_device_template.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_device_template.h
>-index 75dbaf315f5..a4ef5dfa310 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_device_template.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_device_template.h
>-@@ -103,14 +103,14 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
>-   int32_t PlayoutDeviceName(uint16_t index,
>-                             char name[kAdmMaxDeviceNameSize],
>-                             char guid[kAdmMaxGuidSize]) override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-   int32_t RecordingDeviceName(uint16_t index,
>-                               char name[kAdmMaxDeviceNameSize],
>-                               char guid[kAdmMaxGuidSize]) override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-@@ -123,7 +123,7 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
>- 
>-   int32_t SetPlayoutDevice(
>-       AudioDeviceModule::WindowsDeviceType device) override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-@@ -136,7 +136,7 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
>- 
>-   int32_t SetRecordingDevice(
>-       AudioDeviceModule::WindowsDeviceType device) override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-@@ -217,7 +217,7 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
>- 
>-   int32_t SetAGC(bool enable) override {
>-     if (enable) {
>--      FATAL() << "Should never be called";
>-+      RTC_FATAL() << "Should never be called";
>-     }
>-     return -1;
>-   }
>-@@ -278,52 +278,52 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
>-   }
>- 
>-   int32_t SetMicrophoneVolume(uint32_t volume) override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-   int32_t MicrophoneVolume(uint32_t& volume) const override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-   int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-   int32_t MinMicrophoneVolume(uint32_t& minVolume) const override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-   int32_t SpeakerMuteIsAvailable(bool& available) override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-   int32_t SetSpeakerMute(bool enable) override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-   int32_t SpeakerMute(bool& enabled) const override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-   int32_t MicrophoneMuteIsAvailable(bool& available) override {
>--    FATAL() << "Not implemented";
>-+    RTC_FATAL() << "Not implemented";
>-     return -1;
>-   }
>- 
>-   int32_t SetMicrophoneMute(bool enable) override {
>--    FATAL() << "Not implemented";
>-+    RTC_FATAL() << "Not implemented";
>-     return -1;
>-   }
>- 
>-   int32_t MicrophoneMute(bool& enabled) const override {
>--    FATAL() << "Not implemented";
>-+    RTC_FATAL() << "Not implemented";
>-     return -1;
>-   }
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_record_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_record_jni.cc
>-index 79f8c8b0f9b..b8a564f06f2 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_record_jni.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_record_jni.cc
>-@@ -196,7 +196,7 @@ int32_t AudioRecordJni::EnableBuiltInAEC(bool enable) {
>- 
>- int32_t AudioRecordJni::EnableBuiltInAGC(bool enable) {
>-   // TODO(henrika): possibly remove when no longer used by any client.
>--  FATAL() << "Should never be called";
>-+  RTC_FATAL() << "Should never be called";
>-   return -1;
>- }
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc
>-index 62d8ebb84d9..bae652e2835 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc
>-@@ -43,7 +43,7 @@ int main(int argc, char* argv[]) {
>-       return 0;
>-     }
>-   }
>--  FATAL() << kErrorMessage;
>-+  RTC_FATAL() << kErrorMessage;
>- }
>- 
>- }  // namespace test
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/wav_based_simulator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/wav_based_simulator.cc
>-index f53d1e5d8d9..6ed8e529416 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/wav_based_simulator.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/wav_based_simulator.cc
>-@@ -40,7 +40,7 @@ WavBasedSimulator::GetCustomEventChain(const std::string& filename) {
>-       case '\n':
>-         break;
>-       default:
>--        FATAL() << "Incorrect custom call order file, reverting to using the "
>-+        RTC_FATAL() << "Incorrect custom call order file, reverting to using the "
>-                    "default call order";
>-         fclose(stream);
>-         return WavBasedSimulator::GetDefaultEventChain();
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.cc
>-index 7492dd47aae..b9f0f61e5d8 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.cc
>-@@ -332,7 +332,7 @@ RtpCapabilities OrtcFactory::GetRtpSenderCapabilities(
>-       return RtpCapabilities();
>-   }
>-   // Not reached; avoids compile warning.
>--  FATAL();
>-+  RTC_FATAL();
>- }
>- 
>- RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>>
>-@@ -397,7 +397,7 @@ RtpCapabilities OrtcFactory::GetRtpReceiverCapabilities(
>-       return RtpCapabilities();
>-   }
>-   // Not reached; avoids compile warning.
>--  FATAL();
>-+  RTC_FATAL();
>- }
>- 
>- RTCErrorOr<std::unique_ptr<OrtcRtpReceiverInterface>>
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion.cc
>-index 0a657eca8f3..bc702909e34 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion.cc
>-@@ -62,7 +62,7 @@ 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 e9a19e6051f..cf2965fd77e 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h
>-@@ -73,7 +73,7 @@ NO_RETURN 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.
>-@@ -135,7 +135,7 @@ std::string* MakeCheckOpString(const t1& v1, const t2& v2, const char* names) {
>- }
>- 
>- // MSVC doesn't like complex extern templates and DLLs.
>--#if !defined(COMPILER_MSVC)
>-+#if 0
>- // Commonly used instantiations of MakeCheckOpString<>. Explicitly instantiated
>- // in logging.cc.
>- extern template std::string* MakeCheckOpString<int, int>(
>-@@ -222,7 +222,7 @@ class FatalMessageVoidify {
>- #define RTC_NOTREACHED() RTC_DCHECK(RTC_UNREACHABLE_CODE_HIT)
>- 
>- // TODO(bugs.webrtc.org/8454): Add an RTC_ prefix or rename differently.
>--#define FATAL() rtc::FatalMessage(__FILE__, __LINE__).stream()
>-+#define RTC_FATAL() rtc::FatalMessage(__FILE__, __LINE__).stream()
>- // TODO(ajm): Consider adding RTC_NOTIMPLEMENTED macro when
>- // base/logging.h and system_wrappers/logging.h are consolidated such that we
>- // can match the Chromium behavior.
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>-index 4f2b11a9624..1c24a76ace4 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>-@@ -57,7 +57,7 @@ void Flag::SetToDefault() {
>-       variable_->s = default_.s;
>-       return;
>-   }
>--  FATAL() << "unreachable code";
>-+  RTC_FATAL() << "unreachable code";
>- }
>- 
>- 
>-@@ -68,7 +68,7 @@ static const char* Type2String(Flag::Type type) {
>-     case Flag::FLOAT: return "float";
>-     case Flag::STRING: return "string";
>-   }
>--  FATAL() << "unreachable code";
>-+  RTC_FATAL() << "unreachable code";
>- }
>- 
>- 
>-@@ -87,7 +87,7 @@ static void PrintFlagValue(Flag::Type type, FlagValue* p) {
>-       printf("%s", p->s);
>-       return;
>-   }
>--  FATAL() << "unreachable code";
>-+  RTC_FATAL() << "unreachable code";
>- }
>- 
>- 
>-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/opensslidentity.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.cc
>-index 85ef176de09..bfb167aacef 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.cc
>-@@ -411,11 +411,11 @@ 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;
>-@@ -432,11 +432,11 @@ 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/test/rtp_file_reader.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.cc
>-index 03674875432..8d7207d4e8a 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.cc
>-@@ -90,7 +90,7 @@ class InterleavedRtpFileReader : public RtpFileReaderImpl {
>-     uint32_t len = 0;
>-     TRY(ReadUint32(&len, file_));
>-     if (packet->length < len) {
>--      FATAL() << "Packet is too large to fit: " << len << " bytes vs "
>-+      RTC_FATAL() << "Packet is too large to fit: " << len << " bytes vs "
>-               << packet->length
>-               << " bytes allocated. Consider increasing the buffer "
>-                  "size";
>-@@ -178,7 +178,7 @@ class RtpDumpReader : public RtpFileReaderImpl {
>-     // Use 'len' here because a 'plen' of 0 specifies rtcp.
>-     len -= kPacketHeaderSize;
>-     if (packet->length < len) {
>--      FATAL() << "Packet is too large to fit: " << len << " bytes vs "
>-+      RTC_FATAL() << "Packet is too large to fit: " << len << " bytes vs "
>-               << packet->length
>-               << " bytes allocated. Consider increasing the buffer "
>-                  "size";
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility.cc
>-index 939870245f1..5caa73ae451 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility.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,
>-                                        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();
>-   }
>--- 
>-2.13.5 (Apple Git-94)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0006-Disabling-VP8.patch b/Source/ThirdParty/libwebrtc/WebKit/0006-Disabling-VP8.patch
>deleted file mode 100644
>index 9773bf0dab5..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0006-Disabling-VP8.patch
>+++ /dev/null
>@@ -1,100 +0,0 @@
>-From 201c211d31aa1e9e3b3cef70feff291512efde6d Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Thu, 14 Dec 2017 13:56:52 -0800
>-Subject: [PATCH 6/8] Disabling VP8
>-
>----
>- .../libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc  | 3 ++-
>- .../libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc  | 3 ++-
>- .../libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc       | 6 +++++-
>- .../Source/webrtc/modules/video_coding/video_codec_initializer.cc   | 6 ++++--
>- 4 files changed, 13 insertions(+), 5 deletions(-)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc
>-index e8cecb79a43..57daca256c4 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc
>-@@ -22,7 +22,8 @@ namespace webrtc {
>- std::vector<SdpVideoFormat> InternalDecoderFactory::GetSupportedFormats()
>-     const {
>-   std::vector<SdpVideoFormat> formats;
>--  formats.push_back(SdpVideoFormat(cricket::kVp8CodecName));
>-+  if (VP8Decoder::IsSupported())
>-+    formats.push_back(SdpVideoFormat(cricket::kVp8CodecName));
>-   if (VP9Decoder::IsSupported())
>-     formats.push_back(SdpVideoFormat(cricket::kVp9CodecName));
>-   for (const SdpVideoFormat& h264_format : SupportedH264Codecs())
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc
>-index 5f8479cfb82..0390646add4 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc
>-@@ -22,7 +22,8 @@ namespace webrtc {
>- std::vector<SdpVideoFormat> InternalEncoderFactory::GetSupportedFormats()
>-     const {
>-   std::vector<SdpVideoFormat> supported_codecs;
>--  supported_codecs.push_back(SdpVideoFormat(cricket::kVp8CodecName));
>-+  if (VP8Encoder::IsSupported())
>-+    supported_codecs.push_back(SdpVideoFormat(cricket::kVp8CodecName));
>-   if (webrtc::VP9Encoder::IsSupported())
>-     supported_codecs.push_back(SdpVideoFormat(cricket::kVp9CodecName));
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc
>-index 2adbca9b712..418b8b7e5ff 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc
>-@@ -17,13 +17,17 @@
>- namespace webrtc {
>- 
>- bool QpParser::GetQp(const VCMEncodedFrame& frame, int* qp) {
>--  switch (frame.CodecSpecific()->codecType) {
>-+    switch (frame.CodecSpecific()->codecType) {
>-+#if !RTC_DISABLE_VP8
>-     case kVideoCodecVP8:
>-       // QP range: [0, 127].
>-       return vp8::GetQp(frame.Buffer(), frame.Length(), qp);
>-+#endif
>-+#if !RTC_DISABLE_VP9
>-     case kVideoCodecVP9:
>-       // QP range: [0, 255].
>-       return vp9::GetQp(frame.Buffer(), frame.Length(), qp);
>-+#endif
>-     default:
>-       return false;
>-   }
>-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 3e1b0e1163b..89f6f0bad9d 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
>-@@ -42,7 +42,7 @@ bool VideoCodecInitializer::SetupCodec(
>-   *codec =
>-       VideoEncoderConfigToVideoCodec(config, streams, settings.payload_name,
>-                                      settings.payload_type, nack_enabled);
>--
>-+#if !defined(RTC_DISABLE_VP8)
>-   std::unique_ptr<TemporalLayersFactory> tl_factory;
>-   switch (codec->codecType) {
>-     case kVideoCodecVP8: {
>-@@ -68,7 +68,7 @@ bool VideoCodecInitializer::SetupCodec(
>-     }
>-   }
>-   *bitrate_allocator = CreateBitrateAllocator(*codec, std::move(tl_factory));
>--
>-+#endif
>-   return true;
>- }
>- 
>-@@ -80,9 +80,11 @@ VideoCodecInitializer::CreateBitrateAllocator(
>- 
>-   switch (codec.codecType) {
>-     case kVideoCodecVP8: {
>-+#if !defined(RTC_DISABLE_VP8)
>-       // Set up default VP8 temporal layer factory, if not provided.
>-       rate_allocator.reset(
>-           new SimulcastRateAllocator(codec, std::move(tl_factory)));
>-+#endif
>-     } break;
>-     default:
>-       rate_allocator.reset(new DefaultVideoBitrateAllocator(codec));
>--- 
>-2.13.5 (Apple Git-94)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0007-Fix-RTC_STRINGIZE.patch b/Source/ThirdParty/libwebrtc/WebKit/0007-Fix-RTC_STRINGIZE.patch
>deleted file mode 100644
>index 2a56c84bb4e..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0007-Fix-RTC_STRINGIZE.patch
>+++ /dev/null
>@@ -1,38 +0,0 @@
>-From efa6c8d735872ad827a9ec5c3c7e09ab5cfac1c1 Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Thu, 14 Dec 2017 13:57:16 -0800
>-Subject: [PATCH 7/8] Fix RTC_STRINGIZE
>-
>----
>- Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.h         | 2 +-
>- Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h | 2 +-
>- 2 files changed, 2 insertions(+), 2 deletions(-)
>-
>-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/stringize_macros.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>-index aee8d14551d..acf0ba404ca 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>-@@ -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_
>--- 
>-2.13.5 (Apple Git-94)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0008-Fix-sanitizer.patch b/Source/ThirdParty/libwebrtc/WebKit/0008-Fix-sanitizer.patch
>deleted file mode 100644
>index 18ae7715c54..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0008-Fix-sanitizer.patch
>+++ /dev/null
>@@ -1,60 +0,0 @@
>-From 826df71603f4e35466cb5330bd18920d503e80b2 Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Thu, 14 Dec 2017 13:57:39 -0800
>-Subject: [PATCH 8/8] Fix sanitizer
>-
>----
>- .../libwebrtc/Source/webrtc/rtc_base/sanitizer.h         | 16 ++++++++++++++++
>- 1 file changed, 16 insertions(+)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer.h
>-index 1b94e1edf47..96d024883fa 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer.h
>-@@ -52,6 +52,10 @@ static inline void rtc_AsanPoison(const volatile void* ptr,
>-                                   size_t num_elements) {
>- #if RTC_HAS_ASAN
>-   ASAN_POISON_MEMORY_REGION(ptr, element_size * num_elements);
>-+#else
>-+  (void)ptr;
>-+  (void)element_size;
>-+  (void)num_elements;
>- #endif
>- }
>- 
>-@@ -63,6 +67,10 @@ static inline void rtc_AsanUnpoison(const volatile void* ptr,
>-                                     size_t num_elements) {
>- #if RTC_HAS_ASAN
>-   ASAN_UNPOISON_MEMORY_REGION(ptr, element_size * num_elements);
>-+#else
>-+  (void)ptr;
>-+  (void)element_size;
>-+  (void)num_elements;
>- #endif
>- }
>- 
>-@@ -73,6 +81,10 @@ static inline void rtc_MsanMarkUninitialized(const volatile void* ptr,
>-                                              size_t num_elements) {
>- #if RTC_HAS_MSAN
>-   __msan_poison(ptr, element_size * num_elements);
>-+#else
>-+  (void)ptr;
>-+  (void)element_size;
>-+  (void)num_elements;
>- #endif
>- }
>- 
>-@@ -84,6 +96,10 @@ static inline void rtc_MsanCheckInitialized(const volatile void* ptr,
>-                                             size_t num_elements) {
>- #if RTC_HAS_MSAN
>-   __msan_check_mem_is_initialized(ptr, element_size * num_elements);
>-+#else
>-+  (void)ptr;
>-+  (void)element_size;
>-+  (void)num_elements;
>- #endif
>- }
>- 
>--- 
>-2.13.5 (Apple Git-94)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0009-Remove-dispatch_set_target_queue.patch b/Source/ThirdParty/libwebrtc/WebKit/0009-Remove-dispatch_set_target_queue.patch
>deleted file mode 100644
>index c61406951b6..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0009-Remove-dispatch_set_target_queue.patch
>+++ /dev/null
>@@ -1,23 +0,0 @@
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_gcd.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_gcd.cc
>-index a13e088ede6..675b8005722 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_gcd.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_gcd.cc
>-@@ -139,7 +139,7 @@ class TaskQueue::Impl : public RefCountInterface {
>- TaskQueue::Impl::Impl(const char* queue_name,
>-                       TaskQueue* task_queue,
>-                       Priority priority)
>--    : queue_(dispatch_queue_create(queue_name, DISPATCH_QUEUE_SERIAL)),
>-+    : queue_(dispatch_queue_create_with_target(queue_name, DISPATCH_QUEUE_SERIAL, dispatch_get_global_queue(TaskQueuePriorityToGCD(priority), 0))),
>-       context_(new QueueContext(task_queue)) {
>-   RTC_DCHECK(queue_name);
>-   RTC_CHECK(queue_);
>-@@ -148,9 +148,6 @@ TaskQueue::Impl::Impl(const char* queue_name,
>-   // to the queue is released.  This may run after the TaskQueue object has
>-   // been deleted.
>-   dispatch_set_finalizer_f(queue_, &QueueContext::DeleteContext);
>--
>--  dispatch_set_target_queue(
>--      queue_, dispatch_get_global_queue(TaskQueuePriorityToGCD(priority), 0));
>- }
>- 
>- TaskQueue::Impl::~Impl() {
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0010-Fix-RTCVideoEncoderH264-CVPixelBuffer-leak.patch b/Source/ThirdParty/libwebrtc/WebKit/0010-Fix-RTCVideoEncoderH264-CVPixelBuffer-leak.patch
>deleted file mode 100644
>index 83098c349c6..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0010-Fix-RTCVideoEncoderH264-CVPixelBuffer-leak.patch
>+++ /dev/null
>@@ -1,12 +0,0 @@
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-index 42f940426ae..16a841d9458 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-@@ -429,6 +429,7 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>-       }
>-       _frameScaleBuffer.shrink_to_fit();
>-       if (![rtcPixelBuffer cropAndScaleTo:pixelBuffer withTempBuffer:_frameScaleBuffer.data()]) {
>-+        CVBufferRelease(pixelBuffer);
>-         return WEBRTC_VIDEO_CODEC_ERROR;
>-       }
>-     }
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0011-Fix-AudioDeviceID-array-leak.patch b/Source/ThirdParty/libwebrtc/WebKit/0011-Fix-AudioDeviceID-array-leak.patch
>deleted file mode 100644
>index 0bed7f48b6c..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0011-Fix-AudioDeviceID-array-leak.patch
>+++ /dev/null
>@@ -1,56 +0,0 @@
>-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 c585c32cafc..89ea9317c1f 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
>-@@ -19,6 +19,7 @@
>- #include <ApplicationServices/ApplicationServices.h>
>- #include <libkern/OSAtomic.h>  // OSAtomicCompareAndSwap()
>- #include <mach/mach.h>         // mach_task_self()
>-+#include <memory>              // std::make_unique<>() and std::unique_ptr<>()
>- #include <sys/sysctl.h>        // sysctlbyname()
>- 
>- namespace webrtc {
>-@@ -1564,8 +1565,8 @@ int32_t AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
>-     return 0;
>-   }
>- 
>--  AudioDeviceID* deviceIds = (AudioDeviceID*)malloc(size);
>-   UInt32 numberDevices = size / sizeof(AudioDeviceID);
>-+  auto deviceIds = std::make_unique<AudioDeviceID[]>(numberDevices);
>-   AudioBufferList* bufferList = NULL;
>-   UInt32 numberScopedDevices = 0;
>- 
>-@@ -1597,7 +1598,7 @@ int32_t AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
>-   bool listOK = true;
>- 
>-   WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(
>--      kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size, deviceIds));
>-+      kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size, deviceIds.get()));
>-   if (err != noErr) {
>-     listOK = false;
>-   } else {
>-@@ -1641,11 +1642,6 @@ int32_t AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
>-   }
>- 
>-   if (!listOK) {
>--    if (deviceIds) {
>--      free(deviceIds);
>--      deviceIds = NULL;
>--    }
>--
>-     if (bufferList) {
>-       free(bufferList);
>-       bufferList = NULL;
>-@@ -1654,12 +1650,6 @@ int32_t AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
>-     return -1;
>-   }
>- 
>--  // Happy ending
>--  if (deviceIds) {
>--    free(deviceIds);
>--    deviceIds = NULL;
>--  }
>--
>-   return numberScopedDevices;
>- }
>- 
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0012-Add-WK-prefix-to-Objective-C-classes-and-protocols.patch b/Source/ThirdParty/libwebrtc/WebKit/0012-Add-WK-prefix-to-Objective-C-classes-and-protocols.patch
>deleted file mode 100644
>index 5d78ef037a3..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0012-Add-WK-prefix-to-Objective-C-classes-and-protocols.patch
>+++ /dev/null
>@@ -1,1076 +0,0 @@
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>-index 1490c89ac5f..5e65230f321 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>-@@ -33,6 +33,7 @@
>- #include <webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h>
>- 
>- #if !defined(WEBRTC_IOS)
>-+__attribute__((objc_runtime_name("WK_RTCUIApplicationStatusObserver")))
>- @interface RTCUIApplicationStatusObserver : NSObject
>- 
>- + (instancetype)sharedInstance;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h
>-index a2064df5db9..8ee12b62499 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h
>-@@ -12,6 +12,7 @@
>- 
>- #import <Foundation/Foundation.h>
>- 
>-+__attribute__((objc_runtime_name("WK_RTCUIApplicationStatusObserver")))
>- @interface RTCUIApplicationStatusObserver : NSObject
>- 
>- + (instancetype)sharedInstance;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.h
>-index e5987fe22a2..315b1f5a8da 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.h
>-@@ -13,5 +13,6 @@
>- #import "RTCMTLRenderer.h"
>- 
>- NS_AVAILABLE(10_11, 9_0)
>-+__attribute__((objc_runtime_name("WK_RTCMTLI420Renderer")))
>- @interface RTCMTLI420Renderer : RTCMTLRenderer
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.h
>-index 866b7ea17ee..8bbcb204d25 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.h
>-@@ -13,6 +13,7 @@
>- #import "RTCMTLRenderer.h"
>- 
>- NS_AVAILABLE(10_11, 9_0)
>-+__attribute__((objc_runtime_name("WK_RTCMTLNV12Renderer")))
>- @interface RTCMTLNV12Renderer : RTCMTLRenderer
>- 
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h
>-index e279b098b55..110e2124656 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h
>-@@ -21,6 +21,7 @@ NS_ASSUME_NONNULL_BEGIN
>- /**
>-  * Protocol defining ability to render RTCVideoFrame in Metal enabled views.
>-  */
>-+__attribute__((objc_runtime_name("WK_RTCMTLRenderer")))
>- @protocol RTCMTLRenderer<NSObject>
>- 
>- /**
>-@@ -49,6 +50,7 @@ NS_ASSUME_NONNULL_BEGIN
>-  * Implementation of RTCMTLRenderer protocol for rendering native nv12 video frames.
>-  */
>- NS_AVAILABLE(10_11, 9_0)
>-+__attribute__((objc_runtime_name("WK_RTCMTLRenderer")))
>- @interface RTCMTLRenderer : NSObject<RTCMTLRenderer>
>- @end
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDtmfSender+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDtmfSender+Private.h
>-index 197aeb83a42..04f10e8b399 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDtmfSender+Private.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDtmfSender+Private.h
>-@@ -14,6 +14,7 @@
>- 
>- NS_ASSUME_NONNULL_BEGIN
>- 
>-+__attribute__((objc_runtime_name("WK_RTCDtmfSender")))
>- @interface RTCDtmfSender : NSObject <RTCDtmfSender>
>- 
>- @property(nonatomic, readonly) rtc::scoped_refptr<webrtc::DtmfSenderInterface> nativeDtmfSender;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.h
>-index b0b6f044884..5b4570310d4 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.h
>-@@ -18,6 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
>-  * adapter adapts calls made to that interface to the RTCVideoRenderer supplied
>-  * during construction.
>-  */
>-+__attribute__((objc_runtime_name("WK_RTCVideoRendererAdapter")))
>- @interface RTCVideoRendererAdapter : NSObject
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.h
>-index 74cfaeb861f..cb29b692640 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.h
>-@@ -14,6 +14,7 @@
>- #include "api/video_codecs/video_decoder.h"
>- #include "media/base/codec.h"
>- 
>-+__attribute__((objc_runtime_name("WK_RTCWrappedNativeVideoDecoder")))
>- @interface RTCWrappedNativeVideoDecoder : NSObject <RTCVideoDecoder>
>- 
>- - (instancetype)initWithNativeDecoder:(std::unique_ptr<webrtc::VideoDecoder>)decoder;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.h
>-index 5b95d33eba9..29eff125a54 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.h
>-@@ -15,6 +15,7 @@
>- #include "api/video_codecs/video_encoder.h"
>- #include "media/base/codec.h"
>- 
>-+__attribute__((objc_runtime_name("WK_RTCWrappedNativeVideoEncoder")))
>- @interface RTCWrappedNativeVideoEncoder : NSObject <RTCVideoEncoder>
>- 
>- - (instancetype)initWithNativeEncoder:(std::unique_ptr<webrtc::VideoEncoder>)encoder;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/UI/RTCEAGLVideoView.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/UI/RTCEAGLVideoView.m
>-index 8f379c864ae..8f4da302fab 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/UI/RTCEAGLVideoView.m
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/UI/RTCEAGLVideoView.m
>-@@ -23,6 +23,7 @@
>- // refreshes, which should be 30fps. We wrap the display link in order to avoid
>- // a retain cycle since CADisplayLink takes a strong reference onto its target.
>- // The timer is paused by default.
>-+__attribute__((objc_runtime_name("WK_RTCDisplayLinkTimer")))
>- @interface RTCDisplayLinkTimer : NSObject
>- 
>- @property(nonatomic) BOOL isPaused;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.h
>-index 8b73a808639..217437aff2d 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.h
>-@@ -21,6 +21,7 @@ NS_ASSUME_NONNULL_BEGIN
>- // It is meant to be owned by an instance of AVFoundationVideoCapturer.
>- // The reason for this is because other webrtc objects own cricket::VideoCapturer, which is not
>- // ref counted. To prevent bad behavior we do not expose this class directly.
>-+__attribute__((objc_runtime_name("WK_RTCAVFoundationVideoCapturerInternal")))
>- @interface RTCAVFoundationVideoCapturerInternal
>-     : NSObject <AVCaptureVideoDataOutputSampleBufferDelegate>
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.h
>-index ed5125e2e8e..cc2ff4e46be 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.h
>-@@ -16,6 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
>-  *  RTCEAGLVideoView if no external shader is specified. This shader will render
>-  *  the video in a rectangle without any color or geometric transformations.
>-  */
>-+__attribute__((objc_runtime_name("WK_RTCDefaultShader")))
>- @interface RTCDefaultShader : NSObject<RTCVideoViewShading>
>- 
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCI420TextureCache.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCI420TextureCache.h
>-index 19f1bc99934..6726a8842b7 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCI420TextureCache.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCI420TextureCache.h
>-@@ -11,6 +11,7 @@
>- #import "RTCOpenGLDefines.h"
>- #import "WebRTC/RTCVideoFrame.h"
>- 
>-+__attribute__((objc_runtime_name("WK_RTCI420TextureCache")))
>- @interface RTCI420TextureCache : NSObject
>- 
>- @property(nonatomic, readonly) GLuint yTexture;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h
>-index 9cba8232714..d59b73c9a93 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h
>-@@ -14,6 +14,7 @@
>- 
>- NS_ASSUME_NONNULL_BEGIN
>- 
>-+__attribute__((objc_runtime_name("WK_RTCNV12TextureCache")))
>- @interface RTCNV12TextureCache : NSObject
>- 
>- @property(nonatomic, readonly) GLuint yTexture;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h
>-index 85a89e6dd9a..163028eb29d 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h
>-@@ -15,6 +15,7 @@
>- 
>- #include "common_video/include/video_frame_buffer.h"
>- 
>-+__attribute__((objc_runtime_name("WK_RTCVideoFrameBuffer")))
>- @protocol RTCVideoFrameBuffer;
>- 
>- namespace webrtc {
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.h
>-index a0106ff83fe..13467e3898e 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.h
>-@@ -15,6 +15,7 @@
>- #include "media/base/codec.h"
>- #include "media/engine/webrtcvideodecoderfactory.h"
>- 
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderFactory")))
>- @protocol RTCVideoDecoderFactory;
>- 
>- namespace webrtc {
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.h
>-index 6cdfc56f548..727922bfe4e 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.h
>-@@ -16,6 +16,7 @@
>- #include "api/video_codecs/video_encoder_factory.h"
>- #include "media/engine/webrtcvideoencoderfactory.h"
>- 
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderFactory")))
>- @protocol RTCVideoEncoderFactory;
>- 
>- namespace webrtc {
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h
>-index 6d369b34034..de73b477099 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h
>-@@ -27,6 +27,7 @@ NS_ASSUME_NONNULL_BEGIN
>-  * counted video source interface.
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCAVFoundationVideoSource")))
>- @interface RTCAVFoundationVideoSource : RTCVideoSource
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
>-index 354634ec39f..e44aa39ae55 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
>-@@ -28,6 +28,7 @@ extern NSInteger const kRTCAudioSessionErrorConfiguration;
>- // from AVAudioSession and handle them before calling these delegate methods,
>- // at which point applications can perform additional processing if required.
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCAudioSessionDelegate")))
>- @protocol RTCAudioSessionDelegate <NSObject>
>- 
>- @optional
>-@@ -91,6 +92,7 @@ RTC_EXPORT
>-  *  case of this is when CallKit activates the audio session for the application
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCAudioSessionActivationDelegate")))
>- @protocol RTCAudioSessionActivationDelegate <NSObject>
>- 
>- /** Called when the audio session is activated outside of the app by iOS. */
>-@@ -109,6 +111,7 @@ RTC_EXPORT
>-  *  activated only once. See |setActive:error:|.
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCAudioSession")))
>- @interface RTCAudioSession : NSObject <RTCAudioSessionActivationDelegate>
>- 
>- /** Convenience property to access the AVAudioSession singleton. Callers should
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h
>-index 6a02751d294..0097246e7c8 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h
>-@@ -23,6 +23,7 @@ extern const double kRTCAudioSessionLowComplexityIOBufferDuration;
>- 
>- // Struct to hold configuration values.
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCAudioSessionConfiguration")))
>- @interface RTCAudioSessionConfiguration : NSObject
>- 
>- @property(nonatomic, strong) NSString *category;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h
>-index 3db31573cb8..7287c31a738 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h
>-@@ -16,6 +16,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCAudioSource")))
>- @interface RTCAudioSource : RTCMediaSource
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
>-index b83eede528c..bab1b1a0020 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
>-@@ -16,6 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
>- @class RTCAudioSource;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCAudioTrack")))
>- @interface RTCAudioTrack : RTCMediaStreamTrack
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
>-index 9018aec3d18..9f9a59fc87d 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
>-@@ -20,6 +20,7 @@
>-  *  AVCaptureSession.
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCCameraPreviewView")))
>- @interface RTCCameraPreviewView : UIView
>- 
>- /** The capture session being rendered in the view. Capture session
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h
>-index 31b405c5426..b2e012778f3 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h
>-@@ -19,6 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
>- RTC_EXPORT
>- // Camera capture that implements RTCVideoCapturer. Delivers frames to a RTCVideoCapturerDelegate
>- // (usually RTCVideoSource).
>-+__attribute__((objc_runtime_name("WK_RTCCameraVideoCapturer")))
>- @interface RTCCameraVideoCapturer : RTCVideoCapturer
>- 
>- // Capture session that is used for capturing. Valid from initialization to dealloc.
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
>-index fcb62b94168..bf8e50c0504 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
>-@@ -66,6 +66,7 @@ typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCConfiguration")))
>- @interface RTCConfiguration : NSObject
>- 
>- /** An array of Ice Servers available to be used by ICE. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
>-index 893bd0a3365..1785513753a 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
>-@@ -16,6 +16,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDataBuffer")))
>- @interface RTCDataBuffer : NSObject
>- 
>- /** NSData representation of the underlying buffer. */
>-@@ -37,6 +38,7 @@ RTC_EXPORT
>- 
>- @class RTCDataChannel;
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDataChannelDelegate")))
>- @protocol RTCDataChannelDelegate <NSObject>
>- 
>- /** The data channel state changed. */
>-@@ -63,6 +65,7 @@ typedef NS_ENUM(NSInteger, RTCDataChannelState) {
>- };
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDataChannel")))
>- @interface RTCDataChannel : NSObject
>- 
>- /**
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
>-index 65f59318447..0b8913ce549 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
>-@@ -16,6 +16,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDataChannelConfiguration")))
>- @interface RTCDataChannelConfiguration : NSObject
>- 
>- /** Set to YES if ordered delivery is required. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
>-index 3dddead60ef..2166277bc17 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
>-@@ -26,6 +26,7 @@ typedef NS_ENUM(NSInteger, RTCDispatcherQueueType) {
>-  *  shared dispatch queue.
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDispatcher")))
>- @interface RTCDispatcher : NSObject
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDtmfSender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDtmfSender.h
>-index c69af026b7c..b4b6c6cf7d6 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDtmfSender.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDtmfSender.h
>-@@ -15,6 +15,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDtmfSender")))
>- @protocol RTCDtmfSender <NSObject>
>- 
>- /**
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
>-index 60c624e0d25..d4d3b54fee7 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
>-@@ -19,6 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- @class RTCEAGLVideoView;
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCEAGLVideoViewDelegate")))
>- @protocol RTCEAGLVideoViewDelegate
>- 
>- - (void)videoView:(RTCEAGLVideoView *)videoView didChangeVideoSize:(CGSize)size;
>-@@ -30,6 +31,7 @@ RTC_EXPORT
>-  * bounds using OpenGLES 2.0 or OpenGLES 3.0.
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCEAGLVideoView")))
>- @interface RTCEAGLVideoView : UIView <RTCVideoRenderer>
>- 
>- @property(nonatomic, weak) id<RTCEAGLVideoViewDelegate> delegate;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
>-index 1e8a591b73f..33df4efcf0f 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
>-@@ -34,6 +34,7 @@ NS_ASSUME_NONNULL_BEGIN
>- // For kRTCFileLoggerTypeApp, the oldest log is overwritten.
>- // This class is not threadsafe.
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCFileLogger")))
>- @interface RTCFileLogger : NSObject
>- 
>- // The severity level to capture. The default is kRTCFileLoggerSeverityInfo.
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileVideoCapturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileVideoCapturer.h
>-index 7b898e494cf..70f482e35c7 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileVideoCapturer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileVideoCapturer.h
>-@@ -26,6 +26,7 @@ typedef void (^RTCFileVideoCapturerErrorBlock)(NSError *error);
>- RTC_EXPORT
>- 
>- NS_CLASS_AVAILABLE_IOS(10)
>-+__attribute__((objc_runtime_name("WK_RTCFileVideoCapturer")))
>- @interface RTCFileVideoCapturer : RTCVideoCapturer
>- 
>- /**
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
>-index 3b7f3e26e5a..902925203ea 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
>-@@ -15,6 +15,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCIceCandidate")))
>- @interface RTCIceCandidate : NSObject
>- 
>- /**
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h
>-index 727da8acbeb..d1feda40fe7 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h
>-@@ -20,6 +20,7 @@ typedef NS_ENUM(NSUInteger, RTCTlsCertPolicy) {
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCIceServer")))
>- @interface RTCIceServer : NSObject
>- 
>- /** URI(s) for this server represented as NSStrings. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
>-index 9384b153683..11eb4dc6233 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
>-@@ -12,6 +12,7 @@
>- 
>- NS_ASSUME_NONNULL_BEGIN
>- 
>-+__attribute__((objc_runtime_name("WK_RTCIntervalRange")))
>- @interface RTCIntervalRange : NSObject
>- 
>- @property(nonatomic, readonly) NSInteger min;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h
>-index 14d996c1cec..bbc4c035f78 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h
>-@@ -16,6 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- /** This does not currently conform to the spec. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCLegacyStatsReport")))
>- @interface RTCLegacyStatsReport : NSObject
>- 
>- /** Time since 1970-01-01T00:00:00Z in milliseconds. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h
>-index 4990e2529ef..b18fc777402 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h
>-@@ -12,6 +12,7 @@
>- #import "WebRTC/RTCVideoRenderer.h"
>- 
>- NS_AVAILABLE_MAC(10.11)
>-+__attribute__((objc_runtime_name("WK_RTCMTLNSVideoView")))
>- @interface RTCMTLNSVideoView : NSView<RTCVideoRenderer>
>- + (BOOL)isMetalAvailable;
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h
>-index 226c62a6a17..b62d0889b06 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h
>-@@ -31,6 +31,7 @@ NS_ASSUME_NONNULL_BEGIN
>- NS_CLASS_AVAILABLE_IOS(9)
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCMTLVideoView")))
>- @interface RTCMTLVideoView : UIView <RTCVideoRenderer>
>- 
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
>-index 26a116888a4..e6321940c44 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
>-@@ -40,6 +40,7 @@ RTC_EXTERN NSString * const kRTCMediaConstraintsValueTrue;
>- RTC_EXTERN NSString * const kRTCMediaConstraintsValueFalse;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCMediaConstraints")))
>- @interface RTCMediaConstraints : NSObject
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h
>-index 6cfcb7d2ce8..1a4003b5283 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h
>-@@ -22,6 +22,7 @@ typedef NS_ENUM(NSInteger, RTCSourceState) {
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCMediaSource")))
>- @interface RTCMediaSource : NSObject
>- 
>- /** The current state of the RTCMediaSource. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h
>-index b97960d436c..9cefde13c5f 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h
>-@@ -19,6 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
>- @class RTCVideoTrack;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCMediaStream")))
>- @interface RTCMediaStream : NSObject
>- 
>- /** The audio tracks in this stream. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
>-index c42f0cfb1a0..a1161c2ac2c 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
>-@@ -26,6 +26,7 @@ RTC_EXTERN NSString * const kRTCMediaStreamTrackKindAudio;
>- RTC_EXTERN NSString * const kRTCMediaStreamTrackKindVideo;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCMediaStreamTrack")))
>- @interface RTCMediaStreamTrack : NSObject
>- 
>- /**
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h
>-index f01bea9b71b..28b71d1f9e4 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h
>-@@ -15,6 +15,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCMetricsSampleInfo")))
>- @interface RTCMetricsSampleInfo : NSObject
>- 
>- /**
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
>-index 9e6a9ce9483..e29ec9f3c9d 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
>-@@ -20,12 +20,14 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- @class RTCNSGLVideoView;
>-+__attribute__((objc_runtime_name("WK_RTCNSGLVideoViewDelegate")))
>- @protocol RTCNSGLVideoViewDelegate
>- 
>- - (void)videoView:(RTCNSGLVideoView *)videoView didChangeVideoSize:(CGSize)size;
>- 
>- @end
>- 
>-+__attribute__((objc_runtime_name("WK_RTCNSGLVideoView")))
>- @interface RTCNSGLVideoView : NSOpenGLView <RTCVideoRenderer>
>- 
>- @property(nonatomic, weak) id<RTCNSGLVideoViewDelegate> delegate;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
>-index a4c113b1d65..041232e341b 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
>-@@ -69,6 +69,7 @@ typedef NS_ENUM(NSInteger, RTCStatsOutputLevel) {
>- @class RTCPeerConnection;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCPeerConnectionDelegate")))
>- @protocol RTCPeerConnectionDelegate <NSObject>
>- 
>- /** Called when the SignalingState changed. */
>-@@ -109,6 +110,7 @@ RTC_EXPORT
>- @end
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCPeerConnection")))
>- @interface RTCPeerConnection : NSObject
>- 
>- /** The object that will be notifed about events such as state changes and
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
>-index 56c399e27a8..8acaf43ce26 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
>-@@ -24,11 +24,15 @@ NS_ASSUME_NONNULL_BEGIN
>- @class RTCVideoSource;
>- @class RTCVideoTrack;
>- @class RTCPeerConnectionFactoryOptions;
>-+__attribute__((objc_runtime_name("WK_RTCPeerConnectionDelegate")))
>- @protocol RTCPeerConnectionDelegate;
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderFactory")))
>- @protocol RTCVideoDecoderFactory;
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderFactory")))
>- @protocol RTCVideoEncoderFactory;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCPeerConnectionFactory")))
>- @interface RTCPeerConnectionFactory : NSObject
>- 
>- /* Initialize object with default H264 video encoder/decoder factories */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h
>-index a65abc6c8b5..9a91ae218ae 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h
>-@@ -15,6 +15,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCPeerConnectionFactoryOptions")))
>- @interface RTCPeerConnectionFactoryOptions : NSObject
>- 
>- @property(nonatomic, assign) BOOL disableEncryption;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
>-index c7ca2f59c00..2725e53d31e 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
>-@@ -33,6 +33,7 @@ RTC_EXTERN const NSString * const kRTCH264CodecName;
>- 
>- /** Defined in http://w3c.github.io/webrtc-pc/#idl-def-RTCRtpCodecParameters */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpCodecParameters")))
>- @interface RTCRtpCodecParameters : NSObject
>- 
>- /** The RTP payload type. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
>-index a956f0d4c37..4d2e1e67152 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
>-@@ -15,6 +15,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_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/Framework/Headers/WebRTC/RTCRtpParameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
>-index bdebf84884c..1972749d9a1 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
>-@@ -17,6 +17,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpParameters")))
>- @interface RTCRtpParameters : NSObject
>- 
>- /** The currently active encodings in the order of preference. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
>-index 6156db9f932..324271876b7 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
>-@@ -26,6 +26,7 @@ typedef NS_ENUM(NSInteger, RTCRtpMediaType) {
>- @class RTCRtpReceiver;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpReceiverDelegate")))
>- @protocol RTCRtpReceiverDelegate <NSObject>
>- 
>- /** Called when the first RTP packet is received.
>-@@ -46,6 +47,7 @@ RTC_EXPORT
>- @end
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpReceiver")))
>- @protocol RTCRtpReceiver <NSObject>
>- 
>- /** A unique identifier for this receiver. */
>-@@ -73,6 +75,7 @@ RTC_EXPORT
>- @end
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpReceiver")))
>- @interface RTCRtpReceiver : NSObject <RTCRtpReceiver>
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
>-index f6e91250267..bf2fbc6ac51 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
>-@@ -18,6 +18,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpSender")))
>- @protocol RTCRtpSender <NSObject>
>- 
>- /** A unique identifier for this sender. */
>-@@ -41,6 +42,7 @@ RTC_EXPORT
>- @end
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpSender")))
>- @interface RTCRtpSender : NSObject <RTCRtpSender>
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
>-index 41439804a8a..ba05e22066f 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
>-@@ -25,6 +25,7 @@ typedef NS_ENUM(NSInteger, RTCSdpType) {
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCSessionDescription")))
>- @interface RTCSessionDescription : NSObject
>- 
>- /** The type of session description. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h
>-index 3addec83b1d..72215872bb9 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h
>-@@ -15,11 +15,13 @@ NS_ASSUME_NONNULL_BEGIN
>- @class RTCVideoCapturer;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoCapturerDelegate")))
>- @protocol RTCVideoCapturerDelegate <NSObject>
>- - (void)capturer:(RTCVideoCapturer *)capturer didCaptureVideoFrame:(RTCVideoFrame *)frame;
>- @end
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoCapturer")))
>- @interface RTCVideoCapturer : NSObject
>- 
>- @property(nonatomic, readonly, weak) id<RTCVideoCapturerDelegate> delegate;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
>-index 470eb023260..345c22e53c2 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
>-@@ -37,6 +37,7 @@ typedef NS_ENUM(NSUInteger, RTCVideoContentType) {
>- 
>- /** Represents an encoded frame. Corresponds to webrtc::EncodedImage. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCEncodedImage")))
>- @interface RTCEncodedImage : NSObject
>- 
>- @property(nonatomic, strong) NSData *buffer;
>-@@ -58,6 +59,7 @@ RTC_EXPORT
>- 
>- /** Information for header. Corresponds to webrtc::RTPFragmentationHeader. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpFragmentationHeader")))
>- @interface RTCRtpFragmentationHeader : NSObject
>- 
>- @property(nonatomic, strong) NSArray<NSNumber *> *fragmentationOffset;
>-@@ -71,6 +73,7 @@ RTC_EXPORT
>-  *  Corresponds to webrtc::CodecSpecificInfo.
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCCodecSpecificInfo")))
>- @protocol RTCCodecSpecificInfo <NSObject>
>- 
>- @end
>-@@ -90,6 +93,7 @@ typedef NS_ENUM(NSUInteger, RTCVideoCodecMode) {
>- 
>- /** Holds information to identify a codec. Corresponds to cricket::VideoCodec. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoCodecInfo")))
>- @interface RTCVideoCodecInfo : NSObject <NSCoding>
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-@@ -109,6 +113,7 @@ RTC_EXPORT
>- 
>- /** Settings for encoder. Corresponds to webrtc::VideoCodec. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderSettings")))
>- @interface RTCVideoEncoderSettings : NSObject
>- 
>- @property(nonatomic, strong) NSString *name;
>-@@ -130,6 +135,7 @@ RTC_EXPORT
>- 
>- /** QP thresholds for encoder. Corresponds to webrtc::VideoEncoder::QpThresholds. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderQpThresholds")))
>- @interface RTCVideoEncoderQpThresholds : NSObject
>- 
>- - (instancetype)initWithThresholdsLow:(NSInteger)low high:(NSInteger)high;
>-@@ -141,6 +147,7 @@ RTC_EXPORT
>- 
>- /** Protocol for encoder implementations. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoder")))
>- @protocol RTCVideoEncoder <NSObject>
>- 
>- - (void)setCallback:(RTCVideoEncoderCallback)callback;
>-@@ -162,6 +169,7 @@ RTC_EXPORT
>- 
>- /** Protocol for decoder implementations. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoder")))
>- @protocol RTCVideoDecoder <NSObject>
>- 
>- - (void)setCallback:(RTCVideoDecoderCallback)callback;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h
>-index 0ef2c8a655c..9fb1cb71d99 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h
>-@@ -17,6 +17,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- /** RTCVideoEncoderFactory is an Objective-C version of webrtc::VideoEncoderFactory. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderFactory")))
>- @protocol RTCVideoEncoderFactory <NSObject>
>- 
>- - (nullable id<RTCVideoEncoder>)createEncoder:(RTCVideoCodecInfo *)info;
>-@@ -26,6 +27,7 @@ RTC_EXPORT
>- 
>- /** RTCVideoDecoderFactory is an Objective-C version of webrtc::VideoDecoderFactory. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderFactory")))
>- @protocol RTCVideoDecoderFactory <NSObject>
>- 
>- - (nullable id<RTCVideoDecoder>)createDecoder:(RTCVideoCodecInfo *)info;
>-@@ -39,6 +41,7 @@ RTC_EXPORT
>-  *  codecs, create custom implementations of RTCVideoEncoderFactory and RTCVideoDecoderFactory.
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDefaultVideoEncoderFactory")))
>- @interface RTCDefaultVideoEncoderFactory : NSObject <RTCVideoEncoderFactory>
>- 
>- @property(nonatomic, retain) RTCVideoCodecInfo *preferredCodec;
>-@@ -48,6 +51,7 @@ RTC_EXPORT
>- @end
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDefaultVideoDecoderFactory")))
>- @interface RTCDefaultVideoDecoderFactory : NSObject <RTCVideoDecoderFactory>
>- @end
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
>-index 1c307ebe84a..c8d7fc007f1 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
>-@@ -20,6 +20,7 @@ typedef NS_ENUM(NSUInteger, RTCH264PacketizationMode) {
>- };
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCCodecSpecificInfoH264")))
>- @interface RTCCodecSpecificInfoH264 : NSObject <RTCCodecSpecificInfo>
>- 
>- @property(nonatomic, assign) RTCH264PacketizationMode packetizationMode;
>-@@ -28,6 +29,7 @@ RTC_EXPORT
>- 
>- /** Encoder. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderH264")))
>- @interface RTCVideoEncoderH264 : NSObject<RTCVideoEncoder>
>- 
>- - (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo;
>-@@ -36,15 +38,18 @@ RTC_EXPORT
>- 
>- /** Decoder. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderH264")))
>- @interface RTCVideoDecoderH264 : NSObject<RTCVideoDecoder>
>- @end
>- 
>- /** Encoder factory. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderFactoryH264")))
>- @interface RTCVideoEncoderFactoryH264 : NSObject<RTCVideoEncoderFactory>
>- @end
>- 
>- /** Decoder factory. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderFactoryH264")))
>- @interface RTCVideoDecoderFactoryH264 : NSObject<RTCVideoDecoderFactory>
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h
>-index 962b6312f7f..3e19ef8909c 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h
>-@@ -14,6 +14,7 @@
>- #import <WebRTC/RTCVideoCodec.h>
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderVP8")))
>- @interface RTCVideoDecoderVP8 : NSObject
>- 
>- /* This returns a VP8 decoder that can be returned from a RTCVideoDecoderFactory injected into
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h
>-index faf66ef065b..3fe79bc4a52 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h
>-@@ -14,6 +14,7 @@
>- #import <WebRTC/RTCVideoCodec.h>
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderVP9")))
>- @interface RTCVideoDecoderVP9 : NSObject
>- 
>- /* This returns a VP9 decoder that can be returned from a RTCVideoDecoderFactory injected into
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h
>-index e63b24dde75..52155b52551 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h
>-@@ -14,6 +14,7 @@
>- #import <WebRTC/RTCVideoCodec.h>
>- 
>- RTC_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/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h
>-index cef8c82e96f..866e626a05e 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h
>-@@ -14,6 +14,7 @@
>- #import <WebRTC/RTCVideoCodec.h>
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderVP9")))
>- @interface RTCVideoEncoderVP9 : NSObject
>- 
>- /* This returns a VP9 encoder that can be returned from a RTCVideoEncoderFactory injected into
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
>-index dcc4c6e7b5b..5b8fba602df 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
>-@@ -22,10 +22,12 @@ typedef NS_ENUM(NSInteger, RTCVideoRotation) {
>-   RTCVideoRotation_270 = 270,
>- };
>- 
>-+__attribute__((objc_runtime_name("WK_RTCVideoFrameBuffer")))
>- @protocol RTCVideoFrameBuffer;
>- 
>- // RTCVideoFrame is an ObjectiveC version of webrtc::VideoFrame.
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoFrame")))
>- @interface RTCVideoFrame : NSObject
>- 
>- /** Width without rotation applied. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h
>-index 4a683b0c926..8b825d8050c 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h
>-@@ -13,10 +13,12 @@
>- 
>- NS_ASSUME_NONNULL_BEGIN
>- 
>-+__attribute__((objc_runtime_name("WK_RTCI420Buffer")))
>- @protocol RTCI420Buffer;
>- 
>- // RTCVideoFrameBuffer is an ObjectiveC version of webrtc::VideoFrameBuffer.
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoFrameBuffer")))
>- @protocol RTCVideoFrameBuffer <NSObject>
>- 
>- @property(nonatomic, readonly) int width;
>-@@ -27,6 +29,7 @@ RTC_EXPORT
>- @end
>- 
>- /** Protocol for RTCVideoFrameBuffers containing YUV planar data. */
>-+__attribute__((objc_runtime_name("WK_RTCYUVPlanarBuffer")))
>- @protocol RTCYUVPlanarBuffer <RTCVideoFrameBuffer>
>- 
>- @property(nonatomic, readonly) int chromaWidth;
>-@@ -48,6 +51,7 @@ RTC_EXPORT
>- @end
>- 
>- /** Extension of the YUV planar data buffer with mutable data access */
>-+__attribute__((objc_runtime_name("WK_RTCMutableYUVPlanarBuffer")))
>- @protocol RTCMutableYUVPlanarBuffer <RTCYUVPlanarBuffer>
>- 
>- @property(nonatomic, readonly) uint8_t *mutableDataY;
>-@@ -57,15 +61,18 @@ RTC_EXPORT
>- @end
>- 
>- /** Protocol for RTCYUVPlanarBuffers containing I420 data */
>-+__attribute__((objc_runtime_name("WK_RTCI420Buffer")))
>- @protocol RTCI420Buffer <RTCYUVPlanarBuffer>
>- @end
>- 
>- /** Extension of the I420 buffer with mutable data access */
>-+__attribute__((objc_runtime_name("WK_RTCMutableI420Buffer")))
>- @protocol RTCMutableI420Buffer <RTCI420Buffer, RTCMutableYUVPlanarBuffer>
>- @end
>- 
>- /** RTCVideoFrameBuffer containing a CVPixelBufferRef */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCCVPixelBuffer")))
>- @interface RTCCVPixelBuffer : NSObject <RTCVideoFrameBuffer>
>- 
>- @property(nonatomic, readonly) CVPixelBufferRef pixelBuffer;
>-@@ -95,11 +102,13 @@ RTC_EXPORT
>- 
>- /** RTCI420Buffer implements the RTCI420Buffer protocol */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCI420Buffer")))
>- @interface RTCI420Buffer : NSObject <RTCI420Buffer>
>- @end
>- 
>- /** Mutable version of RTCI420Buffer */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCMutableI420Buffer")))
>- @interface RTCMutableI420Buffer : RTCI420Buffer <RTCMutableI420Buffer>
>- @end
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h
>-index 5e2e820918a..c821a1e3008 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h
>-@@ -20,6 +20,7 @@ NS_ASSUME_NONNULL_BEGIN
>- @class RTCVideoFrame;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoRenderer")))
>- @protocol RTCVideoRenderer <NSObject>
>- 
>- /** The size of the frame. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h
>-index a9ebc06c88e..60075e3435a 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h
>-@@ -18,6 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>- 
>-+__attribute__((objc_runtime_name("WK_RTCVideoSource")))
>- @interface RTCVideoSource : RTCMediaSource <RTCVideoCapturerDelegate>
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
>-index 899d7c34780..ecbdbb93c68 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
>-@@ -14,11 +14,13 @@
>- 
>- NS_ASSUME_NONNULL_BEGIN
>- 
>-+__attribute__((objc_runtime_name("WK_RTCVideoRenderer")))
>- @protocol RTCVideoRenderer;
>- @class RTCPeerConnectionFactory;
>- @class RTCVideoSource;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoTrack")))
>- @interface RTCVideoTrack : RTCMediaStreamTrack
>- 
>- /** The video source for this video track. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoViewShading.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoViewShading.h
>-index 6f3ece94a1f..5e33ff235ca 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoViewShading.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoViewShading.h
>-@@ -19,6 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
>-  * rendering for the RTCEAGLVideoView/RTCNSGLVideoView.
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoViewShading")))
>- @protocol RTCVideoViewShading <NSObject>
>- 
>- /** Callback for I420 frames. Each plane is given as a texture. */
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0013-Fix-SafeSetError-use-after-move.patch b/Source/ThirdParty/libwebrtc/WebKit/0013-Fix-SafeSetError-use-after-move.patch
>deleted file mode 100644
>index db7ce41afe6..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0013-Fix-SafeSetError-use-after-move.patch
>+++ /dev/null
>@@ -1,27 +0,0 @@
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc
>-index 6281d57e3db..1bebb0509a1 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc
>-@@ -226,18 +226,19 @@ uint32_t ConvertIceTransportTypeToCandidateFilter(
>- }
>- 
>- // Helper to set an error and return from a method.
>--bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
>-+static bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
>-   if (error) {
>-     error->set_type(type);
>-   }
>-   return type == webrtc::RTCErrorType::NONE;
>- }
>- 
>--bool SafeSetError(webrtc::RTCError error, webrtc::RTCError* error_out) {
>-+static bool SafeSetError(webrtc::RTCError&& error, webrtc::RTCError* error_out) {
>-+  bool result = error.ok();
>-   if (error_out) {
>-     *error_out = std::move(error);
>-   }
>--  return error.ok();
>-+  return result;
>- }
>- 
>- std::string GetSignalingStateString(
>diff --git a/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj b/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj
>index 2e4ca280b99..6bc19ce4012 100644
>--- a/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj
>+++ b/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj
>@@ -21,6 +21,51 @@
> /* End PBXAggregateTarget section */
> 
> /* Begin PBXBuildFile section */
>+		4102F68D21273206006AE8D7 /* rtcstatstraversal.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F68421273204006AE8D7 /* rtcstatstraversal.h */; };
>+		4102F68E21273206006AE8D7 /* jseptransportcontroller.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F68521273204006AE8D7 /* jseptransportcontroller.h */; };
>+		4102F68F21273206006AE8D7 /* transportstats.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F68621273205006AE8D7 /* transportstats.cc */; };
>+		4102F69021273206006AE8D7 /* jseptransportcontroller.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F68721273205006AE8D7 /* jseptransportcontroller.cc */; };
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>+		41F2637D21267B7700274F59 /* peerconnectioninterface.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41F2637421267B7500274F59 /* peerconnectioninterface.cc */; };
>+		41F2637E21267B7700274F59 /* audio_options.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41F2637521267B7600274F59 /* audio_options.cc */; };
>+		41F2637F21267B7700274F59 /* fec_controller.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F2637621267B7600274F59 /* fec_controller.h */; };
>+		41F2638021267B7700274F59 /* datachannelinterface.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41F2637721267B7600274F59 /* datachannelinterface.cc */; };
>+		41F2638121267B7700274F59 /* audio_options.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F2637821267B7600274F59 /* audio_options.h */; };
>+		41F2638221267B7700274F59 /* bitrate_constraints.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F2637921267B7700274F59 /* bitrate_constraints.h */; };
>+		41F2638321267B7700274F59 /* rtpreceiverinterface.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41F2637A21267B7700274F59 /* rtpreceiverinterface.cc */; };
>+		41F2638421267B7700274F59 /* jsepicecandidate.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41F2637B21267B7700274F59 /* jsepicecandidate.cc */; };
>+		41F2638521267B7700274F59 /* rtptransceiverinterface.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41F2637C21267B7700274F59 /* rtptransceiverinterface.cc */; };
>+		41F2638921267F4000274F59 /* fft.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F2638721267F4000274F59 /* fft.h */; };
>+		41F2638A21267F4000274F59 /* fft.c in Sources */ = {isa = PBXBuildFile; fileRef = 41F2638821267F4000274F59 /* fft.c */; };
>+		41F2638D21267F4900274F59 /* g711.c in Sources */ = {isa = PBXBuildFile; fileRef = 41F2638B21267F4900274F59 /* g711.c */; };
>+		41F2638E21267F4900274F59 /* g711.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F2638C21267F4900274F59 /* g711.h */; };
>+		41F2639221267F5300274F59 /* g722_encode.c in Sources */ = {isa = PBXBuildFile; fileRef = 41F2638F21267F5200274F59 /* g722_encode.c */; };
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>+		41F2639A21267F5E00274F59 /* pa_ringbuffer.c in Sources */ = {isa = PBXBuildFile; fileRef = 41F2639721267F5E00274F59 /* pa_ringbuffer.c */; };
>+		41F2639E2126800000274F59 /* aligned_malloc.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41F2639B21267FFF00274F59 /* aligned_malloc.cc */; };
>+		41F2639F2126800000274F59 /* aligned_malloc.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F2639C21267FFF00274F59 /* aligned_malloc.h */; };
>+		41F263A02126800000274F59 /* aligned_array.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F2639D2126800000274F59 /* aligned_array.h */; };
>+		41F263AC212680A800274F59 /* file_wrapper.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41F263A3212680A600274F59 /* file_wrapper.cc */; };
>+		41F263AD212680A800274F59 /* ignore_warnings.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F263A4212680A600274F59 /* ignore_warnings.h */; };
>+		41F263AE212680A800274F59 /* fallthrough.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F263A5212680A700274F59 /* fallthrough.h */; };
>+		41F263AF212680A800274F59 /* inline.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F263A6212680A700274F59 /* inline.h */; };
>+		41F263B0212680A800274F59 /* unused.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F263A7212680A700274F59 /* unused.h */; };
>+		41F263B1212680A800274F59 /* asm_defines.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F263A9212680A700274F59 /* asm_defines.h */; };
>+		41F263B2212680A800274F59 /* file_wrapper.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F263AA212680A800274F59 /* file_wrapper.h */; };
>+		41F263B3212680A800274F59 /* arch.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F263AB212680A800274F59 /* arch.h */; };
>+		41F263B72126810C00274F59 /* base64.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41F263B52126810B00274F59 /* base64.cc */; };
>+		41F263B82126810C00274F59 /* base64.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F263B62126810C00274F59 /* base64.h */; };
>+		41F263BB2126811500274F59 /* sigslot.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41F263B92126811400274F59 /* sigslot.cc */; };
>+		41F263BC2126811500274F59 /* sigslot.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F263BA2126811400274F59 /* sigslot.h */; };
> 		41F411AD1EF8D91F00343C26 /* null_aec_dump_factory.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41F411AC1EF8D91E00343C26 /* null_aec_dump_factory.cc */; };
> 		41F411B01EF8DA0100343C26 /* rtcerror.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41F411AE1EF8DA0000343C26 /* rtcerror.cc */; };
> 		41F411B11EF8DA0200343C26 /* rtcerror.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F411AF1EF8DA0000343C26 /* rtcerror.h */; };
>@@ -2819,6 +3176,59 @@
> /* Begin PBXFileReference section */
> 		37119A8020CCB6DD002C6DC9 /* Version.xcconfig */ = {isa = PBXFileReference; lastKnownFileType = text.xcconfig; path = Version.xcconfig; sourceTree = "<group>"; };
> 		37119A8120CCB6DD002C6DC9 /* WebKitTargetConditionals.xcconfig */ = {isa = PBXFileReference; lastKnownFileType = text.xcconfig; path = WebKitTargetConditionals.xcconfig; sourceTree = "<group>"; };
>+		4102F68421273204006AE8D7 /* rtcstatstraversal.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtcstatstraversal.h; sourceTree = "<group>"; };
>+		4102F68521273204006AE8D7 /* jseptransportcontroller.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = jseptransportcontroller.h; sourceTree = "<group>"; };
>+		4102F68621273205006AE8D7 /* transportstats.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = transportstats.cc; sourceTree = "<group>"; };
>+		4102F68721273205006AE8D7 /* jseptransportcontroller.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = jseptransportcontroller.cc; sourceTree = "<group>"; };
>+		4102F68821273205006AE8D7 /* rtpparametersconversion.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtpparametersconversion.h; sourceTree = "<group>"; };
>+		4102F68921273205006AE8D7 /* rtcstatstraversal.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rtcstatstraversal.cc; sourceTree = "<group>"; };
>+		4102F68A21273206006AE8D7 /* rtpparametersconversion.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rtpparametersconversion.cc; sourceTree = "<group>"; };
>+		4102F68B21273206006AE8D7 /* transportstats.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = transportstats.h; sourceTree = "<group>"; };
>+		4102F68C21273206006AE8D7 /* peerconnectioninternal.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = peerconnectioninternal.h; sourceTree = "<group>"; };
>+		4102F69621273260006AE8D7 /* transport_feedback_packet_loss_tracker.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = transport_feedback_packet_loss_tracker.h; sourceTree = "<group>"; };
>+		4102F69721273260006AE8D7 /* remix_resample.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = remix_resample.cc; sourceTree = "<group>"; };
>+		4102F69821273260006AE8D7 /* audio_level.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = audio_level.h; sourceTree = "<group>"; };
>+		4102F69921273261006AE8D7 /* channel_proxy.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = channel_proxy.h; sourceTree = "<group>"; };
>+		4102F69A21273261006AE8D7 /* audio_level.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = audio_level.cc; sourceTree = "<group>"; };
>+		4102F69B21273261006AE8D7 /* channel.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = channel.h; sourceTree = "<group>"; };
>+		4102F69C21273261006AE8D7 /* channel_proxy.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = channel_proxy.cc; sourceTree = "<group>"; };
>+		4102F69D21273261006AE8D7 /* remix_resample.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = remix_resample.h; sourceTree = "<group>"; };
>+		4102F69E21273262006AE8D7 /* channel.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = channel.cc; sourceTree = "<group>"; };
>+		4102F69F21273262006AE8D7 /* transport_feedback_packet_loss_tracker.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = transport_feedback_packet_loss_tracker.cc; sourceTree = "<group>"; };
>+		4102F6AA212732B0006AE8D7 /* rtp_payload_params.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = rtp_payload_params.cc; path = call/rtp_payload_params.cc; sourceTree = "<group>"; };
>+		4102F6AC212732E6006AE8D7 /* expand_uma_logger.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = expand_uma_logger.cc; sourceTree = "<group>"; };
>+		4102F6AD212732E7006AE8D7 /* expand_uma_logger.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = expand_uma_logger.h; sourceTree = "<group>"; };
>+		4102F6B021273330006AE8D7 /* mediaconfig.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = mediaconfig.h; path = base/mediaconfig.h; sourceTree = "<group>"; };
>+		4102F6B121273330006AE8D7 /* mediachannel.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = mediachannel.cc; path = base/mediachannel.cc; sourceTree = "<group>"; };
>+		4102F6B521273381006AE8D7 /* packet_receiver.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = packet_receiver.h; path = call/packet_receiver.h; sourceTree = "<group>"; };
>+		4102F6B621273381006AE8D7 /* audio_receive_stream.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = audio_receive_stream.cc; path = call/audio_receive_stream.cc; sourceTree = "<group>"; };
>+		4102F6B721273381006AE8D7 /* degraded_call.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = degraded_call.cc; path = call/degraded_call.cc; sourceTree = "<group>"; };
>+		4102F6B821273381006AE8D7 /* flexfec_receive_stream.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = flexfec_receive_stream.cc; path = call/flexfec_receive_stream.cc; sourceTree = "<group>"; };
>+		4102F6BE212733B5006AE8D7 /* video_send_stream_impl.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = video_send_stream_impl.h; sourceTree = "<group>"; };
>+		4102F6BF212733B6006AE8D7 /* video_quality_observer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = video_quality_observer.cc; sourceTree = "<group>"; };
>+		4102F6C0212733B6006AE8D7 /* video_stream_encoder.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = video_stream_encoder.h; sourceTree = "<group>"; };
>+		4102F6C1212733B6006AE8D7 /* video_stream_decoder_impl.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = video_stream_decoder_impl.cc; sourceTree = "<group>"; };
>+		4102F6C2212733B6006AE8D7 /* video_send_stream_impl.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = video_send_stream_impl.cc; sourceTree = "<group>"; };
>+		4102F6C4212733B6006AE8D7 /* video_stream_decoder_impl.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = video_stream_decoder_impl.h; sourceTree = "<group>"; };
>+		4102F6C5212733B6006AE8D7 /* video_quality_observer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = video_quality_observer.h; sourceTree = "<group>"; };
>+		4102F6D121273414006AE8D7 /* video_stream_decoder_create.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = video_stream_decoder_create.cc; sourceTree = "<group>"; };
>+		4102F6D221273414006AE8D7 /* color_space.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = color_space.cc; sourceTree = "<group>"; };
>+		4102F6D321273415006AE8D7 /* video_sink_interface.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = video_sink_interface.h; sourceTree = "<group>"; };
>+		4102F6D421273415006AE8D7 /* video_stream_decoder_create.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = video_stream_decoder_create.h; sourceTree = "<group>"; };
>+		4102F6D521273415006AE8D7 /* video_stream_encoder_create.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = video_stream_encoder_create.cc; sourceTree = "<group>"; };
>+		4102F6D621273415006AE8D7 /* i010_buffer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = i010_buffer.cc; sourceTree = "<group>"; };
>+		4102F6D721273415006AE8D7 /* video_stream_encoder_settings.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = video_stream_encoder_settings.h; sourceTree = "<group>"; };
>+		4102F6D821273416006AE8D7 /* color_space.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = color_space.h; sourceTree = "<group>"; };
>+		4102F6D921273416006AE8D7 /* video_bitrate_allocation.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = video_bitrate_allocation.cc; sourceTree = "<group>"; };
>+		4102F6DA21273416006AE8D7 /* video_bitrate_allocator_factory.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = video_bitrate_allocator_factory.h; sourceTree = "<group>"; };
>+		4102F6E52127345E006AE8D7 /* rtp_generic_frame_descriptor.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = rtp_generic_frame_descriptor.cc; sourceTree = "<group>"; };
>+		4102F6E62127345F006AE8D7 /* rtp_generic_frame_descriptor.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = rtp_generic_frame_descriptor.h; sourceTree = "<group>"; };
>+		4102F6E72127345F006AE8D7 /* rtp_video_header.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = rtp_video_header.cc; sourceTree = "<group>"; };
>+		4102F6E82127345F006AE8D7 /* rtp_generic_frame_descriptor_extension.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = rtp_generic_frame_descriptor_extension.h; sourceTree = "<group>"; };
>+		4102F6E92127345F006AE8D7 /* rtp_packet_to_send.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = rtp_packet_to_send.cc; sourceTree = "<group>"; };
>+		4102F6EA2127345F006AE8D7 /* rtp_video_header.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = rtp_video_header.h; sourceTree = "<group>"; };
>+		4102F6EB21273460006AE8D7 /* contributing_sources.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = contributing_sources.cc; sourceTree = "<group>"; };
>+		41053B4D212CD8EC004DAC6C /* UIDevice+H264Profile.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; name = "UIDevice+H264Profile.mm"; path = "Video/UIDevice+H264Profile.mm"; sourceTree = "<group>"; };
> 		41109AA61E5FA19200C0955A /* i420_buffer_pool.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = i420_buffer_pool.h; path = include/i420_buffer_pool.h; sourceTree = "<group>"; };
> 		41109AA71E5FA19200C0955A /* video_frame_buffer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = video_frame_buffer.h; path = include/video_frame_buffer.h; sourceTree = "<group>"; };
> 		41109AA81E5FA19200C0955A /* frame_callback.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = frame_callback.h; path = include/frame_callback.h; sourceTree = "<group>"; };
>@@ -2829,6 +3239,16 @@
> 		4124554A1EF8874300F11809 /* video_frame_buffer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = video_frame_buffer.cc; sourceTree = "<group>"; };
> 		412455541EF887FA00F11809 /* packetlossestimator.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = packetlossestimator.cc; sourceTree = "<group>"; };
> 		412455551EF887FA00F11809 /* packetlossestimator.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = packetlossestimator.h; sourceTree = "<group>"; };
>+		41299B8E2127367A00B3414B /* isac_vad.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = isac_vad.c; path = source/isac_vad.c; sourceTree = "<group>"; };
>+		41299B8F2127367B00B3414B /* isac_vad.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = isac_vad.h; path = source/isac_vad.h; sourceTree = "<group>"; };
>+		41299B902127367B00B3414B /* pitch_filter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = pitch_filter.h; path = source/pitch_filter.h; sourceTree = "<group>"; };
>+		41299B942127369000B3414B /* fake_network_pipe.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = fake_network_pipe.cc; path = call/fake_network_pipe.cc; sourceTree = "<group>"; };
>+		41299B952127369100B3414B /* fake_network_pipe.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = fake_network_pipe.h; path = call/fake_network_pipe.h; sourceTree = "<group>"; };
>+		41299B99212736DE00B3414B /* pitch_search_internal.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = pitch_search_internal.h; sourceTree = "<group>"; };
>+		41299B9A212736DE00B3414B /* pitch_search_internal.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = pitch_search_internal.cc; sourceTree = "<group>"; };
>+		41299B9B212736DE00B3414B /* pitch_info.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = pitch_info.h; sourceTree = "<group>"; };
>+		41299B9C212736DE00B3414B /* rnn.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = rnn.cc; sourceTree = "<group>"; };
>+		41299B9D212736DF00B3414B /* spectral_features_internal.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = spectral_features_internal.h; sourceTree = "<group>"; };
> 		413091F11EF8CF9100757C55 /* rtp_video_stream_receiver.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtp_video_stream_receiver.h; sourceTree = "<group>"; };
> 		413091F21EF8CF9100757C55 /* rtp_video_stream_receiver.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rtp_video_stream_receiver.cc; sourceTree = "<group>"; };
> 		413091F51EF8CFBD00757C55 /* iceserverparsing.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = iceserverparsing.h; sourceTree = "<group>"; };
>@@ -2969,7 +3389,6 @@
> 		413A21C11FE18D9D00373E99 /* copyonwritebuffer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = copyonwritebuffer.cc; path = rtc_base/copyonwritebuffer.cc; sourceTree = "<group>"; };
> 		413A21C21FE18D9E00373E99 /* asyncudpsocket.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = asyncudpsocket.cc; path = rtc_base/asyncudpsocket.cc; sourceTree = "<group>"; };
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>+		41F2639121267F5300274F59 /* g722_decode.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = g722_decode.c; path = g722/g722_decode.c; sourceTree = "<group>"; };
>+		41F2639521267F5E00274F59 /* pa_memorybarrier.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = pa_memorybarrier.h; path = portaudio/pa_memorybarrier.h; sourceTree = "<group>"; };
>+		41F2639621267F5E00274F59 /* pa_ringbuffer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = pa_ringbuffer.h; path = portaudio/pa_ringbuffer.h; sourceTree = "<group>"; };
>+		41F2639721267F5E00274F59 /* pa_ringbuffer.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = pa_ringbuffer.c; path = portaudio/pa_ringbuffer.c; sourceTree = "<group>"; };
>+		41F2639B21267FFF00274F59 /* aligned_malloc.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = aligned_malloc.cc; path = rtc_base/memory/aligned_malloc.cc; sourceTree = "<group>"; };
>+		41F2639C21267FFF00274F59 /* aligned_malloc.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = aligned_malloc.h; path = rtc_base/memory/aligned_malloc.h; sourceTree = "<group>"; };
>+		41F2639D2126800000274F59 /* aligned_array.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = aligned_array.h; path = rtc_base/memory/aligned_array.h; sourceTree = "<group>"; };
>+		41F263A3212680A600274F59 /* file_wrapper.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = file_wrapper.cc; sourceTree = "<group>"; };
>+		41F263A4212680A600274F59 /* ignore_warnings.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ignore_warnings.h; sourceTree = "<group>"; };
>+		41F263A5212680A700274F59 /* fallthrough.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = fallthrough.h; sourceTree = "<group>"; };
>+		41F263A6212680A700274F59 /* inline.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = inline.h; sourceTree = "<group>"; };
>+		41F263A7212680A700274F59 /* unused.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = unused.h; sourceTree = "<group>"; };
>+		41F263A8212680A700274F59 /* BUILD.gn */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = BUILD.gn; sourceTree = "<group>"; };
>+		41F263A9212680A700274F59 /* asm_defines.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = asm_defines.h; sourceTree = "<group>"; };
>+		41F263AA212680A800274F59 /* file_wrapper.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = file_wrapper.h; sourceTree = "<group>"; };
>+		41F263AB212680A800274F59 /* arch.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = arch.h; sourceTree = "<group>"; };
>+		41F263B52126810B00274F59 /* base64.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = base64.cc; path = base64/base64.cc; sourceTree = "<group>"; };
>+		41F263B62126810C00274F59 /* base64.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = base64.h; path = base64/base64.h; sourceTree = "<group>"; };
>+		41F263B92126811400274F59 /* sigslot.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sigslot.cc; path = sigslot/sigslot.cc; sourceTree = "<group>"; };
>+		41F263BA2126811400274F59 /* sigslot.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = sigslot.h; path = sigslot/sigslot.h; sourceTree = "<group>"; };
>+		41F263BD2126818800274F59 /* video_stream_encoder_observer.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = video_stream_encoder_observer.h; sourceTree = "<group>"; };
>+		41F263BE2126818800274F59 /* video_source_interface.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = video_source_interface.h; sourceTree = "<group>"; };
>+		41F263BF2126818900274F59 /* i010_buffer.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = i010_buffer.h; sourceTree = "<group>"; };
>+		41F263C02126818900274F59 /* video_stream_encoder_create.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = video_stream_encoder_create.h; sourceTree = "<group>"; };
>+		41F263C12126818900274F59 /* video_bitrate_allocator.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = video_bitrate_allocator.h; sourceTree = "<group>"; };
>+		41F263C22126818900274F59 /* encoded_frame.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = encoded_frame.h; sourceTree = "<group>"; };
>+		41F263C32126818900274F59 /* video_stream_encoder_interface.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = video_stream_encoder_interface.h; sourceTree = "<group>"; };
>+		41F263C42126818900274F59 /* video_bitrate_allocation.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = video_bitrate_allocation.h; sourceTree = "<group>"; };
>+		41F263C52126818A00274F59 /* video_stream_decoder.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = video_stream_decoder.h; sourceTree = "<group>"; };
>+		41F263C62126818A00274F59 /* video_source_interface.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = video_source_interface.cc; sourceTree = "<group>"; };
>+		41F263C72126818A00274F59 /* encoded_frame.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = encoded_frame.cc; sourceTree = "<group>"; };
> 		41F411AC1EF8D91E00343C26 /* null_aec_dump_factory.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = null_aec_dump_factory.cc; path = Source/webrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc; sourceTree = SOURCE_ROOT; };
> 		41F411AE1EF8DA0000343C26 /* rtcerror.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rtcerror.cc; sourceTree = "<group>"; };
> 		41F411AF1EF8DA0000343C26 /* rtcerror.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtcerror.h; sourceTree = "<group>"; };
>@@ -5686,6 +6426,20 @@
> 		412455581EF88AD900F11809 /* video_codecs */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41E02CC22127358700C27CD6 /* builtin_video_encoder_factory.cc */,
>+				41E02CC32127358800C27CD6 /* video_encoder_config.cc */,
>+				41DDB27321267AC000296D47 /* builtin_video_decoder_factory.cc */,
>+				41DDB27621267AC100296D47 /* builtin_video_decoder_factory.h */,
>+				41DDB27721267AC100296D47 /* builtin_video_encoder_factory.h */,
>+				41DDB27A21267AC200296D47 /* sdp_video_format.cc */,
>+				41DDB27021267AC000296D47 /* video_codec.cc */,
>+				41DDB27121267AC000296D47 /* video_codec.h */,
>+				41DDB27421267AC100296D47 /* video_decoder_software_fallback_wrapper.cc */,
>+				41DDB27821267AC100296D47 /* video_decoder_software_fallback_wrapper.h */,
>+				41DDB27221267AC000296D47 /* video_decoder.cc */,
>+				41DDB27B21267AC200296D47 /* video_encoder_config.h */,
>+				41DDB27521267AC100296D47 /* video_encoder_software_fallback_wrapper.cc */,
>+				41DDB27921267AC200296D47 /* video_encoder_software_fallback_wrapper.h */,
> 				4145F6191FE1F38B00EB9CAF /* sdp_video_format.h */,
> 				4145E48B1EF88B9600FCF6E6 /* video_decoder.h */,
> 				4145F61A1FE1F38C00EB9CAF /* video_decoder_factory.h */,
>@@ -5696,6 +6450,34 @@
> 			name = video_codecs;
> 			sourceTree = "<group>";
> 		};
>+		41299B98212736B200B3414B /* rnn_vad */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419241362127372300634FCF /* common.h */,
>+				419241312127372200634FCF /* features_extraction.cc */,
>+				419241322127372200634FCF /* features_extraction.h */,
>+				419241342127372300634FCF /* fft_util.cc */,
>+				419241392127372400634FCF /* fft_util.h */,
>+				419241352127372300634FCF /* lp_residual.cc */,
>+				419241372127372400634FCF /* lp_residual.h */,
>+				41299B9B212736DE00B3414B /* pitch_info.h */,
>+				41299B9A212736DE00B3414B /* pitch_search_internal.cc */,
>+				41299B99212736DE00B3414B /* pitch_search_internal.h */,
>+				419241382127372400634FCF /* pitch_search.cc */,
>+				419241332127372300634FCF /* pitch_search.h */,
>+				419241A7212758D300634FCF /* ring_buffer.h */,
>+				41299B9C212736DE00B3414B /* rnn.cc */,
>+				419241A3212758D200634FCF /* rnn.h */,
>+				419241A5212758D200634FCF /* sequence_buffer.h */,
>+				419241A1212758D200634FCF /* spectral_features_internal.cc */,
>+				41299B9D212736DF00B3414B /* spectral_features_internal.h */,
>+				419241A6212758D200634FCF /* spectral_features.cc */,
>+				419241A4212758D200634FCF /* spectral_features.h */,
>+				419241A2212758D200634FCF /* symmetric_matrix_buffer.h */,
>+			);
>+			path = rnn_vad;
>+			sourceTree = "<group>";
>+		};
> 		4130922A1EF8D73500757C55 /* aec_dump */ = {
> 			isa = PBXGroup;
> 			children = (
>@@ -5722,10 +6504,11 @@
> 		413A21431FE18D0100373E99 /* rtc_base */ = {
> 			isa = PBXGroup;
> 			children = (
>-				413A21801FE18D7A00373E99 /* DEPS */,
>-				41F9BFC82051DD8E00ABF0B9 /* experiments */,
>-				419C83411FE246380040C30F /* numerics */,
>-				413A21FD1FE18DBB00373E99 /* OWNERS */,
>+				4192417A2127390C00634FCF /* time */,
>+				419241652127387900634FCF /* experiments */,
>+				41F2639D2126800000274F59 /* aligned_array.h */,
>+				41F2639B21267FFF00274F59 /* aligned_malloc.cc */,
>+				41F2639C21267FFF00274F59 /* aligned_malloc.h */,
> 				413A22921FE18E0000373E99 /* arraysize.h */,
> 				413A214D1FE18D5000373E99 /* asyncinvoker-inl.h */,
> 				413A22391FE18DD700373E99 /* asyncinvoker.cc */,
>@@ -5736,76 +6519,78 @@
> 				413A21501FE18D5100373E99 /* asyncresolverinterface.h */,
> 				413A22431FE18DDC00373E99 /* asyncsocket.cc */,
> 				413A220B1FE18DC100373E99 /* asyncsocket.h */,
>+				413A22741FE18DF600373E99 /* asynctcpsocket_unittest.cc */,
> 				413A22581FE18DE500373E99 /* asynctcpsocket.cc */,
> 				413A21641FE18D5F00373E99 /* asynctcpsocket.h */,
>-				413A22741FE18DF600373E99 /* asynctcpsocket_unittest.cc */,
>+				413A22911FE18E0000373E99 /* asyncudpsocket_unittest.cc */,
> 				413A21C21FE18D9E00373E99 /* asyncudpsocket.cc */,
> 				413A215C1FE18D5800373E99 /* asyncudpsocket.h */,
>-				413A22911FE18E0000373E99 /* asyncudpsocket_unittest.cc */,
>-				413A21981FE18D8A00373E99 /* atomicops.h */,
> 				413A225D1FE18DE600373E99 /* atomicops_unittest.cc */,
>+				413A21981FE18D8A00373E99 /* atomicops.h */,
> 				413A21811FE18D7B00373E99 /* base64_unittest.cc */,
>-				413A22091FE18DC000373E99 /* bind.h */,
> 				413A222F1FE18DD200373E99 /* bind_unittest.cc */,
>+				413A22091FE18DC000373E99 /* bind.h */,
>+				413A22571FE18DE400373E99 /* bitbuffer_unittest.cc */,
> 				413A21E71FE18DB100373E99 /* bitbuffer.cc */,
> 				413A22241FE18DCD00373E99 /* bitbuffer.h */,
>-				413A22571FE18DE400373E99 /* bitbuffer_unittest.cc */,
>+				413A221C1FE18DC800373E99 /* bitrateallocationstrategy_unittest.cc */,
> 				413A214A1FE18D4E00373E99 /* bitrateallocationstrategy.cc */,
> 				413A21D21FE18DA700373E99 /* bitrateallocationstrategy.h */,
>-				413A221C1FE18DC800373E99 /* bitrateallocationstrategy_unittest.cc */,
>-				413A223B1FE18DD900373E99 /* buffer.h */,
> 				413A22131FE18DC400373E99 /* buffer_unittest.cc */,
>+				413A223B1FE18DD900373E99 /* buffer.h */,
>+				413A21711FE18D6D00373E99 /* bufferqueue_unittest.cc */,
> 				413A22481FE18DDE00373E99 /* bufferqueue.cc */,
> 				413A21D71FE18DAA00373E99 /* bufferqueue.h */,
>-				413A21711FE18D6D00373E99 /* bufferqueue_unittest.cc */,
> 				413A228F1FE18DFF00373E99 /* BUILD.gn */,
>+				413A21C01FE18D9D00373E99 /* bytebuffer_unittest.cc */,
> 				413A22A01FE18E0600373E99 /* bytebuffer.cc */,
> 				413A22511FE18DE100373E99 /* bytebuffer.h */,
>-				413A21C01FE18D9D00373E99 /* bytebuffer_unittest.cc */,
>-				413A21EA1FE18DB300373E99 /* byteorder.h */,
> 				413A22361FE18DD500373E99 /* byteorder_unittest.cc */,
>+				413A21EA1FE18DB300373E99 /* byteorder.h */,
>+				413A218A1FE18D8200373E99 /* callback_unittest.cc */,
> 				413A21C31FE18D9E00373E99 /* callback.h */,
> 				413A21B71FE18D9900373E99 /* callback.h.pump */,
>-				413A218A1FE18D8200373E99 /* callback_unittest.cc */,
> 				413A22201FE18DCA00373E99 /* checks.cc */,
> 				413A21FB1FE18DBA00373E99 /* checks.h */,
> 				413A22321FE18DD300373E99 /* compile_assert_c.h */,
> 				413A22191FE18DC600373E99 /* constructormagic.h */,
>+				413A22961FE18E0200373E99 /* copyonwritebuffer_unittest.cc */,
> 				413A21C11FE18D9D00373E99 /* copyonwritebuffer.cc */,
> 				413A22591FE18DE500373E99 /* copyonwritebuffer.h */,
>-				413A22961FE18E0200373E99 /* copyonwritebuffer_unittest.cc */,
>+				413A22611FE18DE800373E99 /* cpu_time_unittest.cc */,
> 				413A21B81FE18D9900373E99 /* cpu_time.cc */,
> 				413A22901FE18E0000373E99 /* cpu_time.h */,
>-				413A22611FE18DE800373E99 /* cpu_time_unittest.cc */,
>+				413A22651FE18DEB00373E99 /* crc32_unittest.cc */,
> 				413A21EF1FE18DB500373E99 /* crc32.cc */,
> 				413A21511FE18D5200373E99 /* crc32.h */,
>-				413A22651FE18DEB00373E99 /* crc32_unittest.cc */,
>+				413A21D01FE18DA700373E99 /* criticalsection_unittest.cc */,
> 				413A22351FE18DD500373E99 /* criticalsection.cc */,
> 				413A22161FE18DC500373E99 /* criticalsection.h */,
>-				413A21D01FE18DA700373E99 /* criticalsection_unittest.cc */,
> 				413A21CE1FE18DA500373E99 /* cryptstring.cc */,
> 				413A21D51FE18DA900373E99 /* cryptstring.h */,
>+				41E02CBB2127356A00C27CD6 /* data_rate_limiter.cc */,
>+				41E02CBC2127356A00C27CD6 /* data_rate_limiter.h */,
> 				413A21FA1FE18DBA00373E99 /* deprecation.h */,
>+				413A21801FE18D7A00373E99 /* DEPS */,
> 				413A21F41FE18DB700373E99 /* dscp.h */,
>-				413A22641FE18DEB00373E99 /* event.cc */,
>-				413A22521FE18DE200373E99 /* event.h */,
> 				413A223C1FE18DD900373E99 /* event_tracer.cc */,
> 				413A222B1FE18DCF00373E99 /* event_tracer.h */,
>-				413A227D1FE18DF900373E99 /* event_tracer_unittest.cc */,
>-				413A21C41FE18D9F00373E99 /* event_unittest.cc */,
>+				413A22641FE18DEB00373E99 /* event.cc */,
>+				413A22521FE18DE200373E99 /* event.h */,
>+				41F9BFC82051DD8E00ABF0B9 /* experiments */,
> 				413A21BC1FE18D9B00373E99 /* fakeclock.cc */,
> 				413A222C1FE18DD000373E99 /* fakeclock.h */,
> 				413A215D1FE18D5900373E99 /* fakenetwork.h */,
> 				413A22791FE18DF800373E99 /* fakesslidentity.cc */,
> 				413A21A31FE18D9000373E99 /* fakesslidentity.h */,
>-				413A21911FE18D8500373E99 /* file.cc */,
>-				413A21821FE18D7C00373E99 /* file.h */,
> 				413A22851FE18DFC00373E99 /* file_posix.cc */,
> 				413A218B1FE18D8200373E99 /* file_unittest.cc */,
> 				413A21521FE18D5200373E99 /* file_win.cc */,
>+				413A21911FE18D8500373E99 /* file.cc */,
>+				413A21821FE18D7C00373E99 /* file.h */,
>+				413A225A1FE18DE500373E99 /* filerotatingstream_unittest.cc */,
> 				413A22871FE18DFD00373E99 /* filerotatingstream.cc */,
> 				413A218C1FE18D8300373E99 /* filerotatingstream.h */,
>-				413A225A1FE18DE500373E99 /* filerotatingstream_unittest.cc */,
> 				413A22621FE18DE900373E99 /* fileutils.cc */,
> 				413A226A1FE18DF000373E99 /* fileutils.h */,
> 				413A22171FE18DC600373E99 /* firewallsocketserver.cc */,
>@@ -5813,58 +6598,58 @@
> 				413A21611FE18D5B00373E99 /* flags.cc */,
> 				413A22691FE18DEF00373E99 /* flags.h */,
> 				413A21471FE18D4D00373E99 /* format_macros.h */,
>-				413A21451FE18D4C00373E99 /* function_view.h */,
> 				413A21891FE18D8100373E99 /* function_view_unittest.cc */,
>+				413A21451FE18D4C00373E99 /* function_view.h */,
> 				413A21AC1FE18D9400373E99 /* gtest_prod_util.h */,
>-				413A21951FE18D8800373E99 /* gunit.h */,
> 				413A220A1FE18DC000373E99 /* gunit_prod.h */,
>+				413A21951FE18D8800373E99 /* gunit.h */,
>+				413A21C51FE18D9F00373E99 /* helpers_unittest.cc */,
> 				413A221B1FE18DC700373E99 /* helpers.cc */,
> 				413A21831FE18D7D00373E99 /* helpers.h */,
>-				413A21C51FE18D9F00373E99 /* helpers_unittest.cc */,
>+				413A21E61FE18DB100373E99 /* httpbase_unittest.cc */,
> 				413A21B21FE18D9700373E99 /* httpbase.cc */,
> 				413A22601FE18DE700373E99 /* httpbase.h */,
>-				413A21E61FE18DB100373E99 /* httpbase_unittest.cc */,
>+				413A21B31FE18D9700373E99 /* httpcommon_unittest.cc */,
> 				413A225F1FE18DE700373E99 /* httpcommon-inl.h */,
> 				413A221A1FE18DC700373E99 /* httpcommon.cc */,
> 				413A22421FE18DDB00373E99 /* httpcommon.h */,
>-				413A21B31FE18D9700373E99 /* httpcommon_unittest.cc */,
>+				413A21541FE18D5300373E99 /* httpserver_unittest.cc */,
> 				413A21FE1FE18DBC00373E99 /* httpserver.cc */,
> 				413A21701FE18D6C00373E99 /* httpserver.h */,
>-				413A21541FE18D5300373E99 /* httpserver_unittest.cc */,
>-				413A22971FE18E0200373E99 /* ifaddrs-android.cc */,
>-				413A216F1FE18D6B00373E99 /* ifaddrs-android.h */,
> 				413A22211FE18DCB00373E99 /* ifaddrs_converter.cc */,
> 				413A227A1FE18DF800373E99 /* ifaddrs_converter.h */,
>+				413A22971FE18E0200373E99 /* ifaddrs-android.cc */,
>+				413A216F1FE18D6B00373E99 /* ifaddrs-android.h */,
> 				413A22931FE18E0100373E99 /* ignore_wundef.h */,
>+				413A216A1FE18D6700373E99 /* ipaddress_unittest.cc */,
> 				413A229B1FE18E0400373E99 /* ipaddress.cc */,
> 				413A214E1FE18D5000373E99 /* ipaddress.h */,
>-				413A216A1FE18D6700373E99 /* ipaddress_unittest.cc */,
>+				413A21F21FE18DB600373E99 /* json_unittest.cc */,
> 				413A21DC1FE18DAD00373E99 /* json.cc */,
> 				413A21EC1FE18DB400373E99 /* json.h */,
>-				413A21F21FE18DB600373E99 /* json_unittest.cc */,
> 				413A22121FE18DC400373E99 /* keep_ref_until_done.h */,
> 				413A22991FE18E0300373E99 /* location.cc */,
> 				413A229A1FE18E0400373E99 /* location.h */,
>-				413A21721FE18D6D00373E99 /* logging.cc */,
>-				413A21DE1FE18DAE00373E99 /* logging.h */,
> 				413A21671FE18D6300373E99 /* logging_mac.mm */,
> 				413A21E01FE18DAE00373E99 /* logging_unittest.cc */,
>+				413A21721FE18D6D00373E99 /* logging.cc */,
>+				413A21DE1FE18DAE00373E99 /* logging.h */,
> 				413A226B1FE18DF000373E99 /* logsinks.cc */,
> 				413A21FF1FE18DBC00373E99 /* logsinks.h */,
> 				413A21D81FE18DAB00373E99 /* macifaddrs_converter.cc */,
> 				413A21E41FE18DB000373E99 /* macutils.cc */,
> 				413A22441FE18DDC00373E99 /* macutils.h */,
>+				413A21F01FE18DB500373E99 /* memory_usage_unittest.cc */,
> 				413A222D1FE18DD100373E99 /* memory_usage.cc */,
> 				413A21B41FE18D9700373E99 /* memory_usage.h */,
>-				413A21F01FE18DB500373E99 /* memory_usage_unittest.cc */,
>+				413A21E11FE18DAF00373E99 /* messagedigest_unittest.cc */,
> 				413A21A41FE18D9000373E99 /* messagedigest.cc */,
> 				413A21A51FE18D9100373E99 /* messagedigest.h */,
>-				413A21E11FE18DAF00373E99 /* messagedigest_unittest.cc */,
> 				413A22181FE18DC600373E99 /* messagehandler.cc */,
> 				413A215F1FE18D5A00373E99 /* messagehandler.h */,
>+				413A217A1FE18D7500373E99 /* messagequeue_unittest.cc */,
> 				413A21561FE18D5400373E99 /* messagequeue.cc */,
> 				413A22561FE18DE400373E99 /* messagequeue.h */,
>-				413A217A1FE18D7500373E99 /* messagequeue_unittest.cc */,
> 				413A21C81FE18DA100373E99 /* nat_unittest.cc */,
> 				413A21531FE18D5300373E99 /* natserver.cc */,
> 				413A216B1FE18D6800373E99 /* natserver.h */,
>@@ -5876,45 +6661,54 @@
> 				413A214F1FE18D5100373E99 /* nethelper.h */,
> 				413A224E1FE18DE000373E99 /* nethelpers.cc */,
> 				413A22371FE18DD600373E99 /* nethelpers.h */,
>-				413A22941FE18E0100373E99 /* network.cc */,
>-				413A22041FE18DBE00373E99 /* network.h */,
> 				413A22231FE18DCC00373E99 /* network_constants.h */,
> 				413A22681FE18DEE00373E99 /* network_unittest.cc */,
>+				413A22941FE18E0100373E99 /* network.cc */,
>+				413A22041FE18DBE00373E99 /* network.h */,
> 				413A22381FE18DD600373E99 /* networkmonitor.cc */,
> 				413A22291FE18DCF00373E99 /* networkmonitor.h */,
> 				413A21691FE18D6500373E99 /* networkroute.h */,
> 				413A220C1FE18DC100373E99 /* noop.cc */,
> 				413A229C1FE18E0400373E99 /* noop.mm */,
>+				413A227C1FE18DF800373E99 /* nullsocketserver_unittest.cc */,
> 				413A227F1FE18DF900373E99 /* nullsocketserver.cc */,
> 				413A226D1FE18DF100373E99 /* nullsocketserver.h */,
>-				413A227C1FE18DF800373E99 /* nullsocketserver_unittest.cc */,
>-				413A21E21FE18DAF00373E99 /* onetimeevent.h */,
>+				419C83411FE246380040C30F /* numerics */,
> 				413A21621FE18D5D00373E99 /* onetimeevent_unittest.cc */,
>+				413A21E21FE18DAF00373E99 /* onetimeevent.h */,
> 				413A22751FE18DF600373E99 /* openssl.h */,
>+				413A227E1FE18DF900373E99 /* openssladapter_unittest.cc */,
> 				413A219A1FE18D8B00373E99 /* openssladapter.cc */,
> 				413A21DB1FE18DAC00373E99 /* openssladapter.h */,
>-				413A227E1FE18DF900373E99 /* openssladapter_unittest.cc */,
>+				41DDB25E212679A100296D47 /* opensslcertificate.cc */,
>+				41DDB260212679A200296D47 /* opensslcertificate.h */,
> 				413A22401FE18DDB00373E99 /* openssldigest.cc */,
> 				413A218E1FE18D8400373E99 /* openssldigest.h */,
> 				413A21681FE18D6400373E99 /* opensslidentity.cc */,
> 				413A22491FE18DDF00373E99 /* opensslidentity.h */,
>+				41DDB25C212679A000296D47 /* opensslsessioncache.cc */,
>+				41DDB25D212679A000296D47 /* opensslsessioncache.h */,
> 				413A218D1FE18D8300373E99 /* opensslstreamadapter.cc */,
> 				413A22631FE18DEA00373E99 /* opensslstreamadapter.h */,
>+				41DDB25F212679A200296D47 /* opensslutility.cc */,
>+				41DDB25B2126799F00296D47 /* opensslutility.h */,
>+				413A21C71FE18DA100373E99 /* optionsfile_unittest.cc */,
> 				413A21F61FE18DB800373E99 /* optionsfile.cc */,
> 				413A21F11FE18DB600373E99 /* optionsfile.h */,
>-				413A21C71FE18DA100373E99 /* optionsfile_unittest.cc */,
>+				413A21FD1FE18DBB00373E99 /* OWNERS */,
>+				413A219D1FE18D8D00373E99 /* pathutils_unittest.cc */,
> 				413A21E81FE18DB200373E99 /* pathutils.cc */,
> 				413A223D1FE18DD900373E99 /* pathutils.h */,
>-				413A219D1FE18D8D00373E99 /* pathutils_unittest.cc */,
>+				413A227B1FE18DF800373E99 /* physicalsocketserver_unittest.cc */,
> 				413A21E91FE18DB200373E99 /* physicalsocketserver.cc */,
> 				413A22981FE18E0300373E99 /* physicalsocketserver.h */,
>-				413A227B1FE18DF800373E99 /* physicalsocketserver_unittest.cc */,
> 				413A22071FE18DBF00373E99 /* platform_file.cc */,
> 				413A21791FE18D7400373E99 /* platform_file.h */,
>-				413A21CA1FE18DA300373E99 /* platform_thread.cc */,
>-				413A22531FE18DE200373E99 /* platform_thread.h */,
>+				41DDB253212678E300296D47 /* platform_thread_types.cc */,
> 				413A21C91FE18DA200373E99 /* platform_thread_types.h */,
> 				413A21D11FE18DA700373E99 /* platform_thread_unittest.cc */,
>+				413A21CA1FE18DA300373E99 /* platform_thread.cc */,
>+				413A22531FE18DE200373E99 /* platform_thread.h */,
> 				413A22731FE18DF600373E99 /* protobuf_utils.h */,
> 				413A214B1FE18D4E00373E99 /* proxy_unittest.cc */,
> 				413A226F1FE18DF200373E99 /* proxyinfo.cc */,
>@@ -5923,118 +6717,125 @@
> 				413A22721FE18DF500373E99 /* proxyserver.h */,
> 				413A21A61FE18D9100373E99 /* race_checker.cc */,
> 				413A229D1FE18E0500373E99 /* race_checker.h */,
>+				413A21661FE18D6200373E99 /* random_unittest.cc */,
> 				413A22781FE18DF700373E99 /* random.cc */,
> 				413A21F81FE18DB900373E99 /* random.h */,
>-				413A21661FE18D6200373E99 /* random_unittest.cc */,
>+				413A229E1FE18E0500373E99 /* rate_limiter_unittest.cc */,
> 				413A223A1FE18DD800373E99 /* rate_limiter.cc */,
> 				413A220D1FE18DC200373E99 /* rate_limiter.h */,
>-				413A229E1FE18E0500373E99 /* rate_limiter_unittest.cc */,
>+				413A21731FE18D6E00373E99 /* rate_statistics_unittest.cc */,
> 				413A22051FE18DBE00373E99 /* rate_statistics.cc */,
> 				413A21B91FE18D9A00373E99 /* rate_statistics.h */,
>-				413A21731FE18D6E00373E99 /* rate_statistics_unittest.cc */,
>+				413A221F1FE18DC900373E99 /* ratetracker_unittest.cc */,
> 				413A220F1FE18DC200373E99 /* ratetracker.cc */,
> 				413A21AF1FE18D9500373E99 /* ratetracker.h */,
>-				413A221F1FE18DC900373E99 /* ratetracker_unittest.cc */,
> 				413A21AD1FE18D9400373E99 /* refcount.h */,
>-				413A22141FE18DC400373E99 /* refcountedobject.h */,
> 				413A21CD1FE18DA500373E99 /* refcountedobject_unittest.cc */,
>+				413A22141FE18DC400373E99 /* refcountedobject.h */,
> 				413A21CC1FE18DA400373E99 /* refcounter.h */,
>-				413A22541FE18DE300373E99 /* rollingaccumulator.h */,
> 				413A217B1FE18D7600373E99 /* rollingaccumulator_unittest.cc */,
>+				413A22541FE18DE300373E99 /* rollingaccumulator.h */,
>+				413A22881FE18DFD00373E99 /* rtccertificate_unittest.cc */,
> 				413A21BB1FE18D9B00373E99 /* rtccertificate.cc */,
> 				413A21CF1FE18DA600373E99 /* rtccertificate.h */,
>-				413A22881FE18DFD00373E99 /* rtccertificate_unittest.cc */,
>+				413A21851FE18D7E00373E99 /* rtccertificategenerator_unittest.cc */,
> 				413A21651FE18D6100373E99 /* rtccertificategenerator.cc */,
> 				413A21861FE18D7F00373E99 /* rtccertificategenerator.h */,
>-				413A21851FE18D7E00373E99 /* rtccertificategenerator_unittest.cc */,
> 				413A21D41FE18DA900373E99 /* sanitizer.h */,
> 				413A21B01FE18D9600373E99 /* scoped_ref_ptr.h */,
>-				413A217E1FE18D7800373E99 /* sequenced_task_checker.h */,
> 				413A224A1FE18DDF00373E99 /* sequenced_task_checker_impl.cc */,
> 				413A225B1FE18DE600373E99 /* sequenced_task_checker_impl.h */,
> 				413A21BD1FE18D9C00373E99 /* sequenced_task_checker_unittest.cc */,
>+				413A217E1FE18D7800373E99 /* sequenced_task_checker.h */,
>+				413A22221FE18DCB00373E99 /* signalthread_unittest.cc */,
> 				413A22081FE18DC000373E99 /* signalthread.cc */,
> 				413A222E1FE18DD100373E99 /* signalthread.h */,
>-				413A22221FE18DCB00373E99 /* signalthread_unittest.cc */,
>-				413A21761FE18D7100373E99 /* sigslot.h */,
> 				413A21A21FE18D8F00373E99 /* sigslot_unittest.cc */,
>+				413A21761FE18D7100373E99 /* sigslot.h */,
> 				413A21DF1FE18DAE00373E99 /* sigslotrepeater.h */,
>-				413A22671FE18DEC00373E99 /* sigslottester.h */,
> 				413A22461FE18DDD00373E99 /* sigslottester_unittest.cc */,
>-				413A21881FE18D8000373E99 /* socket.h */,
>+				413A22671FE18DEC00373E99 /* sigslottester.h */,
> 				413A21BA1FE18D9A00373E99 /* socket_unittest.cc */,
> 				413A21871FE18D8000373E99 /* socket_unittest.h */,
>+				41DDB2552126790A00296D47 /* socket.cc */,
>+				413A21881FE18D8000373E99 /* socket.h */,
> 				413A219B1FE18D8B00373E99 /* socketadapters.cc */,
> 				413A220E1FE18DC200373E99 /* socketadapters.h */,
>+				413A22101FE18DC300373E99 /* socketaddress_unittest.cc */,
> 				413A22151FE18DC500373E99 /* socketaddress.cc */,
> 				413A22761FE18DF700373E99 /* socketaddress.h */,
>-				413A22101FE18DC300373E99 /* socketaddress_unittest.cc */,
> 				413A21F91FE18DB900373E99 /* socketaddresspair.cc */,
> 				413A22771FE18DF700373E99 /* socketaddresspair.h */,
> 				413A216C1FE18D6900373E99 /* socketfactory.h */,
> 				413A22661FE18DEC00373E99 /* socketserver.h */,
> 				413A217C1FE18D7700373E99 /* socketstream.cc */,
> 				413A22001FE18DBC00373E99 /* socketstream.h */,
>+				413A22341FE18DD400373E99 /* ssladapter_unittest.cc */,
> 				413A22251FE18DCD00373E99 /* ssladapter.cc */,
> 				413A224F1FE18DE100373E99 /* ssladapter.h */,
>-				413A22341FE18DD400373E99 /* ssladapter_unittest.cc */,
>+				41E02CB92127356A00C27CD6 /* sslcertificate.cc */,
>+				41E02CB82127356900C27CD6 /* sslcertificate.h */,
> 				413A229F1FE18E0600373E99 /* sslfingerprint.cc */,
> 				413A225C1FE18DE600373E99 /* sslfingerprint.h */,
>+				413A226C1FE18DF000373E99 /* sslidentity_unittest.cc */,
> 				413A21601FE18D5B00373E99 /* sslidentity.cc */,
> 				413A218F1FE18D8400373E99 /* sslidentity.h */,
>-				413A226C1FE18DF000373E99 /* sslidentity_unittest.cc */,
> 				413A223F1FE18DDA00373E99 /* sslroots.h */,
>+				413A21EB1FE18DB300373E99 /* sslstreamadapter_unittest.cc */,
> 				413A21AA1FE18D9300373E99 /* sslstreamadapter.cc */,
> 				413A226E1FE18DF100373E99 /* sslstreamadapter.h */,
>-				413A21EB1FE18DB300373E99 /* sslstreamadapter_unittest.cc */,
>+				413A22501FE18DE100373E99 /* stream_unittest.cc */,
> 				413A22801FE18DFA00373E99 /* stream.cc */,
> 				413A21A81FE18D9200373E99 /* stream.h */,
>-				413A22501FE18DE100373E99 /* stream_unittest.cc */,
>+				413A21781FE18D7300373E99 /* string_to_number_unittest.cc */,
> 				413A22831FE18DFC00373E99 /* string_to_number.cc */,
> 				413A22111FE18DC300373E99 /* string_to_number.h */,
>-				413A21781FE18D7300373E99 /* string_to_number_unittest.cc */,
>+				413A22031FE18DBE00373E99 /* stringencode_unittest.cc */,
> 				413A215B1FE18D5700373E99 /* stringencode.cc */,
> 				413A21961FE18D8900373E99 /* stringencode.h */,
>-				413A22031FE18DBE00373E99 /* stringencode_unittest.cc */,
>-				413A225E1FE18DE700373E99 /* stringize_macros.h */,
> 				413A22471FE18DDD00373E99 /* stringize_macros_unittest.cc */,
>+				413A225E1FE18DE700373E99 /* stringize_macros.h */,
>+				41DDB267212679AE00296D47 /* strings */,
>+				413A22311FE18DD300373E99 /* stringutils_unittest.cc */,
> 				413A21BF1FE18D9C00373E99 /* stringutils.cc */,
> 				413A21771FE18D7200373E99 /* stringutils.h */,
>-				413A22311FE18DD300373E99 /* stringutils_unittest.cc */,
>-				413A21F31FE18DB700373E99 /* swap_queue.h */,
> 				413A21971FE18D8900373E99 /* swap_queue_unittest.cc */,
>-				413A21E51FE18DB100373E99 /* task_queue.h */,
>+				413A21F31FE18DB700373E99 /* swap_queue.h */,
>+				4192415C2127379200634FCF /* synchronization */,
>+				41F263A22126808800274F59 /* system */,
> 				413A228B1FE18DFE00373E99 /* task_queue_gcd.cc */,
> 				413A21741FE18D7000373E99 /* task_queue_libevent.cc */,
> 				413A21D91FE18DAB00373E99 /* task_queue_posix.cc */,
> 				413A222A1FE18DCF00373E99 /* task_queue_posix.h */,
> 				413A21B11FE18D9600373E99 /* task_queue_unittest.cc */,
> 				413A219E1FE18D8D00373E99 /* task_queue_win.cc */,
>+				413A21E51FE18DB100373E99 /* task_queue.h */,
> 				413A21A91FE18D9200373E99 /* template_util.h */,
> 				413A21441FE18D4C00373E99 /* testbase64.h */,
>+				413A22811FE18DFB00373E99 /* testclient_unittest.cc */,
> 				413A21D31FE18DA800373E99 /* testclient.cc */,
> 				413A22821FE18DFB00373E99 /* testclient.h */,
>-				413A22811FE18DFB00373E99 /* testclient_unittest.cc */,
> 				413A21491FE18D4E00373E99 /* testechoserver.cc */,
> 				413A21D61FE18DAA00373E99 /* testechoserver.h */,
> 				413A219F1FE18D8E00373E99 /* testutils.cc */,
> 				413A22861FE18DFC00373E99 /* testutils.h */,
>-				413A21901FE18D8500373E99 /* thread.cc */,
>-				413A21EE1FE18DB400373E99 /* thread.h */,
>-				413A216E1FE18D6A00373E99 /* thread_annotations.h */,
>+				41F263B4212680E800274F59 /* third_party */,
> 				413A22841FE18DFC00373E99 /* thread_annotations_unittest.cc */,
>-				413A21931FE18D8700373E99 /* thread_checker.h */,
>+				413A216E1FE18D6A00373E99 /* thread_annotations.h */,
> 				413A21FC1FE18DBB00373E99 /* thread_checker_impl.cc */,
> 				413A215E1FE18D5900373E99 /* thread_checker_impl.h */,
> 				413A21B61FE18D9800373E99 /* thread_checker_unittest.cc */,
>+				413A21931FE18D8700373E99 /* thread_checker.h */,
> 				413A228A1FE18DFE00373E99 /* thread_darwin.mm */,
> 				413A22261FE18DCD00373E99 /* thread_unittest.cc */,
>+				413A21901FE18D8500373E99 /* thread.cc */,
>+				413A21EE1FE18DB400373E99 /* thread.h */,
>+				413A224B1FE18DDF00373E99 /* timestampaligner_unittest.cc */,
> 				413A22281FE18DCE00373E99 /* timestampaligner.cc */,
> 				413A22011FE18DBD00373E99 /* timestampaligner.h */,
>-				413A224B1FE18DDF00373E99 /* timestampaligner_unittest.cc */,
>+				413A228E1FE18DFF00373E99 /* timeutils_unittest.cc */,
> 				413A22701FE18DF300373E99 /* timeutils.cc */,
> 				413A21F51FE18DB700373E99 /* timeutils.h */,
>-				413A228E1FE18DFF00373E99 /* timeutils_unittest.cc */,
> 				413A21AB1FE18D9300373E99 /* trace_event.h */,
> 				413A21F71FE18DB800373E99 /* type_traits.h */,
> 				413A224C1FE18DE000373E99 /* unittest_main.cc */,
>@@ -6043,8 +6844,11 @@
> 				413A21DA1FE18DAC00373E99 /* virtualsocket_unittest.cc */,
> 				413A21BE1FE18D9C00373E99 /* virtualsocketserver.cc */,
> 				413A21631FE18D5E00373E99 /* virtualsocketserver.h */,
>-				413A21941FE18D8800373E99 /* weak_ptr.cc */,
> 				413A22711FE18DF400373E99 /* weak_ptr_unittest.cc */,
>+				413A21941FE18D8800373E99 /* weak_ptr.cc */,
>+				41E02CBA2127356A00C27CD6 /* weak_ptr.h */,
>+				41DDB2572126792800296D47 /* zero_memory.cc */,
>+				41DDB2582126792800296D47 /* zero_memory.h */,
> 			);
> 			name = rtc_base;
> 			sourceTree = "<group>";
>@@ -6129,10 +6933,11 @@
> 		413AD19F21265ADF003F7263 /* absl */ = {
> 			isa = PBXGroup;
> 			children = (
>-				41DDB24A21265BA800296D47 /* types */,
>-				41DDB24521265B8200296D47 /* container */,
>-				413AD1A621265B35003F7263 /* base */,
> 				413AD1A021265AFA003F7263 /* algorithm */,
>+				413AD1A621265B35003F7263 /* base */,
>+				41DDB24521265B8200296D47 /* container */,
>+				41A08BB621268A6A001D5D7B /* memory */,
>+				41DDB24A21265BA800296D47 /* types */,
> 			);
> 			name = absl;
> 			path = "abseil-cpp/absl";
>@@ -6150,6 +6955,8 @@
> 		413AD1A621265B35003F7263 /* base */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41A08BCA2126961F001D5D7B /* raw_logging.cc */,
>+				41A08BC92126961F001D5D7B /* raw_logging.h */,
> 				413AD1A921265B59003F7263 /* casts.h */,
> 				413AD1A821265B59003F7263 /* dynamic_annotations.cc */,
> 				413AD1AA21265B59003F7263 /* macros.h */,
>@@ -6275,6 +7082,8 @@
> 		4145E4931EF890E500FCF6E6 /* Video */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41AF2D11212CD40300D033D5 /* UIDevice+H264Profile.h */,
>+				41053B4D212CD8EC004DAC6C /* UIDevice+H264Profile.mm */,
> 				41ECEAC520641302009D5141 /* RTCCVPixelBuffer.mm */,
> 				41ECEACE2064136F009D5141 /* RTCI420Buffer+Private.h */,
> 				41ECEACD2064136F009D5141 /* RTCI420Buffer.mm */,
>@@ -6336,6 +7145,158 @@
> 			name = Common;
> 			sourceTree = "<group>";
> 		};
>+		4192415C2127379200634FCF /* synchronization */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419241602127385B00634FCF /* rw_lock_posix.cc */,
>+				4192415E2127385B00634FCF /* rw_lock_posix.h */,
>+				4192415D2127385A00634FCF /* rw_lock_wrapper.cc */,
>+				4192415F2127385B00634FCF /* rw_lock_wrapper.h */,
>+			);
>+			name = synchronization;
>+			path = rtc_base/synchronization;
>+			sourceTree = "<group>";
>+		};
>+		419241652127387900634FCF /* experiments */ = {
>+			isa = PBXGroup;
>+			children = (
>+				4192416B2127389B00634FCF /* congestion_controller_experiment.cc */,
>+				419241672127389A00634FCF /* congestion_controller_experiment.h */,
>+				419241682127389A00634FCF /* field_trial_parser.cc */,
>+				419241692127389B00634FCF /* field_trial_parser.h */,
>+				419241E621275A7600634FCF /* field_trial_units.cc */,
>+				419241E521275A7600634FCF /* field_trial_units.h */,
>+				4192416A2127389B00634FCF /* quality_scaling_experiment.cc */,
>+				419241662127389A00634FCF /* quality_scaling_experiment.h */,
>+			);
>+			name = experiments;
>+			path = rtc_base/experiments;
>+			sourceTree = "<group>";
>+		};
>+		4192417A2127390C00634FCF /* time */ = {
>+			isa = PBXGroup;
>+			children = (
>+				4192417B2127392300634FCF /* timestamp_extrapolator.cc */,
>+				4192417C2127392300634FCF /* timestamp_extrapolator.h */,
>+			);
>+			name = time;
>+			path = rtc_base/time;
>+			sourceTree = "<group>";
>+		};
>+		419241912127583900634FCF /* rnnnoise */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419241922127585000634FCF /* src */,
>+			);
>+			name = rnnnoise;
>+			path = rnnoise;
>+			sourceTree = "<group>";
>+		};
>+		419241922127585000634FCF /* src */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419241952127586500634FCF /* kiss_fft.cc */,
>+				419241962127586500634FCF /* kiss_fft.h */,
>+				419241972127586500634FCF /* rnn_activations.h */,
>+				419241942127586400634FCF /* rnn_vad_weights.cc */,
>+				419241932127586400634FCF /* rnn_vad_weights.h */,
>+			);
>+			name = src;
>+			sourceTree = "<group>";
>+		};
>+		419241B12127591C00634FCF /* bbr */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419241B62127593D00634FCF /* bandwidth_sampler.cc */,
>+				419241BE2127593F00634FCF /* bandwidth_sampler.h */,
>+				419241BB2127593E00634FCF /* bbr_factory.cc */,
>+				419241B82127593E00634FCF /* bbr_factory.h */,
>+				419241B92127593E00634FCF /* bbr_network_controller.cc */,
>+				419241B52127593D00634FCF /* bbr_network_controller.h */,
>+				419241B32127593D00634FCF /* data_transfer_tracker.cc */,
>+				419241BF2127593F00634FCF /* data_transfer_tracker.h */,
>+				419241B72127593D00634FCF /* loss_rate_filter.cc */,
>+				419241B22127593C00634FCF /* loss_rate_filter.h */,
>+				419241BA2127593E00634FCF /* packet_number_indexed_queue.h */,
>+				419241BC2127593E00634FCF /* rtt_stats.cc */,
>+				419241BD2127593E00634FCF /* rtt_stats.h */,
>+				419241B42127593D00634FCF /* windowed_filter.h */,
>+			);
>+			path = bbr;
>+			sourceTree = "<group>";
>+		};
>+		419241D421275A1800634FCF /* units */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419241D921275A2F00634FCF /* data_rate.cc */,
>+				419241D621275A2F00634FCF /* data_rate.h */,
>+				419241D721275A2F00634FCF /* data_size.cc */,
>+				419241DC21275A3000634FCF /* data_size.h */,
>+				419241DA21275A3000634FCF /* time_delta.cc */,
>+				419241D521275A2F00634FCF /* time_delta.h */,
>+				419241DB21275A3000634FCF /* timestamp.cc */,
>+				419241D821275A2F00634FCF /* timestamp.h */,
>+			);
>+			path = units;
>+			sourceTree = "<group>";
>+		};
>+		419241F7212765EE00634FCF /* Native */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419241F82127660E00634FCF /* api */,
>+				419241FA2127661700634FCF /* src */,
>+			);
>+			name = Native;
>+			path = sdk/objc/Framework/Native;
>+			sourceTree = "<group>";
>+		};
>+		419241F82127660E00634FCF /* api */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419241FB2127664100634FCF /* audio_device_module.h */,
>+				419241FC2127664100634FCF /* audio_device_module.mm */,
>+				419242062127664600634FCF /* video_capturer.h */,
>+				419241FD2127664100634FCF /* video_capturer.mm */,
>+				419242012127664200634FCF /* video_decoder_factory.h */,
>+				419242032127664200634FCF /* video_decoder_factory.mm */,
>+				419242042127664500634FCF /* video_encoder_factory.h */,
>+				419242052127664500634FCF /* video_encoder_factory.mm */,
>+				419242082127664700634FCF /* video_frame_buffer.h */,
>+				419242072127664600634FCF /* video_frame_buffer.mm */,
>+				419242002127664200634FCF /* video_frame.h */,
>+				419241FF2127664100634FCF /* video_frame.mm */,
>+				419242022127664200634FCF /* video_renderer.h */,
>+				419241FE2127664100634FCF /* video_renderer.mm */,
>+			);
>+			path = api;
>+			sourceTree = "<group>";
>+		};
>+		419241FA2127661700634FCF /* src */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419242322127665A00634FCF /* audio_device_ios.h */,
>+				4192422F2127665900634FCF /* audio_device_ios.mm */,
>+				419242312127665900634FCF /* audio_device_module_ios.h */,
>+				419242332127665A00634FCF /* audio_device_module_ios.mm */,
>+				419242302127665900634FCF /* audio_session_observer.h */,
>+				419242352127665A00634FCF /* voice_processing_audio_unit.h */,
>+				419242342127665A00634FCF /* voice_processing_audio_unit.mm */,
>+				4192421B2127664C00634FCF /* objc_frame_buffer.h */,
>+				4192421D2127664D00634FCF /* objc_frame_buffer.mm */,
>+				419242182127664B00634FCF /* objc_video_decoder_factory.h */,
>+				4192421A2127664C00634FCF /* objc_video_decoder_factory.mm */,
>+				419242222127664E00634FCF /* objc_video_encoder_factory.h */,
>+				419242172127664B00634FCF /* objc_video_encoder_factory.mm */,
>+				419242192127664B00634FCF /* objc_video_frame.h */,
>+				4192421E2127664D00634FCF /* objc_video_frame.mm */,
>+				419242212127664E00634FCF /* objc_video_renderer.h */,
>+				4192421F2127664D00634FCF /* objc_video_renderer.mm */,
>+				4192421C2127664C00634FCF /* objc_video_track_source.h */,
>+				419242202127664D00634FCF /* objc_video_track_source.mm */,
>+			);
>+			path = src;
>+			sourceTree = "<group>";
>+		};
> 		419C82E71FE20E7D0040C30F /* opus */ = {
> 			isa = PBXGroup;
> 			children = (
>@@ -6416,6 +7377,8 @@
> 				419C834B1FE246640040C30F /* safe_compare.h */,
> 				419C83471FE246630040C30F /* safe_conversions.h */,
> 				419C83481FE246630040C30F /* safe_minmax.h */,
>+				41E02CD22127360700C27CD6 /* sample_counter.cc */,
>+				41E02CD12127360600C27CD6 /* sample_counter.h */,
> 			);
> 			name = numerics;
> 			sourceTree = "<group>";
>@@ -6549,6 +7512,59 @@
> 			path = audio_codecs/ilbc;
> 			sourceTree = "<group>";
> 		};
>+		41A08BB621268A6A001D5D7B /* memory */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41A08BB721268A7D001D5D7B /* memory.h */,
>+			);
>+			name = memory;
>+			sourceTree = "<group>";
>+		};
>+		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 */,
>+				41D6B46D212731A4008F9353 /* agc2_common.h */,
>+				41D6B45E212731A0008F9353 /* biquad_filter.cc */,
>+				41D6B463212731A2008F9353 /* biquad_filter.h */,
>+				41A08BE621272F86001D5D7B /* compute_interpolated_gain_curve.cc */,
>+				41A08BDA21272F84001D5D7B /* compute_interpolated_gain_curve.h */,
>+				41A08BE821272F86001D5D7B /* down_sampler.cc */,
>+				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 */,
>+				41A08BEA21272F86001D5D7B /* limiter.h */,
>+				41D6B470212731A6008F9353 /* limiter.cc */,
>+				41D6B461212731A1008F9353 /* noise_level_estimator.cc */,
>+				41D6B46A212731A3008F9353 /* noise_spectrum_estimator.cc */,
>+				41A08BE221272F85001D5D7B /* noise_spectrum_estimator.h */,
>+				41D6B468212731A3008F9353 /* saturation_protector.cc */,
>+				41D6B46B212731A3008F9353 /* saturation_protector.h */,
>+				41D6B46F212731A4008F9353 /* signal_classifier.cc */,
>+				41A08BEB21272F87001D5D7B /* signal_classifier.h */,
>+				41D6B46C212731A4008F9353 /* vad_with_level.cc */,
>+				41A08BE321272F85001D5D7B /* vad_with_level.h */,
>+				41D6B466212731A2008F9353 /* vector_float_frame.cc */,
>+				41A08BDD21272F84001D5D7B /* vector_float_frame.h */,
>+			);
>+			path = agc2;
>+			sourceTree = "<group>";
>+		};
> 		41A391431EFC446E00C4516A /* sha */ = {
> 			isa = PBXGroup;
> 			children = (
>@@ -6811,12 +7827,81 @@
> 		41DDB24A21265BA800296D47 /* types */ = {
> 			isa = PBXGroup;
> 			children = (
>+				4192417F2127497100634FCF /* bad_variant_access.cc */,
>+				419241802127497100634FCF /* bad_variant_access.h */,
>+				41A08BC6212695DE001D5D7B /* bad_optional_access.cc */,
>+				41A08BC5212695DE001D5D7B /* bad_optional_access.h */,
> 				41DDB25021265BE900296D47 /* optional.cc */,
> 				41DDB24F21265BE900296D47 /* optional.h */,
> 			);
> 			name = types;
> 			sourceTree = "<group>";
> 		};
>+		41DDB267212679AE00296D47 /* strings */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41DDB26B212679D200296D47 /* audio_format_to_string.cc */,
>+				41DDB269212679D100296D47 /* audio_format_to_string.h */,
>+				41DDB26A212679D200296D47 /* string_builder.cc */,
>+				41DDB268212679D100296D47 /* string_builder.h */,
>+			);
>+			name = strings;
>+			path = rtc_base/strings;
>+			sourceTree = "<group>";
>+		};
>+		41E02C8F212734E800C27CD6 /* goog_cc */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41E02C942127352A00C27CD6 /* acknowledged_bitrate_estimator.cc */,
>+				41E02C922127352A00C27CD6 /* acknowledged_bitrate_estimator.h */,
>+				41E02CA12127352C00C27CD6 /* alr_detector.cc */,
>+				41E02C992127352B00C27CD6 /* alr_detector.h */,
>+				41E02C9A2127352B00C27CD6 /* bitrate_estimator.cc */,
>+				41E02C982127352B00C27CD6 /* bitrate_estimator.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 */,
>+				41E02C9B2127352B00C27CD6 /* median_slope_estimator.h */,
>+				41E02C9E2127352C00C27CD6 /* probe_bitrate_estimator.cc */,
>+				41E02C932127352A00C27CD6 /* probe_bitrate_estimator.h */,
>+				41E02CA02127352C00C27CD6 /* probe_controller.cc */,
>+				41E02C902127352900C27CD6 /* probe_controller.h */,
>+				41E02C9C2127352B00C27CD6 /* trendline_estimator.cc */,
>+				41E02C9F2127352C00C27CD6 /* trendline_estimator.h */,
>+			);
>+			path = goog_cc;
>+			sourceTree = "<group>";
>+		};
>+		41E02CC62127359F00C27CD6 /* transport */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41E02CC7212735B600C27CD6 /* bitrate_settings.cc */,
>+				41E02CCB212735B700C27CD6 /* bitrate_settings.h */,
>+				41E02CCA212735B700C27CD6 /* network_control.h */,
>+				41E02CC8212735B700C27CD6 /* network_types.cc */,
>+				41E02CC9212735B700C27CD6 /* network_types.h */,
>+			);
>+			path = transport;
>+			sourceTree = "<group>";
>+		};
>+		41E02CD52127362300C27CD6 /* rtp */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41924189212749C800634FCF /* pacer_controller.h */,
>+				41924188212749C700634FCF /* send_side_congestion_controller.cc */,
>+				41924187212749C700634FCF /* send_time_history.h */,
>+				4192418A212749C800634FCF /* transport_feedback_adapter.cc */,
>+				41E02CD62127363C00C27CD6 /* pacer_controller.cc */,
>+				41E02CD82127363D00C27CD6 /* send_time_history.cc */,
>+				41E02CD72127363C00C27CD6 /* transport_feedback_adapter.h */,
>+			);
>+			path = rtp;
>+			sourceTree = "<group>";
>+		};
> 		41EA53A21EFC2BAF002FF04C /* fipsmodule */ = {
> 			isa = PBXGroup;
> 			children = (
>@@ -6972,6 +8057,7 @@
> 		41ECEAB2206300E3009D5141 /* PeerConnection */ = {
> 			isa = PBXGroup;
> 			children = (
>+				4192423F21276D4700634FCF /* RTCPeerConnectionFactory.mm */,
> 				41ECEAC720641328009D5141 /* RTCEncodedImage.mm */,
> 				41ECEAD9206414A6009D5141 /* RTCRtpFragmentationHeader.mm */,
> 				41ECEAB320630107009D5141 /* RTCVideoCodec+Private.h */,
>@@ -6987,6 +8073,63 @@
> 			name = PeerConnection;
> 			sourceTree = "<group>";
> 		};
>+		41F2636B21267B2300274F59 /* third_party */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41F2637121267B4A00274F59 /* spl_sqrt_floor.c */,
>+				41F2637021267B4A00274F59 /* spl_sqrt_floor.h */,
>+				41F2636C21267B3D00274F59 /* fft4g.c */,
>+				41F2636D21267B3E00274F59 /* fft4g.h */,
>+			);
>+			path = third_party;
>+			sourceTree = "<group>";
>+		};
>+		41F2638621267F2500274F59 /* third_party */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41F2639521267F5E00274F59 /* pa_memorybarrier.h */,
>+				41F2639721267F5E00274F59 /* pa_ringbuffer.c */,
>+				41F2639621267F5E00274F59 /* pa_ringbuffer.h */,
>+				41F2639121267F5300274F59 /* g722_decode.c */,
>+				41F2639021267F5300274F59 /* g722_enc_dec.h */,
>+				41F2638F21267F5200274F59 /* g722_encode.c */,
>+				41F2638B21267F4900274F59 /* g711.c */,
>+				41F2638C21267F4900274F59 /* g711.h */,
>+				41F2638821267F4000274F59 /* fft.c */,
>+				41F2638721267F4000274F59 /* fft.h */,
>+			);
>+			path = third_party;
>+			sourceTree = "<group>";
>+		};
>+		41F263A22126808800274F59 /* system */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41F263AB212680A800274F59 /* arch.h */,
>+				41F263A9212680A700274F59 /* asm_defines.h */,
>+				41F263A8212680A700274F59 /* BUILD.gn */,
>+				41F263A5212680A700274F59 /* fallthrough.h */,
>+				41F263A3212680A600274F59 /* file_wrapper.cc */,
>+				41F263AA212680A800274F59 /* file_wrapper.h */,
>+				41F263A4212680A600274F59 /* ignore_warnings.h */,
>+				41F263A6212680A700274F59 /* inline.h */,
>+				41F263A7212680A700274F59 /* unused.h */,
>+			);
>+			name = system;
>+			path = rtc_base/system;
>+			sourceTree = "<group>";
>+		};
>+		41F263B4212680E800274F59 /* third_party */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41F263B92126811400274F59 /* sigslot.cc */,
>+				41F263BA2126811400274F59 /* sigslot.h */,
>+				41F263B52126810B00274F59 /* base64.cc */,
>+				41F263B62126810C00274F59 /* base64.h */,
>+			);
>+			name = third_party;
>+			path = rtc_base/third_party;
>+			sourceTree = "<group>";
>+		};
> 		41F411BD1EF8DB8200343C26 /* vp8 */ = {
> 			isa = PBXGroup;
> 			children = (
>@@ -7539,7 +8682,9 @@
> 				5C4B48AB1E42C1E3002651C8 /* fakevideorenderer.h */,
> 				4145E4BD1EF894F500FCF6E6 /* h264_profile_level_id.cc */,
> 				4145E4BC1EF894F100FCF6E6 /* h264_profile_level_id.h */,
>+				4102F6B121273330006AE8D7 /* mediachannel.cc */,
> 				5C4B48AD1E42C1E3002651C8 /* mediachannel.h */,
>+				4102F6B021273330006AE8D7 /* mediaconfig.h */,
> 				5C4B48AE1E42C1E3002651C8 /* mediaconstants.cc */,
> 				5C4B48AF1E42C1E3002651C8 /* mediaconstants.h */,
> 				5C4B48B01E42C1E3002651C8 /* mediaengine.cc */,
>@@ -7570,6 +8715,8 @@
> 		5C4B484B1E42C199002651C8 /* engine */ = {
> 			isa = PBXGroup;
> 			children = (
>+				419241F321275C3200634FCF /* vp8_encoder_simulcast_proxy.cc */,
>+				419241F221275C3100634FCF /* vp8_encoder_simulcast_proxy.h */,
> 				41F411C61EF972F100343C26 /* adm_helpers.cc */,
> 				41F411C51EF972EC00343C26 /* adm_helpers.h */,
> 				5CD284C01E6A5FFC0094FDC8 /* apm_helpers.cc */,
>@@ -7768,6 +8915,7 @@
> 			children = (
> 				5C4B4B5A1E431C63002651C8 /* Classes */,
> 				413A24001FE198BE00373E99 /* Headers */,
>+				419241F7212765EE00634FCF /* Native */,
> 			);
> 			name = Framework;
> 			sourceTree = "<group>";
>@@ -7804,6 +8952,7 @@
> 		5C4B4C221E431F81002651C8 /* common_audio */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41F2636B21267B2300274F59 /* third_party */,
> 				5CDD87431E43BAE900621E92 /* resampler */,
> 				5CDD86091E43B8A300621E92 /* signal_processing */,
> 				5CDD86C41E43BA1800621E92 /* vad */,
>@@ -7947,6 +9096,7 @@
> 		5C63F8CB1E416CC3002CA531 /* third_party */ = {
> 			isa = PBXGroup;
> 			children = (
>+				419241912127583900634FCF /* rnnnoise */,
> 				413AD19E21265ACB003F7263 /* abseil-cpp */,
> 				5C63FC631E4184C0002CA531 /* boringssl */,
> 				5C63F8CC1E416CEB002CA531 /* jsoncpp */,
>@@ -8001,19 +9151,26 @@
> 		5C63F8E21E41732B002CA531 /* api */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41F9BF792051C7FD00ABF0B9 /* array_view.h */,
> 				5CD2853B1E6A61BE0094FDC8 /* audio_codecs */,
>+				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 */,
>-				4145E4C01EF8962800FCF6E6 /* stats */,
>-				5CD284601E6A57DD0094FDC8 /* video */,
>-				412455581EF88AD900F11809 /* video_codecs */,
>-				41F9BF792051C7FD00ABF0B9 /* array_view.h */,
> 				419C830E1FE2427F0040C30F /* candidate.cc */,
> 				41F9BF772051C7FD00ABF0B9 /* candidate.h */,
> 				41F9BF822051C80000ABF0B9 /* cryptoparams.h */,
>+				41F2637721267B7600274F59 /* datachannelinterface.cc */,
> 				5C63F8E71E41737B002CA531 /* datachannelinterface.h */,
> 				5C63F8EB1E41737B002CA531 /* dtmfsenderinterface.h */,
>+				415F1FC8212730F000064CBF /* echo_control.h */,
>+				41F2637621267B7600274F59 /* fec_controller.h */,
> 				41F9BF782051C7FD00ABF0B9 /* jsep.cc */,
> 				5C63F8EF1E41737B002CA531 /* jsep.h */,
>+				41F2637B21267B7700274F59 /* jsepicecandidate.cc */,
> 				5C63F8F11E41737B002CA531 /* jsepicecandidate.h */,
> 				5C63F8F31E41737B002CA531 /* jsepsessiondescription.h */,
> 				5C63F8F61E41737B002CA531 /* mediaconstraintsinterface.cc */,
>@@ -8026,6 +9183,7 @@
> 				5CD286311E6A67BF0094FDC8 /* mediatypes.h */,
> 				5C63F9021E41737B002CA531 /* notifier.h */,
> 				5C63F9081E41737B002CA531 /* peerconnectionfactoryproxy.h */,
>+				41F2637421267B7500274F59 /* peerconnectioninterface.cc */,
> 				5C63F9091E41737B002CA531 /* peerconnectioninterface.h */,
> 				5C63F90A1E41737B002CA531 /* peerconnectionproxy.h */,
> 				419C830F1FE2427F0040C30F /* proxy.cc */,
>@@ -8038,14 +9196,21 @@
> 				41F9BF7F2051C7FF00ABF0B9 /* rtp_headers.h */,
> 				419C83111FE2427F0040C30F /* rtpparameters.cc */,
> 				5C63F9151E41737B002CA531 /* rtpparameters.h */,
>+				41F2637A21267B7700274F59 /* rtpreceiverinterface.cc */,
> 				5C63F9181E41737B002CA531 /* rtpreceiverinterface.h */,
> 				5C63F91B1E41737B002CA531 /* rtpsenderinterface.h */,
>+				41F2637C21267B7700274F59 /* rtptransceiverinterface.cc */,
> 				41F9BF802051C7FF00ABF0B9 /* rtptransceiverinterface.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;
>@@ -8065,6 +9230,8 @@
> 			children = (
> 				5C63F98A1E4174F6002CA531 /* asyncstuntcpsocket.cc */,
> 				5C63F98B1E4174F6002CA531 /* asyncstuntcpsocket.h */,
>+				4192414A2127376D00634FCF /* basicasyncresolverfactory.cc */,
>+				419241512127376E00634FCF /* basicasyncresolverfactory.h */,
> 				5C63F98C1E4174F6002CA531 /* basicpacketsocketfactory.cc */,
> 				5C63F98D1E4174F6002CA531 /* basicpacketsocketfactory.h */,
> 				5C63F98F1E4174F6002CA531 /* candidatepairinterface.h */,
>@@ -8072,8 +9239,13 @@
> 				419C83D41FE247F10040C30F /* dtlstransport.h */,
> 				419C83D51FE247F10040C30F /* dtlstransportinternal.cc */,
> 				419C83D61FE247F10040C30F /* dtlstransportinternal.h */,
>+				4192414C2127376E00634FCF /* fakecandidatepair.h */,
>+				419241492127376D00634FCF /* fakedtlstransport.h */,
>+				4192414F2127376E00634FCF /* fakeicetransport.h */,
>+				4192414E2127376E00634FCF /* fakepackettransport.h */,
> 				5C63F9951E4174F6002CA531 /* fakeportallocator.h */,
> 				419C83D71FE247F20040C30F /* icetransportinternal.cc */,
>+				419241502127376E00634FCF /* icetransportinternal.h */,
> 				5C63F9971E4174F6002CA531 /* p2pconstants.cc */,
> 				5C63F9981E4174F6002CA531 /* p2pconstants.h */,
> 				5C63F99C1E4174F6002CA531 /* p2ptransportchannel.cc */,
>@@ -8093,6 +9265,8 @@
> 				5C63F9A61E4174F6002CA531 /* portinterface.h */,
> 				5C63F9A81E4174F6002CA531 /* pseudotcp.cc */,
> 				5C63F9A91E4174F6002CA531 /* pseudotcp.h */,
>+				419241482127376D00634FCF /* regatheringcontroller.cc */,
>+				4192414B2127376D00634FCF /* regatheringcontroller.h */,
> 				5C63F9AB1E4174F6002CA531 /* relayport.cc */,
> 				5C63F9AC1E4174F6002CA531 /* relayport.h */,
> 				5C63F9AE1E4174F6002CA531 /* relayserver.cc */,
>@@ -8122,6 +9296,7 @@
> 				5C63F9DA1E4174F6002CA531 /* turnserver.h */,
> 				5C63F9DB1E4174F6002CA531 /* udpport.h */,
> 				419C83D31FE247F00040C30F /* udptransport.cc */,
>+				4192414D2127376E00634FCF /* udptransport.h */,
> 			);
> 			path = base;
> 			sourceTree = "<group>";
>@@ -8152,6 +9327,8 @@
> 				5CD284711E6A5D080094FDC8 /* jsepsessiondescription.cc */,
> 				41F9BFDD2051DE1700ABF0B9 /* jseptransport.cc */,
> 				41F9BFDB2051DE1600ABF0B9 /* jseptransport.h */,
>+				4102F68721273205006AE8D7 /* jseptransportcontroller.cc */,
>+				4102F68521273204006AE8D7 /* jseptransportcontroller.h */,
> 				5CD284CA1E6A60570094FDC8 /* localaudiosource.cc */,
> 				5CD284CB1E6A60570094FDC8 /* localaudiosource.h */,
> 				5C63FA4B1E41761F002CA531 /* mediasession.cc */,
>@@ -8165,6 +9342,7 @@
> 				5CD284D41E6A60570094FDC8 /* peerconnection.h */,
> 				5CD2847D1E6A5E740094FDC8 /* peerconnectionfactory.cc */,
> 				5CD2847E1E6A5E740094FDC8 /* peerconnectionfactory.h */,
>+				4102F68C21273206006AE8D7 /* peerconnectioninternal.h */,
> 				41F9BFDC2051DE1700ABF0B9 /* peerconnectionwrapper.cc */,
> 				41F9BFDA2051DE1600ABF0B9 /* peerconnectionwrapper.h */,
> 				5CD284D91E6A60570094FDC8 /* remoteaudiosource.cc */,
>@@ -8173,8 +9351,12 @@
> 				5C63FA511E41761F002CA531 /* rtcpmuxfilter.h */,
> 				5CD284DB1E6A60570094FDC8 /* rtcstatscollector.cc */,
> 				5CD284DC1E6A60570094FDC8 /* rtcstatscollector.h */,
>+				4102F68921273205006AE8D7 /* rtcstatstraversal.cc */,
>+				4102F68421273204006AE8D7 /* rtcstatstraversal.h */,
> 				419C82C71FE20E2B0040C30F /* rtpmediautils.cc */,
> 				419C82C81FE20E2C0040C30F /* rtpmediautils.h */,
>+				4102F68A21273206006AE8D7 /* rtpparametersconversion.cc */,
>+				4102F68821273205006AE8D7 /* rtpparametersconversion.h */,
> 				5CD284DD1E6A60570094FDC8 /* rtpreceiver.cc */,
> 				5CD284DE1E6A60570094FDC8 /* rtpreceiver.h */,
> 				5CD284DF1E6A60570094FDC8 /* rtpsender.cc */,
>@@ -8202,6 +9384,8 @@
> 				5CD284E51E6A60570094FDC8 /* streamcollection.h */,
> 				5CD284E61E6A60570094FDC8 /* trackmediainfomap.cc */,
> 				5CD284E71E6A60570094FDC8 /* trackmediainfomap.h */,
>+				4102F68621273205006AE8D7 /* transportstats.cc */,
>+				4102F68B21273206006AE8D7 /* transportstats.h */,
> 				5CD284E81E6A60570094FDC8 /* videocapturertracksource.cc */,
> 				5CD284E91E6A60570094FDC8 /* videocapturertracksource.h */,
> 				5CD284EA1E6A60570094FDC8 /* videotrack.cc */,
>@@ -8320,15 +9504,36 @@
> 		5CD284601E6A57DD0094FDC8 /* video */ = {
> 			isa = PBXGroup;
> 			children = (
>+				4102F6D221273414006AE8D7 /* color_space.cc */,
>+				4102F6D821273416006AE8D7 /* color_space.h */,
>+				41F263C72126818A00274F59 /* encoded_frame.cc */,
>+				41F263C22126818900274F59 /* encoded_frame.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 */,
> 				419C840D1FE249AA0040C30F /* video_content_type.cc */,
> 				419C840F1FE249AB0040C30F /* video_content_type.h */,
>-				5CD284641E6A57F40094FDC8 /* video_frame.cc */,
>-				5CD284651E6A57F40094FDC8 /* video_frame.h */,
> 				4124554A1EF8874300F11809 /* video_frame_buffer.cc */,
> 				5CD284631E6A57F40094FDC8 /* video_frame_buffer.h */,
>+				5CD284641E6A57F40094FDC8 /* video_frame.cc */,
>+				5CD284651E6A57F40094FDC8 /* video_frame.h */,
> 				5CD284661E6A57F40094FDC8 /* video_rotation.h */,
>+				4102F6D321273415006AE8D7 /* video_sink_interface.h */,
>+				41F263C62126818A00274F59 /* video_source_interface.cc */,
>+				41F263BE2126818800274F59 /* video_source_interface.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 */,
>+				41F263BD2126818800274F59 /* video_stream_encoder_observer.h */,
>+				4102F6D721273415006AE8D7 /* video_stream_encoder_settings.h */,
> 				419C840C1FE249A90040C30F /* video_timing.cc */,
> 				419C840E1FE249AA0040C30F /* video_timing.h */,
> 			);
>@@ -8360,6 +9565,8 @@
> 				419C835A1FE246770040C30F /* isac */,
> 				419C83EB1FE248770040C30F /* L16 */,
> 				419C82E71FE20E7D0040C30F /* opus */,
>+				4192419E2127588200634FCF /* audio_codec_pair_id.cc */,
>+				4192419D2127588200634FCF /* audio_codec_pair_id.h */,
> 				5CD2853D1E6A61D20094FDC8 /* audio_decoder.cc */,
> 				5CD2853E1E6A61D20094FDC8 /* audio_decoder.h */,
> 				5CD2853C1E6A61D20094FDC8 /* audio_decoder_factory.h */,
>@@ -8506,6 +9713,7 @@
> 		5CDD83681E4399E400621E92 /* modules */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41F2638621267F2500274F59 /* third_party */,
> 				5CDD84051E439AF200621E92 /* audio_coding */,
> 				5C119FEF1E4573CA004F0987 /* audio_device */,
> 				5CD284811E6A5F2B0094FDC8 /* audio_mixer */,
>@@ -8524,6 +9732,14 @@
> 		5CDD83691E4399F400621E92 /* video_coding */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41E02C83212734B800C27CD6 /* decoder_database.cc */,
>+				41E02C7F212734B800C27CD6 /* decoder_database.h */,
>+				41E02C80212734B800C27CD6 /* encoder_database.cc */,
>+				41E02C81212734B800C27CD6 /* encoder_database.h */,
>+				41E02C82212734B800C27CD6 /* fec_controller_default.cc */,
>+				41E02C84212734B900C27CD6 /* fec_controller_default.h */,
>+				41E02C86212734B900C27CD6 /* fec_rate_table.h */,
>+				41E02C85212734B900C27CD6 /* video_coding_defines.cc */,
> 				5CDD853D1E43B3F800621E92 /* codecs */,
> 				5CDD836A1E439A0D00621E92 /* utility */,
> 				5CDD83891E439A6F00621E92 /* codec_timer.cc */,
>@@ -8593,6 +9809,10 @@
> 		5CDD836A1E439A0D00621E92 /* utility */ = {
> 			isa = PBXGroup;
> 			children = (
>+				419241EA21275AFA00634FCF /* simulcast_rate_allocator.cc */,
>+				419241EC21275AFA00634FCF /* simulcast_rate_allocator.h */,
>+				419241EB21275AFA00634FCF /* simulcast_utility.cc */,
>+				419241E921275AF900634FCF /* simulcast_utility.h */,
> 				5CD285371E6A61980094FDC8 /* default_video_bitrate_allocator.cc */,
> 				5CD285381E6A61980094FDC8 /* default_video_bitrate_allocator.h */,
> 				5CDD836B1E439A3500621E92 /* frame_dropper.cc */,
>@@ -8645,6 +9865,7 @@
> 		5CDD84881E43AEC900621E92 /* audio_processing */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41A08BD921272F3E001D5D7B /* agc2 */,
> 				5CDD86A91E43B9B900621E92 /* aec */,
> 				5CD2855F1E6A63050094FDC8 /* aec3 */,
> 				4130922A1EF8D73500757C55 /* aec_dump */,
>@@ -8665,10 +9886,16 @@
> 				5CDD848F1E43AF1300621E92 /* echo_cancellation_impl.h */,
> 				5CDD84901E43AF1300621E92 /* echo_control_mobile_impl.cc */,
> 				5CDD84911E43AF1300621E92 /* echo_control_mobile_impl.h */,
>+				415F1FBC2127304C00064CBF /* echo_cancellation_proxy.cc */,
>+				415F1FBB2127304C00064CBF /* echo_cancellation_proxy.h */,
>+				415F1FBA2127304B00064CBF /* echo_control_mobile_proxy.cc */,
>+				415F1FB92127304B00064CBF /* echo_control_mobile_proxy.h */,
> 				5CDD84921E43AF1300621E92 /* gain_control_for_experimental_agc.cc */,
> 				5CDD84931E43AF1300621E92 /* gain_control_for_experimental_agc.h */,
> 				5CDD84941E43AF1300621E92 /* gain_control_impl.cc */,
> 				5CDD84951E43AF1300621E92 /* gain_control_impl.h */,
>+				41A08BCE21272EE1001D5D7B /* gain_controller2.cc */,
>+				41A08BCD21272EE1001D5D7B /* gain_controller2.h */,
> 				5CDD84981E43AF1300621E92 /* level_estimator_impl.cc */,
> 				5CDD84991E43AF1300621E92 /* level_estimator_impl.h */,
> 				5CD2855B1E6A62ED0094FDC8 /* low_cut_filter.cc */,
>@@ -8748,37 +9975,58 @@
> 		5CDD851A1E43B38F00621E92 /* call */ = {
> 			isa = PBXGroup;
> 			children = (
>+				4102F6B621273381006AE8D7 /* audio_receive_stream.cc */,
> 				5CD2849D1E6A5F9F0094FDC8 /* audio_receive_stream.h */,
> 				5CD2849E1E6A5F9F0094FDC8 /* audio_send_stream.cc */,
> 				5CD2849F1E6A5F9F0094FDC8 /* audio_send_stream.h */,
>+				415F1FC52127308E00064CBF /* audio_state.cc */,
> 				5CD284A01E6A5F9F0094FDC8 /* audio_state.h */,
> 				5CDD851C1E43B39C00621E92 /* bitrate_allocator.cc */,
> 				5CDD851D1E43B39C00621E92 /* bitrate_allocator.h */,
>+				415F1FD72127313F00064CBF /* call_config.cc */,
>+				415F1FDA2127313F00064CBF /* call_config.h */,
> 				5CDD85221E43B39C00621E92 /* call.cc */,
> 				5CD284A61E6A5F9F0094FDC8 /* call.h */,
> 				4145E4E11EF8CD7200FCF6E6 /* callfactory.cc */,
> 				4145E4E31EF8CD7C00FCF6E6 /* callfactory.h */,
>-				5CDD85261E43B39C00621E92 /* flexfec_receive_stream.h */,
>+				4102F6B721273381006AE8D7 /* degraded_call.cc */,
>+				415F1FD22127313E00064CBF /* degraded_call.h */,
>+				41299B942127369000B3414B /* fake_network_pipe.cc */,
>+				41299B952127369100B3414B /* fake_network_pipe.h */,
> 				5CD284A81E6A5F9F0094FDC8 /* flexfec_receive_stream_impl.cc */,
> 				5CD284A91E6A5F9F0094FDC8 /* flexfec_receive_stream_impl.h */,
>+				4102F6B821273381006AE8D7 /* flexfec_receive_stream.cc */,
>+				5CDD85261E43B39C00621E92 /* flexfec_receive_stream.h */,
>+				415F1FD82127313F00064CBF /* packet_receiver.cc */,
>+				4102F6B521273381006AE8D7 /* packet_receiver.h */,
>+				415F1FD02127313D00064CBF /* receive_time_calculator.cc */,
>+				415F1FD52127313E00064CBF /* receive_time_calculator.h */,
> 				419C82A61FE20DC80040C30F /* rtcp_demuxer.cc */,
> 				419C829F1FE20DC60040C30F /* rtcp_demuxer.h */,
> 				419C82AA1FE20DCA0040C30F /* rtcp_packet_sink_interface.h */,
>+				415F1FD92127313F00064CBF /* rtp_bitrate_configurator.cc */,
> 				419C82A11FE20DC70040C30F /* rtp_config.cc */,
> 				419C82A51FE20DC80040C30F /* rtp_config.h */,
> 				413092001EF8D0A600757C55 /* rtp_demuxer.cc */,
> 				413091FF1EF8D0A400757C55 /* rtp_demuxer.h */,
> 				419C82A31FE20DC70040C30F /* rtp_packet_sink_interface.h */,
>+				4102F6AA212732B0006AE8D7 /* rtp_payload_params.cc */,
>+				415F1FD12127313E00064CBF /* rtp_payload_params.h */,
> 				419C82AC1FE20DCB0040C30F /* rtp_rtcp_demuxer_helper.cc */,
> 				419C82A21FE20DC70040C30F /* rtp_rtcp_demuxer_helper.h */,
>+				419C82AD1FE20DCB0040C30F /* rtp_stream_receiver_controller_interface.h */,
> 				419C82A91FE20DCA0040C30F /* rtp_stream_receiver_controller.cc */,
> 				419C82AE1FE20DCC0040C30F /* rtp_stream_receiver_controller.h */,
>-				419C82AD1FE20DCB0040C30F /* rtp_stream_receiver_controller_interface.h */,
>+				413092171EF8D63900757C55 /* rtp_transport_controller_send_interface.h */,
> 				413092181EF8D63F00757C55 /* rtp_transport_controller_send.cc */,
> 				413092161EF8D63400757C55 /* rtp_transport_controller_send.h */,
>-				413092171EF8D63900757C55 /* rtp_transport_controller_send_interface.h */,
>+				415F1FD62127313E00064CBF /* rtp_video_sender_interface.h */,
>+				415F1FD32127313E00064CBF /* rtp_video_sender.cc */,
>+				415F1FD42127313E00064CBF /* rtp_video_sender.h */,
> 				419C82A41FE20DC80040C30F /* rtx_receive_stream.cc */,
> 				419C82AF1FE20DCC0040C30F /* rtx_receive_stream.h */,
>+				419241CE2127597C00634FCF /* simulated_network.cc */,
>+				419241CF2127597C00634FCF /* simulated_network.h */,
> 				41F9BF9D2051C86900ABF0B9 /* ssrc_binding_observer.h */,
> 				5CD284AE1E6A5F9F0094FDC8 /* syncable.cc */,
> 				5CD284AF1E6A5F9F0094FDC8 /* syncable.h */,
>@@ -8814,6 +10062,13 @@
> 		5CDD85591E43B5AE00621E92 /* video */ = {
> 			isa = PBXGroup;
> 			children = (
>+				4102F6BF212733B6006AE8D7 /* video_quality_observer.cc */,
>+				4102F6C5212733B6006AE8D7 /* video_quality_observer.h */,
>+				4102F6C2212733B6006AE8D7 /* video_send_stream_impl.cc */,
>+				4102F6BE212733B5006AE8D7 /* video_send_stream_impl.h */,
>+				4102F6C1212733B6006AE8D7 /* video_stream_decoder_impl.cc */,
>+				4102F6C4212733B6006AE8D7 /* video_stream_decoder_impl.h */,
>+				4102F6C0212733B6006AE8D7 /* video_stream_encoder.h */,
> 				5CDD855C1E43B5C000621E92 /* call_stats.cc */,
> 				5CDD855D1E43B5C000621E92 /* call_stats.h */,
> 				5CDD85601E43B5C000621E92 /* encoder_rtcp_feedback.cc */,
>@@ -9095,6 +10350,9 @@
> 		5CDD876F1E43BB8D00621E92 /* source */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41299B8E2127367A00B3414B /* isac_vad.c */,
>+				41299B8F2127367B00B3414B /* isac_vad.h */,
>+				41299B902127367B00B3414B /* pitch_filter.h */,
> 				5CDD87731E43BC0500621E92 /* arith_routines.c */,
> 				5CDD87741E43BC0500621E92 /* arith_routines.h */,
> 				5CDD87711E43BC0500621E92 /* arith_routines_hist.c */,
>@@ -9177,11 +10435,14 @@
> 		5CDD87FA1E43BE2600621E92 /* source */ = {
> 			isa = PBXGroup;
> 			children = (
>-				5CDD88EB1E43BF2500621E92 /* rtcp_packet */,
>+				419241D2212759A100634FCF /* fec_private_tables_random.cc */,
>+				419241AF2127590200634FCF /* rtp_generic_frame_descriptor_extension.cc */,
> 				5CDD87FC1E43BE3C00621E92 /* byte_io.h */,
>+				4102F6EB21273460006AE8D7 /* contributing_sources.cc */,
> 				5CDD87FE1E43BE3C00621E92 /* dtmf_queue.cc */,
> 				5CDD87FF1E43BE3C00621E92 /* dtmf_queue.h */,
> 				5CDD88001E43BE3C00621E92 /* fec_private_tables_bursty.h */,
>+				4192418F2127581000634FCF /* fec_private_tables_bursty.cc */,
> 				5CDD88011E43BE3C00621E92 /* fec_private_tables_random.h */,
> 				5CDD88021E43BE3C00621E92 /* fec_test_helper.cc */,
> 				5CDD88031E43BE3C00621E92 /* fec_test_helper.h */,
>@@ -9189,10 +10450,10 @@
> 				5CDD88061E43BE3C00621E92 /* flexfec_header_reader_writer.h */,
> 				5CDD880A1E43BE3C00621E92 /* flexfec_receiver.cc */,
> 				5CDD880C1E43BE3C00621E92 /* flexfec_sender.cc */,
>-				5CDD880F1E43BE3C00621E92 /* forward_error_correction.cc */,
>-				5CDD88101E43BE3C00621E92 /* forward_error_correction.h */,
> 				5CDD880D1E43BE3C00621E92 /* forward_error_correction_internal.cc */,
> 				5CDD880E1E43BE3C00621E92 /* forward_error_correction_internal.h */,
>+				5CDD880F1E43BE3C00621E92 /* forward_error_correction.cc */,
>+				5CDD88101E43BE3C00621E92 /* forward_error_correction.h */,
> 				5CDD88151E43BE3C00621E92 /* packet_loss_stats.cc */,
> 				5CDD88161E43BE3C00621E92 /* packet_loss_stats.h */,
> 				5CDD88181E43BE3C00621E92 /* playout_delay_oracle.cc */,
>@@ -9202,20 +10463,19 @@
> 				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 */,
>-				5CDD883A1E43BE3C00621E92 /* rtp_format.cc */,
>-				5CDD883B1E43BE3C00621E92 /* rtp_format.h */,
>+				419C82F61FE20EFB0040C30F /* rtcp_transceiver.cc */,
>+				419C82FA1FE20EFD0040C30F /* rtcp_transceiver.h */,
> 				5CDD882E1E43BE3C00621E92 /* rtp_format_h264.cc */,
> 				5CDD882F1E43BE3C00621E92 /* rtp_format_h264.h */,
> 				5CDD88301E43BE3C00621E92 /* rtp_format_video_generic.cc */,
>@@ -9224,17 +10484,23 @@
> 				5CDD88361E43BE3C00621E92 /* rtp_format_vp8.h */,
> 				5CDD88381E43BE3C00621E92 /* rtp_format_vp9.cc */,
> 				5CDD88391E43BE3C00621E92 /* rtp_format_vp9.h */,
>+				5CDD883A1E43BE3C00621E92 /* rtp_format.cc */,
>+				5CDD883B1E43BE3C00621E92 /* rtp_format.h */,
>+				4102F6E82127345F006AE8D7 /* rtp_generic_frame_descriptor_extension.h */,
>+				4102F6E52127345E006AE8D7 /* rtp_generic_frame_descriptor.cc */,
>+				4102F6E62127345F006AE8D7 /* rtp_generic_frame_descriptor.h */,
> 				4145E4DB1EF8CCEE00FCF6E6 /* rtp_header_extension_map.cc */,
> 				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 */,
> 				5CDD884B1E43BE3C00621E92 /* rtp_payload_registry.cc */,
> 				5CDD884C1E43BE3C00621E92 /* rtp_receiver_audio.cc */,
> 				5CDD884D1E43BE3C00621E92 /* rtp_receiver_audio.h */,
>@@ -9247,14 +10513,16 @@
> 				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 */,
>+				4102F6EA2127345F006AE8D7 /* rtp_video_header.h */,
> 				5CDD88641E43BE3C00621E92 /* time_util.cc */,
> 				5CDD88651E43BE3C00621E92 /* time_util.h */,
> 				5CDD88661E43BE3C00621E92 /* tmmbr_help.cc */,
>@@ -9330,6 +10598,8 @@
> 		5CDD89881E43BF8A00621E92 /* neteq */ = {
> 			isa = PBXGroup;
> 			children = (
>+				4102F6AC212732E6006AE8D7 /* expand_uma_logger.cc */,
>+				4102F6AD212732E7006AE8D7 /* expand_uma_logger.h */,
> 				5CDD8A5F1E43C00000621E92 /* tools */,
> 				5CDD89891E43BFB300621E92 /* accelerate.cc */,
> 				5CDD898A1E43BFB300621E92 /* accelerate.h */,
>@@ -9440,7 +10710,8 @@
> 		5CDD8ABB1E43C22200621E92 /* audio */ = {
> 			isa = PBXGroup;
> 			children = (
>-				5CD2854A1E6A62090094FDC8 /* utility */,
>+				4102F69A21273261006AE8D7 /* audio_level.cc */,
>+				4102F69821273260006AE8D7 /* audio_level.h */,
> 				5CDD8ABD1E43C23900621E92 /* audio_receive_stream.cc */,
> 				5CDD8ABE1E43C23900621E92 /* audio_receive_stream.h */,
> 				5CDD8AC01E43C23900621E92 /* audio_send_stream.cc */,
>@@ -9449,11 +10720,20 @@
> 				5CDD8AC41E43C23900621E92 /* audio_state.h */,
> 				41F9BF992051C84B00ABF0B9 /* audio_transport_impl.cc */,
> 				41F9BF9A2051C84B00ABF0B9 /* audio_transport_impl.h */,
>+				4102F69C21273261006AE8D7 /* channel_proxy.cc */,
>+				4102F69921273261006AE8D7 /* channel_proxy.h */,
>+				4102F69E21273262006AE8D7 /* channel.cc */,
>+				4102F69B21273261006AE8D7 /* channel.h */,
> 				5CDD8AC61E43C23900621E92 /* conversion.h */,
> 				419C83B11FE2472D0040C30F /* null_audio_poller.cc */,
> 				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>";
>@@ -9619,6 +10899,9 @@
> 		5CDD8C451E43C57900621E92 /* call */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41924172212738C200634FCF /* callfactoryinterface.h */,
>+				41924174212738C200634FCF /* transport.cc */,
>+				41924173212738C200634FCF /* transport.h */,
> 				5CDD8C491E43C58E00621E92 /* audio_sink.h */,
> 			);
> 			path = call;
>@@ -9963,6 +11246,8 @@
> 		5CDD8F951E43CCA200621E92 /* pacing */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41A08BD221272EFA001D5D7B /* round_robin_packet_queue.cc */,
>+				41A08BD121272EFA001D5D7B /* round_robin_packet_queue.h */,
> 				5CDD8F981E43CCBE00621E92 /* bitrate_prober.cc */,
> 				5CDD8F991E43CCBE00621E92 /* bitrate_prober.h */,
> 				419C82991FE20CA10040C30F /* interval_budget.cc */,
>@@ -9972,6 +11257,8 @@
> 				419C83281FE245CB0040C30F /* pacer.h */,
> 				419C83261FE245CB0040C30F /* packet_queue.cc */,
> 				419C832A1FE245CC0040C30F /* packet_queue.h */,
>+				419241842127499300634FCF /* packet_queue_interface.cc */,
>+				419241832127499300634FCF /* packet_queue_interface.h */,
> 				5CDD8F9C1E43CCBE00621E92 /* packet_router.cc */,
> 				5CDD8F9D1E43CCBE00621E92 /* packet_router.h */,
> 			);
>@@ -9981,6 +11268,11 @@
> 		5CDD8FA61E43CCE100621E92 /* congestion_controller */ = {
> 			isa = PBXGroup;
> 			children = (
>+				419241B12127591C00634FCF /* bbr */,
>+				41E02CD52127362300C27CD6 /* rtp */,
>+				41E02C8F212734E800C27CD6 /* goog_cc */,
>+				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 */,
>@@ -10044,6 +11336,13 @@
> 		5CDD90341E43CF3700621E92 /* rtc_event_log */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41924178212738FB00634FCF /* rtc_event_log_impl.cc */,
>+				41A08BC221269576001D5D7B /* icelogger.cc */,
>+				41A08BC121269576001D5D7B /* icelogger.h */,
>+				41A08BB921269552001D5D7B /* rtc_event_ice_candidate_pair_config.cc */,
>+				41A08BBB21269552001D5D7B /* rtc_event_ice_candidate_pair_config.h */,
>+				41A08BBC21269553001D5D7B /* rtc_event_ice_candidate_pair.cc */,
>+				41A08BBA21269552001D5D7B /* rtc_event_ice_candidate_pair.h */,
> 				419C83211FE244FE0040C30F /* events */,
> 				419C84201FE24B960040C30F /* output */,
> 				5CDD903E1E43CF6100621E92 /* rtc_event_log.cc */,
>@@ -10415,9 +11714,11 @@
> 			files = (
> 				5CDD8B6D1E43C2B500621E92 /* abs_quant.h in Headers */,
> 				5CDD8B6B1E43C2B500621E92 /* abs_quant_loop.h in Headers */,
>+				41F2636A21267ADF00274F59 /* video_encoder_software_fallback_wrapper.h in Headers */,
> 				5CDD89F91E43BFB300621E92 /* accelerate.h in Headers */,
> 				5CDD85041E43B1EA00621E92 /* acm_codec_database.h in Headers */,
> 				5CDD850A1E43B1EA00621E92 /* acm_receiver.h in Headers */,
>+				41E02CD0212735B700C27CD6 /* bitrate_settings.h in Headers */,
> 				5CDD850C1E43B1EA00621E92 /* acm_resampler.h in Headers */,
> 				5C4B48D81E42C1E3002651C8 /* adaptedvideotracksource.h in Headers */,
> 				5CD285A51E6A63430094FDC8 /* adaptive_fir_filter.h in Headers */,
>@@ -10440,20 +11741,28 @@
> 				5CDD87021E43BA7500621E92 /* agc.h in Headers */,
> 				5CDD87001E43BA7500621E92 /* agc_manager_direct.h in Headers */,
> 				5CDD8FD21E43CD6600621E92 /* aimd_rate_control.h in Headers */,
>+				41D6B45421273159008F9353 /* call_config.h in Headers */,
>+				4192417E2127392300634FCF /* timestamp_extrapolator.h in Headers */,
> 				5C0885281E4A99D200403995 /* alloc.h in Headers */,
> 				41F9BFCC2051DDA200ABF0B9 /* alr_experiment.h in Headers */,
> 				5CDD86ED1E43BA6D00621E92 /* analog_agc.h in Headers */,
> 				5CDD87E61E43BDA100621E92 /* apm_data_dumper.h in Headers */,
> 				5CD284C31E6A5FFC0094FDC8 /* apm_helpers.h in Headers */,
> 				5CDD893C1E43BF3A00621E92 /* app.h in Headers */,
>+				4102F6E221273416006AE8D7 /* color_space.h in Headers */,
> 				5CDD87AA1E43BC0500621E92 /* arith_routines.h in Headers */,
> 				41433D011F79B33400387B4D /* arith_routins.h in Headers */,
> 				41F9BF872051C80100ABF0B9 /* array_view.h in Headers */,
> 				413A23E41FE18E0800373E99 /* arraysize.h in Headers */,
>+				415F1FBF2127304C00064CBF /* echo_cancellation_proxy.h in Headers */,
> 				5C63F8D91E416D53002CA531 /* assertions.h in Headers */,
>+				41E02C8E212734B900C27CD6 /* fec_rate_table.h in Headers */,
>+				41E02CAA2127352D00C27CD6 /* goog_cc_network_control.h in Headers */,
>+				41DDB261212679A300296D47 /* opensslutility.h in Headers */,
> 				413A22A81FE18E0600373E99 /* asyncinvoker-inl.h in Headers */,
> 				413A22B01FE18E0700373E99 /* asyncinvoker.h in Headers */,
> 				413A23A81FE18E0800373E99 /* asyncpacketsocket.h in Headers */,
>+				41A08BAD212681C8001D5D7B /* i010_buffer.h in Headers */,
> 				413A22AB1FE18E0600373E99 /* asyncresolverinterface.h in Headers */,
> 				413A235E1FE18E0700373E99 /* asyncsocket.h in Headers */,
> 				5C63F9E11E4174F6002CA531 /* asyncstuntcpsocket.h in Headers */,
>@@ -10466,7 +11775,10 @@
> 				5CDD84161E439B2900621E92 /* audio_decoder.h in Headers */,
> 				5CD285451E6A61D20094FDC8 /* audio_decoder.h in Headers */,
> 				5CD285431E6A61D20094FDC8 /* audio_decoder_factory.h in Headers */,
>+				41DDB25A2126792800296D47 /* zero_memory.h in Headers */,
> 				41F9BF982051C82100ABF0B9 /* audio_decoder_factory_template.h in Headers */,
>+				41A08BB2212681C8001D5D7B /* video_stream_decoder.h in Headers */,
>+				419241852127499300634FCF /* packet_queue_interface.h in Headers */,
> 				419C83E81FE248350040C30F /* audio_decoder_g711.h in Headers */,
> 				4140B8321E4E3396007409E6 /* audio_decoder_g722.h in Headers */,
> 				413AD1A321265B10003F7263 /* algorithm.h in Headers */,
>@@ -10479,6 +11791,7 @@
> 				5CDD8C0A1E43C34600621E92 /* audio_decoder_isac_t.h in Headers */,
> 				5CDD8C091E43C34600621E92 /* audio_decoder_isac_t_impl.h in Headers */,
> 				419C83F01FE2488D0040C30F /* audio_decoder_L16.h in Headers */,
>+				416D3BE5212731C200775F09 /* gain_curve_applier.h in Headers */,
> 				5CDD8C611E43C60900621E92 /* audio_decoder_opus.h in Headers */,
> 				419C82F21FE20EB50040C30F /* audio_decoder_opus.h in Headers */,
> 				4140B8231E4E3383007409E6 /* audio_decoder_pcm.h in Headers */,
>@@ -10487,8 +11800,11 @@
> 				5C119FFC1E457400004F0987 /* audio_device_buffer.h in Headers */,
> 				5C119FFD1E457400004F0987 /* audio_device_config.h in Headers */,
> 				4145F6141FE1E16F00EB9CAF /* audio_device_data_observer.h in Headers */,
>+				41F2638221267B7700274F59 /* bitrate_constraints.h in Headers */,
>+				4192419B2127586500634FCF /* kiss_fft.h in Headers */,
> 				4145F6121FE1E16F00EB9CAF /* audio_device_defines.h in Headers */,
> 				5C11A01E1E457578004F0987 /* audio_device_dummy.h in Headers */,
>+				41D6B45521273159008F9353 /* degraded_call.h in Headers */,
> 				5C119FFF1E457400004F0987 /* audio_device_generic.h in Headers */,
> 				5CFD537C1E4BA4F500482908 /* audio_device_ios.h in Headers */,
> 				5C11A00C1E457448004F0987 /* audio_device_mac.h in Headers */,
>@@ -10503,6 +11819,7 @@
> 				419C83401FE246230040C30F /* audio_encoder_g722.h in Headers */,
> 				419C833C1FE246230040C30F /* audio_encoder_g722_config.h in Headers */,
> 				5CDD8B711E43C2B500621E92 /* audio_encoder_ilbc.h in Headers */,
>+				41E02CC12127356A00C27CD6 /* data_rate_limiter.h in Headers */,
> 				419C84C01FE319440040C30F /* audio_encoder_ilbc.h in Headers */,
> 				419C84BC1FE319440040C30F /* audio_encoder_ilbc_config.h in Headers */,
> 				419C83661FE246CA0040C30F /* audio_encoder_isac.h in Headers */,
>@@ -10517,6 +11834,7 @@
> 				4140B8211E4E3383007409E6 /* audio_encoder_pcm.h in Headers */,
> 				5CDD8FFF1E43CE3A00621E92 /* audio_encoder_pcm16b.h in Headers */,
> 				5CD285471E6A61D20094FDC8 /* audio_format.h in Headers */,
>+				415F1FCD212730F000064CBF /* audio_mixer.h in Headers */,
> 				5CDD841A1E439B2900621E92 /* audio_format_conversion.h in Headers */,
> 				5CD284931E6A5F410094FDC8 /* audio_frame_manipulator.h in Headers */,
> 				5CD2854E1E6A62130094FDC8 /* audio_frame_operations.h in Headers */,
>@@ -10526,27 +11844,35 @@
> 				5CDD8A021E43BFB300621E92 /* audio_multi_vector.h in Headers */,
> 				5CDD8F7A1E43CBE000621E92 /* audio_network_adaptor_impl.h in Headers */,
> 				5CDD8FEA1E43CDCA00621E92 /* audio_processing.h in Headers */,
>+				419241C32127593F00634FCF /* bbr_network_controller.h in Headers */,
> 				5CDD84AC1E43AF1300621E92 /* audio_processing_impl.h in Headers */,
> 				419C829E1FE20D1C0040C30F /* audio_processing_statistics.h in Headers */,
> 				5CDD8ACD1E43C23900621E92 /* audio_receive_stream.h in Headers */,
>+				41299B972127369100B3414B /* fake_network_pipe.h in Headers */,
> 				5CD284B01E6A5F9F0094FDC8 /* audio_receive_stream.h in Headers */,
>+				4102F6CF212733B7006AE8D7 /* video_quality_observer.h in Headers */,
> 				5C4B4C5D1E431F9C002651C8 /* audio_ring_buffer.h in Headers */,
> 				5CDD8AD01E43C23900621E92 /* audio_send_stream.h in Headers */,
> 				5CD284B21E6A5F9F0094FDC8 /* audio_send_stream.h in Headers */,
> 				5CFD53801E4BA4F500482908 /* audio_session_observer.h in Headers */,
> 				5CDD8C4F1E43C58E00621E92 /* audio_sink.h in Headers */,
>+				41D6B45D21273159008F9353 /* rtp_video_sender.h in Headers */,
> 				5CDD8A921E43C00F00621E92 /* audio_sink.h in Headers */,
> 				5CDD8AD31E43C23900621E92 /* audio_state.h in Headers */,
> 				5CD284B31E6A5F9F0094FDC8 /* audio_state.h in Headers */,
>+				419241472127372800634FCF /* spectral_features_internal.h in Headers */,
> 				41F9BF9C2051C84C00ABF0B9 /* audio_transport_impl.h in Headers */,
> 				41F9BFA02051C88500ABF0B9 /* audio_util.h in Headers */,
> 				5CDD8A051E43BFB300621E92 /* audio_vector.h in Headers */,
> 				5C4B48D91E42C1E3002651C8 /* audiosource.h in Headers */,
>+				415F1FCC212730F000064CBF /* echo_control.h in Headers */,
> 				5CD284F11E6A60570094FDC8 /* audiotrack.h in Headers */,
>+				41F2638E21267F4900274F59 /* g711.h in Headers */,
> 				5CDD8B731E43C2B500621E92 /* augmented_cb_corr.h in Headers */,
> 				5C0885291E4A99D200403995 /* auth.h in Headers */,
> 				5C63F8DA1E416D53002CA531 /* autolink.h in Headers */,
> 				41DDB24E21265BD700296D47 /* inlined_vector.h in Headers */,
>+				41F2639921267F5E00274F59 /* pa_ringbuffer.h in Headers */,
> 				41DDB25121265BE900296D47 /* optional.h in Headers */,
> 				5CDD8A081E43BFB300621E92 /* background_noise.h in Headers */,
> 				5CDD87AF1E43BC0500621E92 /* bandwidth_estimator.h in Headers */,
>@@ -10554,6 +11880,7 @@
> 				5CDD8C0D1E43C34600621E92 /* bandwidth_info.h in Headers */,
> 				5C63F9E31E4174F6002CA531 /* basicpacketsocketfactory.h in Headers */,
> 				5C4B4D0A1E4322F6002651C8 /* basicportallocator.h in Headers */,
>+				41F2639821267F5E00274F59 /* pa_memorybarrier.h in Headers */,
> 				413A235C1FE18E0700373E99 /* bind.h in Headers */,
> 				413A23771FE18E0700373E99 /* bitbuffer.h in Headers */,
> 				41109AB01E5FA19200C0955A /* bitrate_adjuster.h in Headers */,
>@@ -10584,6 +11911,7 @@
> 				4145E4E61EF8CD7D00FCF6E6 /* callfactory.h in Headers */,
> 				41F9BF852051C80100ABF0B9 /* candidate.h in Headers */,
> 				5C63F9E51E4174F6002CA531 /* candidatepairinterface.h in Headers */,
>+				4102F6A021273262006AE8D7 /* transport_feedback_packet_loss_tracker.h in Headers */,
> 				5CD285B31E6A63430094FDC8 /* cascaded_biquad_filter.h in Headers */,
> 				5CDD8B771E43C2B500621E92 /* cb_construct.h in Headers */,
> 				5CDD8B7D1E43C2B500621E92 /* cb_mem_energy.h in Headers */,
>@@ -10597,40 +11925,54 @@
> 				5CDD8B851E43C2B500621E92 /* chebyshev.h in Headers */,
> 				413A234F1FE18E0700373E99 /* checks.h in Headers */,
> 				5C08852A1E4A99D200403995 /* cipher.h in Headers */,
>+				41924175212738C200634FCF /* callfactoryinterface.h in Headers */,
> 				5CDD86A21E43B99400621E92 /* circular_buffer.h in Headers */,
>+				41F2639321267F5300274F59 /* g722_enc_dec.h in Headers */,
> 				5C4B48DC1E42C1E3002651C8 /* codec.h in Headers */,
>+				41E02C7E2127347400C27CD6 /* rtp_video_header.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 */,
>+				4102F6CA212733B7006AE8D7 /* video_stream_encoder.h in Headers */,
>+				4102F68D21273206006AE8D7 /* rtcstatstraversal.h in Headers */,
> 				5CDD8A0E1E43BFB300621E92 /* comfort_noise.h in Headers */,
>+				41299B922127367B00B3414B /* isac_vad.h in Headers */,
> 				5CD285B51E6A63430094FDC8 /* comfort_noise_generator.h in Headers */,
> 				5CDD87271E43BABE00621E92 /* common.h in Headers */,
> 				5CDD84AD1E43AF1300621E92 /* common.h in Headers */,
> 				5CDD8C861E43C66000621E92 /* common.h in Headers */,
> 				5CDD89421E43BF3A00621E92 /* common_header.h in Headers */,
> 				5C6CDD4C1E413598009754E3 /* common_types.h in Headers */,
>+				416D3BE2212731C200775F09 /* down_sampler.h in Headers */,
> 				5CDD8B871E43C2B500621E92 /* comp_corr.h in Headers */,
>+				41F2636121267ADF00274F59 /* builtin_video_encoder_factory.h in Headers */,
> 				413A23851FE18E0700373E99 /* compile_assert_c.h in Headers */,
> 				5CDD86461E43B8B500621E92 /* complex_fft_tables.h in Headers */,
> 				5CDD89451E43BF3A00621E92 /* compound_packet.h in Headers */,
> 				5CDD8FEC1E43CDCA00621E92 /* config.h in Headers */,
> 				5C63F8DB1E416D53002CA531 /* config.h in Headers */,
>+				4192419C2127586500634FCF /* rnn_activations.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 */,
> 				5CDD8F861E43CBE000621E92 /* controller.h in Headers */,
>+				415F1FB621272FBA00064CBF /* vad_with_level.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 */,
>+				41E02CA42127352D00C27CD6 /* probe_controller.h in Headers */,
> 				413A23E21FE18E0800373E99 /* cpu_time.h in Headers */,
> 				5CDD87B21E43BC0500621E92 /* crc.h in Headers */,
>+				4102F6DD21273416006AE8D7 /* video_sink_interface.h in Headers */,
> 				413A22AC1FE18E0600373E99 /* crc32.h in Headers */,
> 				5CDD8B8C1E43C2B500621E92 /* create_augmented_vec.h in Headers */,
>+				419241CD2127593F00634FCF /* data_transfer_tracker.h in Headers */,
> 				413A23691FE18E0700373E99 /* criticalsection.h in Headers */,
> 				5CDD8A101E43BFB300621E92 /* cross_correlation.h in Headers */,
>+				419241A9212758D300634FCF /* symmetric_matrix_buffer.h in Headers */,
> 				5C08852B1E4A99D200403995 /* crypto_kernel.h in Headers */,
> 				5C08852C1E4A99D200403995 /* crypto_types.h in Headers */,
> 				5C4B48DD1E42C1E3002651C8 /* cryptoparams.h in Headers */,
>@@ -10644,7 +11986,9 @@
> 				419C83F81FE248F10040C30F /* decimator.h in Headers */,
> 				5CDD8A171E43BFB300621E92 /* decision_logic.h in Headers */,
> 				5CDD8B901E43C2B500621E92 /* decode.h in Headers */,
>+				41A08BB821268A7D001D5D7B /* memory.h in Headers */,
> 				5CDD8B8E1E43C2B500621E92 /* decode_residual.h in Headers */,
>+				4102F69421273206006AE8D7 /* transportstats.h in Headers */,
> 				419C849D1FE2FEC00040C30F /* decoder.h in Headers */,
> 				5CDD8A1A1E43BFB300621E92 /* decoder_database.h in Headers */,
> 				5CDD8B921E43C2B500621E92 /* decoder_interpolate_lsf.h in Headers */,
>@@ -10652,6 +11996,7 @@
> 				5CD284981E6A5F410094FDC8 /* default_output_rate_calculator.h in Headers */,
> 				419C843B1FE24E800040C30F /* default_temporal_layers.h in Headers */,
> 				5CD2853A1E6A61980094FDC8 /* default_video_bitrate_allocator.h in Headers */,
>+				41F2637F21267B7700274F59 /* fec_controller.h in Headers */,
> 				5CDD85FB1E43B84E00621E92 /* defines.h in Headers */,
> 				5CDD8A1B1E43BFB300621E92 /* defines.h in Headers */,
> 				5CDD8B931E43C2B500621E92 /* defines.h in Headers */,
>@@ -10662,10 +12007,12 @@
> 				5CDD8A211E43BFB300621E92 /* delay_peak_detector.h in Headers */,
> 				413A234E1FE18E0700373E99 /* deprecation.h in Headers */,
> 				5C4B48DE1E42C1E3002651C8 /* device.h in Headers */,
>+				4192418B212749C800634FCF /* send_time_history.h in Headers */,
> 				5CDD86EF1E43BA6D00621E92 /* digital_agc.h in Headers */,
> 				5CDD89481E43BF3A00621E92 /* dlrr.h in Headers */,
> 				5CDD8B951E43C2B500621E92 /* do_plc.h in Headers */,
> 				412455441EF87C0F00F11809 /* dot_product_with_scale.h in Headers */,
>+				41F2636621267ADF00274F59 /* video_decoder_software_fallback_wrapper.h in Headers */,
> 				4145E4D51EF8CC7600FCF6E6 /* downsampled_render_buffer.h in Headers */,
> 				413A23481FE18E0700373E99 /* dscp.h in Headers */,
> 				5CDD8A241E43BFB300621E92 /* dsp_helper.h in Headers */,
>@@ -10683,8 +12030,10 @@
> 				5CDD84AF1E43AF1300621E92 /* echo_cancellation_impl.h in Headers */,
> 				5CD285B91E6A63430094FDC8 /* echo_canceller3.h in Headers */,
> 				5CDD85EB1E43B81000621E92 /* echo_control_mobile.h in Headers */,
>+				41E02CCF212735B700C27CD6 /* network_control.h in Headers */,
> 				5CDD84B11E43AF1300621E92 /* echo_control_mobile_impl.h in Headers */,
> 				5CD285BB1E6A63430094FDC8 /* echo_path_delay_estimator.h in Headers */,
>+				419241632127385B00634FCF /* rw_lock_wrapper.h in Headers */,
> 				5CD285BD1E6A63430094FDC8 /* echo_path_variability.h in Headers */,
> 				5CD285C11E6A63430094FDC8 /* echo_remover.h in Headers */,
> 				5CD285BF1E6A63430094FDC8 /* echo_remover_metrics.h in Headers */,
>@@ -10700,6 +12049,7 @@
> 				5CDD8B991E43C2B500621E92 /* energy_inverse.h in Headers */,
> 				5CDD8B9B1E43C2B500621E92 /* enh_upsample.h in Headers */,
> 				5CDD8B9F1E43C2B500621E92 /* enhancer.h in Headers */,
>+				4102F6C8212733B7006AE8D7 /* video_send_stream_impl.h in Headers */,
> 				5CDD8B9D1E43C2B500621E92 /* enhancer_interface.h in Headers */,
> 				41433D131F79B33400387B4D /* entropy_coding.h in Headers */,
> 				5CDD87B91E43BC0500621E92 /* entropy_coding.h in Headers */,
>@@ -10708,6 +12058,7 @@
> 				5C08852E1E4A99D200403995 /* err.h in Headers */,
> 				413A23A51FE18E0800373E99 /* event.h in Headers */,
> 				5CD286151E6A66130094FDC8 /* event_log_writer.h in Headers */,
>+				419241C22127593F00634FCF /* windowed_filter.h in Headers */,
> 				5C4B4CCF1E4320A9002651C8 /* event_timer_posix.h in Headers */,
> 				413A237E1FE18E0700373E99 /* event_tracer.h in Headers */,
> 				419C83501FE246650040C30F /* exp_filter.h in Headers */,
>@@ -10715,21 +12066,29 @@
> 				5CDD894B1E43BF3A00621E92 /* extended_jitter_report.h in Headers */,
> 				5CDD894E1E43BF3A00621E92 /* extended_reports.h in Headers */,
> 				5CDD8A981E43C00F00621E92 /* fake_decode_from_file.h in Headers */,
>+				41E02CCE212735B700C27CD6 /* network_types.h in Headers */,
>+				41F2638921267F4000274F59 /* fft.h in Headers */,
>+				415F1FB321272FBA00064CBF /* limiter.h in Headers */,
> 				413A237F1FE18E0700373E99 /* fakeclock.h in Headers */,
>+				41924176212738C200634FCF /* transport.h in Headers */,
> 				5C4B48DF1E42C1E3002651C8 /* fakemediaengine.h in Headers */,
> 				413A22B71FE18E0700373E99 /* fakenetwork.h in Headers */,
> 				5C4B48E01E42C1E3002651C8 /* fakenetworkinterface.h in Headers */,
>+				419241D12127597D00634FCF /* simulated_network.h in Headers */,
> 				5C63F9EB1E4174F6002CA531 /* fakeportallocator.h in Headers */,
> 				5C4B48E11E42C1E3002651C8 /* fakertp.h in Headers */,
> 				413A22F81FE18E0700373E99 /* fakesslidentity.h in Headers */,
> 				5C4B48E21E42C1E3002651C8 /* fakevideocapturer.h in Headers */,
> 				5C4B48E31E42C1E3002651C8 /* fakevideorenderer.h in Headers */,
> 				5C63F8DC1E416D53002CA531 /* features.h in Headers */,
>+				41299B8D2127365100B3414B /* transport_feedback_adapter.h in Headers */,
>+				41E02CB32127352D00C27CD6 /* trendline_estimator.h in Headers */,
> 				419C830B1FE20F3C0040C30F /* fec_controller_rplr_based.h in Headers */,
> 				5CDD88791E43BE3C00621E92 /* fec_private_tables_bursty.h in Headers */,
> 				5CDD887A1E43BE3C00621E92 /* fec_private_tables_random.h in Headers */,
> 				5CDD887C1E43BE3C00621E92 /* fec_test_helper.h in Headers */,
> 				5CDD87BB1E43BC0500621E92 /* fft.h in Headers */,
>+				41F2639F2126800000274F59 /* aligned_malloc.h in Headers */,
> 				41433D221F79B33400387B4D /* fft.h in Headers */,
> 				41F9BFD32051DDE500ABF0B9 /* fft_buffer.h in Headers */,
> 				5CD285C81E6A63430094FDC8 /* fft_data.h in Headers */,
>@@ -10742,6 +12101,9 @@
> 				41433CED1F79B33400387B4D /* filterbank_internal.h in Headers */,
> 				41433D0D1F79B33400387B4D /* filterbank_tables.h in Headers */,
> 				5CDD8BA11E43C2B500621E92 /* filtered_cb_vecs.h in Headers */,
>+				419241AE212758D300634FCF /* ring_buffer.h in Headers */,
>+				4192416C2127389B00634FCF /* quality_scaling_experiment.h in Headers */,
>+				419241CC2127593F00634FCF /* bandwidth_sampler.h in Headers */,
> 				5C11A0041E457400004F0987 /* fine_audio_buffer.h in Headers */,
> 				5CDD89511E43BF3A00621E92 /* fir.h in Headers */,
> 				5C4B4C6E1E431F9C002651C8 /* fir_filter.h in Headers */,
>@@ -10755,6 +12117,7 @@
> 				5CD284BA1E6A5F9F0094FDC8 /* flexfec_receive_stream_impl.h in Headers */,
> 				419C83CB1FE247B40040C30F /* flexfec_receiver.h in Headers */,
> 				419C83CE1FE247B40040C30F /* flexfec_sender.h in Headers */,
>+				419241592127376F00634FCF /* fakeicetransport.h in Headers */,
> 				413A22A31FE18E0600373E99 /* format_macros.h in Headers */,
> 				5CDD88891E43BE3C00621E92 /* forward_error_correction.h in Headers */,
> 				5CDD88871E43BE3C00621E92 /* forward_error_correction_internal.h in Headers */,
>@@ -10764,13 +12127,19 @@
> 				5CDD83D21E439A6F00621E92 /* frame_buffer2.h in Headers */,
> 				41109AAF1E5FA19200C0955A /* frame_callback.h in Headers */,
> 				5CDD8BA31E43C2B500621E92 /* frame_classify.h in Headers */,
>+				41A08BAF212681C8001D5D7B /* video_bitrate_allocator.h in Headers */,
> 				5CD2849B1E6A5F410094FDC8 /* frame_combiner.h in Headers */,
>+				419241562127376F00634FCF /* fakecandidatepair.h in Headers */,
> 				5CDD837A1E439A3500621E92 /* frame_dropper.h in Headers */,
>+				419241552127376F00634FCF /* regatheringcontroller.h in Headers */,
> 				5CDD8F911E43CBE000621E92 /* frame_length_controller.h in Headers */,
>+				419241F421275C3200634FCF /* vp8_encoder_simulcast_proxy.h in Headers */,
> 				5CDD83D41E439A6F00621E92 /* frame_object.h in Headers */,
> 				413A22A21FE18E0600373E99 /* function_view.h in Headers */,
> 				4140B8251E4E3383007409E6 /* g711_interface.h in Headers */,
>+				416D3BE1212731C200775F09 /* biquad_filter.h in Headers */,
> 				4140B8391E4E3396007409E6 /* g722_interface.h in Headers */,
>+				41E02CBF2127356A00C27CD6 /* weak_ptr.h in Headers */,
> 				5CDD86F01E43BA6D00621E92 /* gain_control.h in Headers */,
> 				5CDD84B31E43AF1300621E92 /* gain_control_for_experimental_agc.h in Headers */,
> 				5CDD84B51E43AF1300621E92 /* gain_control_impl.h in Headers */,
>@@ -10780,7 +12149,9 @@
> 				5CDD83D61E439A6F00621E92 /* generic_decoder.h in Headers */,
> 				5CDD83D81E439A6F00621E92 /* generic_encoder.h in Headers */,
> 				5CDD8BA91E43C2B500621E92 /* get_cd_vec.h in Headers */,
>+				415F1FB521272FBA00064CBF /* signal_classifier.h in Headers */,
> 				5CDD8BAB1E43C2B500621E92 /* get_lsp_poly.h in Headers */,
>+				4192416D2127389B00634FCF /* congestion_controller_experiment.h in Headers */,
> 				5CDD8BAD1E43C2B500621E92 /* get_sync_seq.h in Headers */,
> 				413A21411FE0F0EF00373E99 /* getopt_s.h in Headers */,
> 				5CDD872A1E43BABE00621E92 /* gmm.h in Headers */,
>@@ -10794,6 +12165,7 @@
> 				4145E4BE1EF894F600FCF6E6 /* h264_profile_level_id.h in Headers */,
> 				5CD285EB1E6A639F0094FDC8 /* h264_sprop_parameter_sets.h in Headers */,
> 				5CDD83DA1E439A6F00621E92 /* h264_sps_pps_tracker.h in Headers */,
>+				4102F6B921273382006AE8D7 /* packet_receiver.h in Headers */,
> 				413A22D81FE18E0700373E99 /* helpers.h in Headers */,
> 				41ECEAD72064147C009D5141 /* helpers.h in Headers */,
> 				5CDD83DC1E439A6F00621E92 /* histogram.h in Headers */,
>@@ -10801,6 +12173,7 @@
> 				5CDD8BAF1E43C2B500621E92 /* hp_input.h in Headers */,
> 				5CDD8BB11E43C2B500621E92 /* hp_output.h in Headers */,
> 				413A23B31FE18E0800373E99 /* httpbase.h in Headers */,
>+				41F2636821267ADF00274F59 /* video_encoder_config.h in Headers */,
> 				413A23B21FE18E0800373E99 /* httpcommon-inl.h in Headers */,
> 				413A23951FE18E0800373E99 /* httpcommon.h in Headers */,
> 				413A22C81FE18E0700373E99 /* httpserver.h in Headers */,
>@@ -10813,6 +12186,7 @@
> 				5CDD8BB41E43C2B500621E92 /* ilbc.h in Headers */,
> 				41109AB11E5FA19200C0955A /* incoming_video_stream.h in Headers */,
> 				5CDD8BB61E43C2B500621E92 /* index_conv_dec.h in Headers */,
>+				41E02C89212734B900C27CD6 /* encoder_database.h in Headers */,
> 				5CDD8BB81E43C2B500621E92 /* index_conv_enc.h in Headers */,
> 				5CDD8BBA1E43C2B500621E92 /* init_decode.h in Headers */,
> 				5CDD8BBC1E43C2B500621E92 /* init_encode.h in Headers */,
>@@ -10828,8 +12202,11 @@
> 				419C83311FE245CD0040C30F /* interval_budget.h in Headers */,
> 				413A22A91FE18E0600373E99 /* ipaddress.h in Headers */,
> 				41433CFF1F79B33400387B4D /* isac_fix_type.h in Headers */,
>+				41DDB266212679A300296D47 /* opensslcertificate.h in Headers */,
>+				419241402127372400634FCF /* lp_residual.h in Headers */,
> 				5CDD87C11E43BC0500621E92 /* isac_float_type.h in Headers */,
> 				5CDD837C1E439A3500621E92 /* ivf_file_writer.h in Headers */,
>+				41A08BB4212681C8001D5D7B /* video_stream_encoder_interface.h in Headers */,
> 				5CDD83E21E439A6F00621E92 /* jitter_buffer.h in Headers */,
> 				5CDD83E01E439A6F00621E92 /* jitter_buffer_common.h in Headers */,
> 				5CDD83E41E439A6F00621E92 /* jitter_estimator.h in Headers */,
>@@ -10855,6 +12232,8 @@
> 				5CDD8BC21E43C2B500621E92 /* lpc_encode.h in Headers */,
> 				5CDD87C81E43BC0500621E92 /* lpc_gain_swb_tables.h in Headers */,
> 				41433D181F79B33400387B4D /* lpc_masking_model.h in Headers */,
>+				41F263B1212680A800274F59 /* asm_defines.h in Headers */,
>+				41E02CA72127352D00C27CD6 /* probe_bitrate_estimator.h in Headers */,
> 				5CDD87CA1E43BC0500621E92 /* lpc_shape_swb12_tables.h in Headers */,
> 				5CDD87CC1E43BC0500621E92 /* lpc_shape_swb16_tables.h in Headers */,
> 				41433D281F79B33400387B4D /* lpc_tables.h in Headers */,
>@@ -10863,6 +12242,7 @@
> 				5CDD8BC61E43C2B500621E92 /* lsf_interpolate_to_poly_dec.h in Headers */,
> 				5CDD8BC81E43C2B500621E92 /* lsf_interpolate_to_poly_enc.h in Headers */,
> 				5CDD8BCA1E43C2B500621E92 /* lsf_to_lsp.h in Headers */,
>+				416D3BDF212731C200775F09 /* agc2_common.h in Headers */,
> 				5CDD8BCC1E43C2B500621E92 /* lsf_to_poly.h in Headers */,
> 				5CDD8BCE1E43C2B500621E92 /* lsp_to_lsf.h in Headers */,
> 				413A23971FE18E0800373E99 /* macutils.h in Headers */,
>@@ -10882,6 +12262,7 @@
> 				5C63F94E1E41737B002CA531 /* mediastreaminterface.h in Headers */,
> 				5CD284FD1E6A60570094FDC8 /* mediastreamobserver.h in Headers */,
> 				5C63F9511E41737B002CA531 /* mediastreamproxy.h in Headers */,
>+				41A08BAC212681C8001D5D7B /* encoded_frame.h in Headers */,
> 				5CD284FE1E6A60570094FDC8 /* mediastreamtrack.h in Headers */,
> 				5C63F9531E41737B002CA531 /* mediastreamtrackproxy.h in Headers */,
> 				5CD286331E6A67BF0094FDC8 /* mediatypes.h in Headers */,
>@@ -10891,9 +12272,11 @@
> 				413A22B91FE18E0700373E99 /* messagehandler.h in Headers */,
> 				413A23A91FE18E0800373E99 /* messagequeue.h in Headers */,
> 				5CDD87071E43BA7500621E92 /* mock_agc.h in Headers */,
>+				4192413C2127372400634FCF /* pitch_search.h in Headers */,
> 				5C11A0051E457400004F0987 /* mock_audio_device_buffer.h in Headers */,
> 				5CDD8FED1E43CDCA00621E92 /* mock_audio_processing.h in Headers */,
> 				419C83561FE246650040C30F /* mod_ops.h in Headers */,
>+				41A08BCF21272EE2001D5D7B /* gain_controller2.h in Headers */,
> 				5CDD837E1E439A3500621E92 /* moving_average.h in Headers */,
> 				5CD286111E6A64C90094FDC8 /* moving_max.h in Headers */,
> 				419C83551FE246650040C30F /* moving_max_counter.h in Headers */,
>@@ -10906,6 +12289,7 @@
> 				5CDD8A331E43BFB300621E92 /* nack_tracker.h in Headers */,
> 				4145E4B71EF8943E00FCF6E6 /* nalu_rewriter.h in Headers */,
> 				413A22C41FE18E0700373E99 /* natserver.h in Headers */,
>+				41E02CAD2127352D00C27CD6 /* alr_detector.h in Headers */,
> 				413A22D91FE18E0700373E99 /* natsocketfactory.h in Headers */,
> 				413A22E71FE18E0700373E99 /* nattypes.h in Headers */,
> 				5CDD8BD21E43C2B500621E92 /* nearest_neighbor.h in Headers */,
>@@ -10919,6 +12303,7 @@
> 				413A23761FE18E0700373E99 /* network_constants.h in Headers */,
> 				413A237C1FE18E0700373E99 /* networkmonitor.h in Headers */,
> 				413A22C21FE18E0700373E99 /* networkroute.h in Headers */,
>+				419241C02127593F00634FCF /* loss_rate_filter.h in Headers */,
> 				5CDD872B1E43BABE00621E92 /* noise_gmm_tables.h in Headers */,
> 				5CDD85FF1E43B84E00621E92 /* noise_suppression.h in Headers */,
> 				5CDD84EA1E43B0B600621E92 /* noise_suppression_impl.h in Headers */,
>@@ -10928,7 +12313,9 @@
> 				5C63F9541E41737B002CA531 /* notifier.h in Headers */,
> 				5CDD86011E43B84E00621E92 /* ns_core.h in Headers */,
> 				41ECEAC120640F28009D5141 /* NSString+StdString.h in Headers */,
>+				41F263B82126810C00274F59 /* base64.h in Headers */,
> 				5CDD86061E43B84E00621E92 /* nsx_core.h in Headers */,
>+				41A08BCB2126961F001D5D7B /* raw_logging.h in Headers */,
> 				5CDD86071E43B84E00621E92 /* nsx_defines.h in Headers */,
> 				419C83B41FE2472E0040C30F /* null_audio_poller.h in Headers */,
> 				5C0885311E4A99D200403995 /* null_auth.h in Headers */,
>@@ -10950,12 +12337,14 @@
> 				5CDD85A51E43B5C000621E92 /* overuse_frame_detector.h in Headers */,
> 				5C63F9EE1E4174F6002CA531 /* p2pconstants.h in Headers */,
> 				5C63F9F31E4174F6002CA531 /* p2ptransportchannel.h in Headers */,
>+				415F1FB721272FBA00064CBF /* vector_float_frame.h in Headers */,
> 				5CDD8FA31E43CCBE00621E92 /* paced_sender.h in Headers */,
> 				419C832E1FE245CD0040C30F /* pacer.h in Headers */,
> 				5CDD8BD41E43C2B500621E92 /* pack_bits.h in Headers */,
> 				5CDD83EF1E439A6F00621E92 /* packet.h in Headers */,
> 				5CDD8AB01E43C00F00621E92 /* packet.h in Headers */,
> 				5CDD8A431E43BFB300621E92 /* packet.h in Headers */,
>+				415F1FAE21272FBA00064CBF /* fixed_digital_level_estimator.h in Headers */,
> 				5CDD8A411E43BFB300621E92 /* packet_buffer.h in Headers */,
> 				5CDD83ED1E439A6F00621E92 /* packet_buffer.h in Headers */,
> 				5CDD888D1E43BE3C00621E92 /* packet_loss_stats.h in Headers */,
>@@ -10968,6 +12357,7 @@
> 				419C83D91FE247F20040C30F /* packettransportinternal.h in Headers */,
> 				413A23901FE18E0800373E99 /* pathutils.h in Headers */,
> 				5C4B48841E42C1BA002651C8 /* payload_type_mapper.h in Headers */,
>+				41D6B45B21273159008F9353 /* rtp_video_sender_interface.h in Headers */,
> 				5CDD90011E43CE3A00621E92 /* pcm16b.h in Headers */,
> 				5CD285001E6A60570094FDC8 /* peerconnection.h in Headers */,
> 				5CD284801E6A5E740094FDC8 /* peerconnectionfactory.h in Headers */,
>@@ -10978,7 +12368,9 @@
> 				419C834E1FE246650040C30F /* percentile_filter.h in Headers */,
> 				413A23EA1FE18E0800373E99 /* physicalsocketserver.h in Headers */,
> 				5CDD872E1E43BABE00621E92 /* pitch_based_vad.h in Headers */,
>+				415F1FB021272FBA00064CBF /* gain_applier.h in Headers */,
> 				41433CF11F79B33400387B4D /* pitch_estimator.h in Headers */,
>+				419241ED21275AFA00634FCF /* simulcast_utility.h in Headers */,
> 				5CDD87D11E43BC0500621E92 /* pitch_estimator.h in Headers */,
> 				5CDD87D41E43BC0600621E92 /* pitch_gain_tables.h in Headers */,
> 				41433D231F79B33400387B4D /* pitch_gain_tables.h in Headers */,
>@@ -10994,10 +12386,14 @@
> 				5CDD8BD61E43C2B500621E92 /* poly_to_lsf.h in Headers */,
> 				5CDD8BD81E43C2B500621E92 /* poly_to_lsp.h in Headers */,
> 				5C63F9F81E4174F6002CA531 /* port.h in Headers */,
>+				415F1FBD2127304C00064CBF /* echo_control_mobile_proxy.h in Headers */,
>+				419241C62127593F00634FCF /* bbr_factory.h in Headers */,
> 				5C63F9FB1E4174F6002CA531 /* portallocator.h in Headers */,
> 				5C63F9FC1E4174F6002CA531 /* portinterface.h in Headers */,
>+				4102F6E421273416006AE8D7 /* video_bitrate_allocator_factory.h in Headers */,
> 				5CDD8A461E43BFB300621E92 /* post_decode_vad.h in Headers */,
> 				5CDD835A1E43257200621E92 /* pps_parser.h in Headers */,
>+				415F1FB421272FBA00064CBF /* noise_spectrum_estimator.h in Headers */,
> 				5CDD8A481E43BFB300621E92 /* preemptive_expand.h in Headers */,
> 				5CDD84E61E43B04D00621E92 /* process_thread_impl.h in Headers */,
> 				5CDD835D1E43257200621E92 /* profile_level_id.h in Headers */,
>@@ -11006,23 +12402,29 @@
> 				413A231F1FE18E0700373E99 /* proxyinfo.h in Headers */,
> 				413A23C51FE18E0800373E99 /* proxyserver.h in Headers */,
> 				5C63F9FF1E4174F6002CA531 /* pseudotcp.h in Headers */,
>+				41D6B45A21273159008F9353 /* rtp_payload_params.h in Headers */,
> 				5CDD89591E43BF3A00621E92 /* psfb.h in Headers */,
> 				5CDD87561E43BAF500621E92 /* push_sinc_resampler.h in Headers */,
> 				4130920A1EF8D11200757C55 /* qp_parser.h in Headers */,
>+				4102F68E21273206006AE8D7 /* jseptransportcontroller.h in Headers */,
> 				5CDD83821E439A3500621E92 /* quality_scaler.h in Headers */,
> 				5CD2862B1E6A66C80094FDC8 /* quality_threshold.h in Headers */,
> 				413A23EF1FE18E0800373E99 /* race_checker.h in Headers */,
> 				413A234C1FE18E0700373E99 /* random.h in Headers */,
>+				41E02C7B2127347400C27CD6 /* rtp_generic_frame_descriptor.h in Headers */,
> 				5CDD8A4B1E43BFB300621E92 /* random_vector.h in Headers */,
> 				5CDD895C1E43BF3A00621E92 /* rapid_resync_request.h in Headers */,
>+				41E02C8C212734B900C27CD6 /* fec_controller_default.h in Headers */,
> 				413A23601FE18E0700373E99 /* rate_limiter.h in Headers */,
> 				413A230D1FE18E0700373E99 /* rate_statistics.h in Headers */,
> 				413A23041FE18E0700373E99 /* ratetracker.h in Headers */,
> 				5C0885331E4A99D200403995 /* rdb.h in Headers */,
> 				5C0885341E4A99D200403995 /* rdbx.h in Headers */,
>+				4102F69521273206006AE8D7 /* peerconnectioninternal.h in Headers */,
> 				5C63F8DF1E416D53002CA531 /* reader.h in Headers */,
> 				5C4B4C781E431F9C002651C8 /* real_fourier.h in Headers */,
> 				5C4B4C731E431F9C002651C8 /* real_fourier_ooura.h in Headers */,
>+				419241C82127593F00634FCF /* packet_number_indexed_queue.h in Headers */,
> 				419C83CA1FE247B40040C30F /* receive_statistics.h in Headers */,
> 				5CDD88921E43BE3C00621E92 /* receive_statistics_impl.h in Headers */,
> 				5CDD85AB1E43B5C000621E92 /* receive_statistics_proxy.h in Headers */,
>@@ -11034,21 +12436,29 @@
> 				413A23671FE18E0700373E99 /* refcountedobject.h in Headers */,
> 				413A23201FE18E0700373E99 /* refcounter.h in Headers */,
> 				5CDD8BDA1E43C2B500621E92 /* refiner.h in Headers */,
>+				41F263AF212680A800274F59 /* inline.h in Headers */,
> 				5C63FA021E4174F6002CA531 /* relayport.h in Headers */,
> 				5C63FA051E4174F6002CA531 /* relayserver.h in Headers */,
>+				4102F69121273206006AE8D7 /* rtpparametersconversion.h in Headers */,
> 				5CDD89621E43BF3A00621E92 /* remb.h in Headers */,
>+				4192419F2127588300634FCF /* audio_codec_pair_id.h in Headers */,
> 				5CDD8FDC1E43CD6600621E92 /* remote_bitrate_estimator_abs_send_time.h in Headers */,
> 				5CDD8FDE1E43CD6600621E92 /* remote_bitrate_estimator_single_stream.h in Headers */,
>+				41F2636F21267B3E00274F59 /* fft4g.h in Headers */,
> 				5CDD8FE21E43CD6600621E92 /* remote_estimator_proxy.h in Headers */,
> 				419C83C51FE247B40040C30F /* remote_ntp_time_estimator.h in Headers */,
> 				5CD285061E6A60570094FDC8 /* remoteaudiosource.h in Headers */,
> 				4145E4DA1EF8CC9B00FCF6E6 /* render_buffer.h in Headers */,
> 				5CD285D61E6A63430094FDC8 /* render_delay_buffer.h in Headers */,
>+				4192416F2127389B00634FCF /* field_trial_parser.h in Headers */,
> 				5CD285DA1E6A63430094FDC8 /* render_delay_controller.h in Headers */,
> 				5CD285D81E6A63430094FDC8 /* render_delay_controller_metrics.h in Headers */,
>+				4192415B2127376F00634FCF /* basicasyncresolverfactory.h in Headers */,
>+				41A08BC7212695DE001D5D7B /* bad_optional_access.h in Headers */,
> 				5CDD84BC1E43AF1300621E92 /* render_queue_item_verifier.h in Headers */,
> 				5CD285DC1E6A63430094FDC8 /* render_signal_analyzer.h in Headers */,
> 				5CDD85191E43B1EA00621E92 /* rent_a_codec.h in Headers */,
>+				41F263BC2126811500274F59 /* sigslot.h in Headers */,
> 				5CDD89651E43BF3A00621E92 /* report_block.h in Headers */,
> 				5CDD85AF1E43B5C000621E92 /* report_block_stats.h in Headers */,
> 				5CDD86651E43B8B500621E92 /* resample_by_2_internal.h in Headers */,
>@@ -11057,20 +12467,26 @@
> 				5CD285DE1E6A63430094FDC8 /* residual_echo_estimator.h in Headers */,
> 				5C4B4C7B1E431F9C002651C8 /* ring_buffer.h in Headers */,
> 				5CDD84C01E43AF1300621E92 /* rms_level.h in Headers */,
>+				41A08BAE212681C8001D5D7B /* video_bitrate_allocation.h in Headers */,
> 				413A23A71FE18E0800373E99 /* rollingaccumulator.h in Headers */,
> 				5CDD896B1E43BF3A00621E92 /* rrtr.h in Headers */,
>+				4102F6A521273262006AE8D7 /* channel.h in Headers */,
> 				419C83AC1FE247080040C30F /* rtc_event.h in Headers */,
> 				41F9BFA92051C8E200ABF0B9 /* rtc_event_alr_state.h in Headers */,
> 				419C839D1FE246F90040C30F /* rtc_event_audio_network_adaptation.h in Headers */,
>+				41F2638121267B7700274F59 /* audio_options.h in Headers */,
> 				419C83981FE246F90040C30F /* rtc_event_audio_playout.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 */,
> 				419C83971FE246F90040C30F /* rtc_event_bwe_update_loss_based.h in Headers */,
> 				5CDD904C1E43CF6100621E92 /* rtc_event_log.h in Headers */,
>+				4102F6A721273262006AE8D7 /* remix_resample.h in Headers */,
>+				41E02C87212734B900C27CD6 /* decoder_database.h in Headers */,
> 				41F411BB1EF8DA9600343C26 /* rtc_event_log_factory.h in Headers */,
> 				41F411BA1EF8DA9600343C26 /* rtc_event_log_factory_interface.h in Headers */,
> 				419C84231FE24BA60040C30F /* rtc_event_log_output_file.h in Headers */,
>+				415F1FCF212730F000064CBF /* audio_frame.h in Headers */,
> 				419C839F1FE246F90040C30F /* rtc_event_probe_cluster_created.h in Headers */,
> 				419C83A01FE246F90040C30F /* rtc_event_probe_result_failure.h in Headers */,
> 				419C839A1FE246F90040C30F /* rtc_event_probe_result_success.h in Headers */,
>@@ -11109,21 +12525,32 @@
> 				413A24491FE1991A00373E99 /* RTCMacros.h in Headers */,
> 				413A24681FE1991A00373E99 /* RTCMediaConstraints.h in Headers */,
> 				413A24511FE1991A00373E99 /* RTCMediaSource.h in Headers */,
>+				419241452127372800634FCF /* pitch_search_internal.h in Headers */,
> 				413A24541FE1991A00373E99 /* RTCMediaStream.h in Headers */,
>+				415F1FA921272FBA00064CBF /* adaptive_mode_level_estimator.h in Headers */,
> 				413A24471FE1991A00373E99 /* RTCMediaStreamTrack.h in Headers */,
> 				413A24641FE1991A00373E99 /* RTCMetrics.h in Headers */,
> 				413A246A1FE1991A00373E99 /* RTCMetricsSampleInfo.h in Headers */,
> 				413A24671FE1991A00373E99 /* RTCMTLNSVideoView.h in Headers */,
> 				413A24711FE1991A00373E99 /* RTCMTLVideoView.h in Headers */,
>+				4102F6A321273262006AE8D7 /* channel_proxy.h in Headers */,
> 				413A24591FE1991A00373E99 /* RTCNSGLVideoView.h in Headers */,
>+				41E02C792127347400C27CD6 /* rtp_generic_frame_descriptor_extension.h in Headers */,
>+				41F263B2212680A800274F59 /* file_wrapper.h in Headers */,
> 				5CDD8A501E43BFB300621E92 /* rtcp.h in Headers */,
> 				419C82B21FE20DCD0040C30F /* rtcp_demuxer.h in Headers */,
>+				41F2637221267B4A00274F59 /* spl_sqrt_floor.h in Headers */,
> 				5CD285FB1E6A64520094FDC8 /* rtcp_nack_stats.h in Headers */,
> 				5CDD88981E43BE3C00621E92 /* rtcp_packet.h in Headers */,
> 				419C82BD1FE20DCD0040C30F /* rtcp_packet_sink_interface.h in Headers */,
> 				5CDD889B1E43BE3C00621E92 /* rtcp_receiver.h in Headers */,
>+				41F263A02126800000274F59 /* aligned_array.h in Headers */,
>+				4192418D212749C800634FCF /* pacer_controller.h in Headers */,
> 				5CDD889E1E43BE3C00621E92 /* rtcp_sender.h in Headers */,
>+				416D3BEA212731C200775F09 /* saturation_protector.h in Headers */,
>+				41DDB26C212679D200296D47 /* string_builder.h in Headers */,
> 				419C83031FE20F020040C30F /* rtcp_transceiver.h in Headers */,
>+				41A08BD321272EFA001D5D7B /* round_robin_packet_queue.h in Headers */,
> 				419C83011FE20F020040C30F /* rtcp_transceiver_config.h in Headers */,
> 				419C83051FE20F020040C30F /* rtcp_transceiver_impl.h in Headers */,
> 				413A243E1FE1991A00373E99 /* RTCPeerConnection.h in Headers */,
>@@ -11135,8 +12562,11 @@
> 				413A245C1FE1991A00373E99 /* RTCRtpSender.h in Headers */,
> 				413A244A1FE1991A00373E99 /* RTCSessionDescription.h in Headers */,
> 				413A24661FE1991A00373E99 /* RTCSSLAdapter.h in Headers */,
>+				41E02CAF2127352D00C27CD6 /* median_slope_estimator.h in Headers */,
> 				4145E4C81EF896D700FCF6E6 /* rtcstats.h in Headers */,
> 				4145E4C71EF896D700FCF6E6 /* rtcstats_objects.h in Headers */,
>+				419241E721275A7600634FCF /* field_trial_units.h in Headers */,
>+				41A08BC321269577001D5D7B /* icelogger.h in Headers */,
> 				5CD285081E6A60570094FDC8 /* rtcstatscollector.h in Headers */,
> 				4145E4C51EF896D700FCF6E6 /* rtcstatscollectorcallback.h in Headers */,
> 				4145E4C61EF896D700FCF6E6 /* rtcstatsreport.h in Headers */,
>@@ -11151,18 +12581,23 @@
> 				413A24601FE1991A00373E99 /* RTCVideoDecoderVP9.h in Headers */,
> 				413A246E1FE1991A00373E99 /* RTCVideoEncoderVP8.h in Headers */,
> 				413A24461FE1991A00373E99 /* RTCVideoEncoderVP9.h in Headers */,
>+				41E02CA62127352D00C27CD6 /* acknowledged_bitrate_estimator.h in Headers */,
> 				413A244F1FE1991A00373E99 /* RTCVideoFrame.h in Headers */,
> 				413A24411FE1991A00373E99 /* RTCVideoFrameBuffer.h in Headers */,
> 				413A246F1FE1991A00373E99 /* RTCVideoRenderer.h in Headers */,
>+				41D6B45821273159008F9353 /* receive_time_calculator.h in Headers */,
> 				413A24611FE1991A00373E99 /* RTCVideoSource.h in Headers */,
> 				413A24501FE1991A00373E99 /* RTCVideoTrack.h in Headers */,
> 				413A243A1FE1991A00373E99 /* RTCVideoViewShading.h in Headers */,
> 				41ECEAE3206416FF009D5141 /* RTCWrappedNativeVideoDecoder.h in Headers */,
> 				41ECEAE1206416FF009D5141 /* RTCWrappedNativeVideoEncoder.h in Headers */,
>+				419241982127586500634FCF /* rnn_vad_weights.h in Headers */,
> 				419C82B81FE20DCD0040C30F /* rtp_config.h in Headers */,
> 				419C83C91FE247B40040C30F /* rtp_cvo.h in Headers */,
> 				413092011EF8D0A600757C55 /* rtp_demuxer.h in Headers */,
> 				5CDD8AB71E43C00F00621E92 /* rtp_file_source.h in Headers */,
>+				41DDB26D212679D200296D47 /* audio_format_to_string.h in Headers */,
>+				41F263AD212680A800274F59 /* ignore_warnings.h in Headers */,
> 				5CDD88B11E43BE3C00621E92 /* rtp_format.h in Headers */,
> 				5CDD88A51E43BE3C00621E92 /* rtp_format_h264.h in Headers */,
> 				5CDD88A71E43BE3C00621E92 /* rtp_format_video_generic.h in Headers */,
>@@ -11175,7 +12610,9 @@
> 				419C83CC1FE247B40040C30F /* rtp_header_parser.h in Headers */,
> 				41F9BF8D2051C80100ABF0B9 /* rtp_headers.h in Headers */,
> 				5CDD88BF1E43BE3D00621E92 /* rtp_packet.h in Headers */,
>+				419241422127372400634FCF /* fft_util.h in Headers */,
> 				5CDD88BA1E43BE3D00621E92 /* rtp_packet_history.h in Headers */,
>+				41F263B0212680A800274F59 /* unused.h in Headers */,
> 				5CDD88BB1E43BE3D00621E92 /* rtp_packet_received.h in Headers */,
> 				419C82B61FE20DCD0040C30F /* rtp_packet_sink_interface.h in Headers */,
> 				5CDD88BC1E43BE3D00621E92 /* rtp_packet_to_send.h in Headers */,
>@@ -11183,9 +12620,12 @@
> 				419C83C71FE247B40040C30F /* rtp_receiver.h in Headers */,
> 				5CDD88C31E43BE3D00621E92 /* rtp_receiver_audio.h in Headers */,
> 				5CDD88C51E43BE3D00621E92 /* rtp_receiver_impl.h in Headers */,
>+				41A08BD721272F1D001D5D7B /* congestion_window_pushback_controller.h in Headers */,
> 				5CDD88C71E43BE3D00621E92 /* rtp_receiver_strategy.h in Headers */,
> 				5CDD88C91E43BE3D00621E92 /* rtp_receiver_video.h in Headers */,
> 				419C83C31FE247B40040C30F /* rtp_rtcp.h in Headers */,
>+				4102F6A221273262006AE8D7 /* audio_level.h in Headers */,
>+				4102F6DE21273416006AE8D7 /* video_stream_decoder_create.h in Headers */,
> 				5CDD88CA1E43BE3D00621E92 /* rtp_rtcp_config.h in Headers */,
> 				419C83C81FE247B40040C30F /* rtp_rtcp_defines.h in Headers */,
> 				419C82B51FE20DCD0040C30F /* rtp_rtcp_demuxer_helper.h in Headers */,
>@@ -11204,12 +12644,14 @@
> 				5CDD896D1E43BF3A00621E92 /* rtpfb.h in Headers */,
> 				419C82CA1FE20E2C0040C30F /* rtpmediautils.h in Headers */,
> 				5C63F9661E41737B002CA531 /* rtpparameters.h in Headers */,
>+				41F2636421267ADF00274F59 /* video_codec.h in Headers */,
> 				5CD2850A1E6A60570094FDC8 /* rtpreceiver.h in Headers */,
> 				5C63F9691E41737B002CA531 /* rtpreceiverinterface.h in Headers */,
> 				5CD2850C1E6A60570094FDC8 /* rtpsender.h in Headers */,
> 				5C63F96C1E41737B002CA531 /* rtpsenderinterface.h in Headers */,
> 				419C82DA1FE20E590040C30F /* rtptransceiver.h in Headers */,
> 				41F9BF8E2051C80100ABF0B9 /* rtptransceiverinterface.h in Headers */,
>+				41DDB263212679A300296D47 /* opensslsessioncache.h in Headers */,
> 				4145E4CE1EF8CB9500FCF6E6 /* rtptransport.h in Headers */,
> 				419C82E51FE20E590040C30F /* rtptransportinternal.h in Headers */,
> 				419C82DB1FE20E590040C30F /* rtptransportinternaladapter.h in Headers */,
>@@ -11222,11 +12664,13 @@
> 				413A23281FE18E0700373E99 /* sanitizer.h in Headers */,
> 				413A23051FE18E0700373E99 /* scoped_ref_ptr.h in Headers */,
> 				41F9BFB62051C93600ABF0B9 /* scopedvideodecoder.h in Headers */,
>+				41A08BB1212681C8001D5D7B /* video_source_interface.h in Headers */,
> 				41F9BFB72051C93600ABF0B9 /* scopedvideoencoder.h in Headers */,
> 				419C84361FE24E800040C30F /* screenshare_layers.h in Headers */,
> 				5CD2863D1E6A681C0094FDC8 /* sctptransport.h in Headers */,
> 				5CD2863E1E6A681C0094FDC8 /* sctptransportinternal.h in Headers */,
> 				5CD2850E1E6A60570094FDC8 /* sctputils.h in Headers */,
>+				41E02CD32127360700C27CD6 /* sample_counter.h in Headers */,
> 				5CDD89701E43BF3A00621E92 /* sdes.h in Headers */,
> 				4145F61D1FE1F38D00EB9CAF /* sdp_video_format.h in Headers */,
> 				419C82DD1FE20E590040C30F /* sdputils.h in Headers */,
>@@ -11239,10 +12683,13 @@
> 				5CDD83FC1E439A6F00621E92 /* session_info.h in Headers */,
> 				41F9BFDF2051DE1800ABF0B9 /* sessiondescription.h in Headers */,
> 				41F9BF922051C80100ABF0B9 /* setremotedescriptionobserverinterface.h in Headers */,
>+				41A08BBF21269553001D5D7B /* rtc_event_ice_candidate_pair_config.h in Headers */,
> 				41433D081F79B33400387B4D /* settings.h in Headers */,
> 				5CDD87D71E43BC0600621E92 /* settings.h in Headers */,
> 				5CD285E01E6A63430094FDC8 /* shadow_filter_update_gain.h in Headers */,
> 				41F9BFC52051CAE500ABF0B9 /* signal_processing_library.h in Headers */,
>+				415F1FAB21272FBA00064CBF /* compute_interpolated_gain_curve.h in Headers */,
>+				419241F021275AFA00634FCF /* simulcast_rate_allocator.h in Headers */,
> 				413A23811FE18E0700373E99 /* signalthread.h in Headers */,
> 				413A22CD1FE18E0700373E99 /* sigslot.h in Headers */,
> 				413A23331FE18E0700373E99 /* sigslotrepeater.h in Headers */,
>@@ -11257,7 +12704,9 @@
> 				5CDD8BE61E43C2B500621E92 /* smooth.h in Headers */,
> 				5CDD8BE41E43C2B500621E92 /* smooth_out_data.h in Headers */,
> 				5CD285F71E6A63F60094FDC8 /* smoothing_filter.h in Headers */,
>+				419241582127376F00634FCF /* fakepackettransport.h in Headers */,
> 				413A22DD1FE18E0700373E99 /* socket.h in Headers */,
>+				415F1FA721272FBA00064CBF /* adaptive_digital_gain_applier.h in Headers */,
> 				413A22DC1FE18E0700373E99 /* socket_unittest.h in Headers */,
> 				413A23611FE18E0700373E99 /* socketadapters.h in Headers */,
> 				413A23C91FE18E0800373E99 /* socketaddress.h in Headers */,
>@@ -11265,6 +12714,9 @@
> 				413A22C51FE18E0700373E99 /* socketfactory.h in Headers */,
> 				413A23B91FE18E0800373E99 /* socketserver.h in Headers */,
> 				413A23531FE18E0700373E99 /* socketstream.h in Headers */,
>+				419241DE21275A3000634FCF /* data_rate.h in Headers */,
>+				4192413B2127372400634FCF /* features_extraction.h in Headers */,
>+				416D3BDC212731C200775F09 /* adaptive_agc.h in Headers */,
> 				5CDD8BE81E43C2B500621E92 /* sort_sq.h in Headers */,
> 				5C4B4C7E1E431F9C002651C8 /* sparse_fir_filter.h in Headers */,
> 				5CDD87D91E43BC0600621E92 /* spectrum_ar_model_tables.h in Headers */,
>@@ -11279,7 +12731,9 @@
> 				419C82E21FE20E590040C30F /* srtptransport.h in Headers */,
> 				413A23AF1FE18E0800373E99 /* sslfingerprint.h in Headers */,
> 				413A22E41FE18E0700373E99 /* sslidentity.h in Headers */,
>+				419241822127497100634FCF /* bad_variant_access.h in Headers */,
> 				413A23921FE18E0800373E99 /* sslroots.h in Headers */,
>+				416D3BDD212731C200775F09 /* adaptive_mode_level_estimator_agc.h in Headers */,
> 				413A23C11FE18E0800373E99 /* sslstreamadapter.h in Headers */,
> 				41F9BF9E2051C86900ABF0B9 /* ssrc_binding_observer.h in Headers */,
> 				5CDD87371E43BABE00621E92 /* standalone_vad.h in Headers */,
>@@ -11292,14 +12746,17 @@
> 				5C63F9731E41737B002CA531 /* statstypes.h in Headers */,
> 				413A22FD1FE18E0700373E99 /* stream.h in Headers */,
> 				5CDD85C01E43B5C000621E92 /* stream_synchronization.h in Headers */,
>+				4102F6CE212733B7006AE8D7 /* video_stream_decoder_impl.h in Headers */,
> 				5CD285111E6A60570094FDC8 /* streamcollection.h in Headers */,
> 				5C4B48F51E42C1E3002651C8 /* streamparams.h in Headers */,
> 				413A23641FE18E0700373E99 /* string_to_number.h in Headers */,
>+				419241CB2127593F00634FCF /* rtt_stats.h in Headers */,
> 				413A22EB1FE18E0700373E99 /* stringencode.h in Headers */,
> 				413A23B11FE18E0800373E99 /* stringize_macros.h in Headers */,
> 				413A22CE1FE18E0700373E99 /* stringutils.h in Headers */,
> 				41433CF01F79B33400387B4D /* structs.h in Headers */,
> 				5CDD87DA1E43BC0600621E92 /* structs.h in Headers */,
>+				419241AA212758D300634FCF /* rnn.h in Headers */,
> 				5C63FA0C1E4174F6002CA531 /* stun.h in Headers */,
> 				5C63FA0F1E4174F6002CA531 /* stunport.h in Headers */,
> 				5C63FA121E4174F6002CA531 /* stunrequest.h in Headers */,
>@@ -11311,6 +12768,7 @@
> 				5CD285E71E6A63430094FDC8 /* suppression_gain.h in Headers */,
> 				5CDD8BF01E43C2B500621E92 /* swap_bytes.h in Headers */,
> 				413A23471FE18E0700373E99 /* swap_queue.h in Headers */,
>+				4102F6AF212732E7006AE8D7 /* expand_uma_logger.h in Headers */,
> 				5CDD8A551E43BFB300621E92 /* sync_buffer.h in Headers */,
> 				5CD284BF1E6A5F9F0094FDC8 /* syncable.h in Headers */,
> 				5CDD89791E43BF3A00621E92 /* target_bitrate.h in Headers */,
>@@ -11319,20 +12777,28 @@
> 				5C63FA181E4174F6002CA531 /* tcpport.h in Headers */,
> 				413A22FE1FE18E0700373E99 /* template_util.h in Headers */,
> 				419C843D1FE24E800040C30F /* temporal_layers.h in Headers */,
>+				419241E421275A3000634FCF /* data_size.h in Headers */,
> 				413A22A11FE18E0600373E99 /* testbase64.h in Headers */,
> 				413A23D51FE18E0800373E99 /* testclient.h in Headers */,
> 				413A232A1FE18E0700373E99 /* testechoserver.h in Headers */,
> 				5C63FA191E4174F6002CA531 /* testrelayserver.h in Headers */,
>+				41299B932127367B00B3414B /* pitch_filter.h in Headers */,
> 				5C63FA1A1E4174F6002CA531 /* teststunserver.h in Headers */,
> 				5C63FA1B1E4174F6002CA531 /* testturnserver.h in Headers */,
>+				41E02CBD2127356A00C27CD6 /* sslcertificate.h in Headers */,
> 				413A23D91FE18E0800373E99 /* testutils.h in Headers */,
> 				413A23421FE18E0700373E99 /* thread.h in Headers */,
> 				413A22C61FE18E0700373E99 /* thread_annotations.h in Headers */,
>+				41E02CAB2127352D00C27CD6 /* delay_based_bwe.h in Headers */,
>+				41E02CB72127352D00C27CD6 /* delay_increase_detector_interface.h in Headers */,
> 				413A22E81FE18E0700373E99 /* thread_checker.h in Headers */,
>+				4192415A2127376F00634FCF /* icetransportinternal.h in Headers */,
> 				413A22B81FE18E0700373E99 /* thread_checker_impl.h in Headers */,
> 				5CDD84C41E43AF1300621E92 /* three_band_filter_bank.h in Headers */,
> 				5CDD8A581E43BFB300621E92 /* tick_timer.h in Headers */,
> 				419C83341FE245EA0040C30F /* time_interval.h in Headers */,
>+				41F2636021267ADF00274F59 /* builtin_video_decoder_factory.h in Headers */,
>+				419241432127372800634FCF /* pitch_info.h in Headers */,
> 				5CDD8A5B1E43BFB300621E92 /* time_stretch.h in Headers */,
> 				5CDD88DB1E43BE3D00621E92 /* time_util.h in Headers */,
> 				5CDD83FE1E439A6F00621E92 /* timestamp_map.h in Headers */,
>@@ -11343,6 +12809,7 @@
> 				5CDD897B1E43BF3A00621E92 /* tmmb_item.h in Headers */,
> 				5CDD897E1E43BF3A00621E92 /* tmmbn.h in Headers */,
> 				5CDD89811E43BF3A00621E92 /* tmmbr.h in Headers */,
>+				419241AB212758D300634FCF /* spectral_features.h in Headers */,
> 				5CDD88DD1E43BE3D00621E92 /* tmmbr_help.h in Headers */,
> 				413A23001FE18E0700373E99 /* trace_event.h in Headers */,
> 				5CD285131E6A60570094FDC8 /* trackmediainfomap.h in Headers */,
>@@ -11351,7 +12818,9 @@
> 				5C63FA1E1E4174F6002CA531 /* transport.h in Headers */,
> 				5CDD85C21E43B5C000621E92 /* transport_adapter.h in Headers */,
> 				5CDD89841E43BF3A00621E92 /* transport_feedback.h in Headers */,
>+				4192413F2127372400634FCF /* common.h in Headers */,
> 				5CDD8FBC1E43CD0700621E92 /* transport_feedback_adapter.h in Headers */,
>+				4102F6B221273331006AE8D7 /* mediaconfig.h in Headers */,
> 				5C63FA261E4174F6002CA531 /* transportdescription.h in Headers */,
> 				5C63FA291E4174F6002CA531 /* transportdescriptionfactory.h in Headers */,
> 				5C63FA2A1E4174F6002CA531 /* transportinfo.h in Headers */,
>@@ -11365,6 +12834,7 @@
> 				5C63FA311E4174F6002CA531 /* udpport.h in Headers */,
> 				413A246B1FE1991A00373E99 /* UIDevice+RTCDevice.h in Headers */,
> 				5CDD88E01E43BE3D00621E92 /* ulpfec_generator.h in Headers */,
>+				416D3BE4212731C200775F09 /* fixed_gain_controller.h in Headers */,
> 				5CDD88E31E43BE3D00621E92 /* ulpfec_header_reader_writer.h in Headers */,
> 				419C83C61FE247B40040C30F /* ulpfec_receiver.h in Headers */,
> 				5CDD88E51E43BE3D00621E92 /* ulpfec_receiver_impl.h in Headers */,
>@@ -11375,6 +12845,7 @@
> 				5CDD87091E43BA7500621E92 /* utility.h in Headers */,
> 				5CDD87DF1E43BC2E00621E92 /* utility.h in Headers */,
> 				5CDD873B1E43BABE00621E92 /* vad_audio_proc.h in Headers */,
>+				41A08BBE21269553001D5D7B /* rtc_event_ice_candidate_pair.h in Headers */,
> 				5CDD87381E43BABE00621E92 /* vad_audio_proc_internal.h in Headers */,
> 				5CDD873E1E43BABE00621E92 /* vad_circular_buffer.h in Headers */,
> 				5CDD86D71E43BA2800621E92 /* vad_core.h in Headers */,
>@@ -11382,6 +12853,7 @@
> 				5CDD86DD1E43BA2800621E92 /* vad_gmm.h in Headers */,
> 				5CDD86E01E43BA2800621E92 /* vad_sp.h in Headers */,
> 				5C63F8E01E416D53002CA531 /* value.h in Headers */,
>+				419241532127376F00634FCF /* fakedtlstransport.h in Headers */,
> 				41F9BFD82051DDE500ABF0B9 /* vector_buffer.h in Headers */,
> 				5CDD84021E439A6F00621E92 /* video_coding_impl.h in Headers */,
> 				419C84131FE249AB0040C30F /* video_content_type.h in Headers */,
>@@ -11391,19 +12863,27 @@
> 				4145F61F1FE1F38D00EB9CAF /* video_encoder_factory.h in Headers */,
> 				5CD2846B1E6A57F40094FDC8 /* video_frame.h in Headers */,
> 				41109AAE1E5FA19200C0955A /* video_frame_buffer.h in Headers */,
>+				41A08BB3212681C8001D5D7B /* video_stream_encoder_create.h in Headers */,
> 				5CD284691E6A57F40094FDC8 /* video_frame_buffer.h in Headers */,
>+				419241DD21275A3000634FCF /* time_delta.h in Headers */,
> 				419C82C41FE20DCD0040C30F /* video_receive_stream.h in Headers */,
>+				41F263AE212680A800274F59 /* fallthrough.h in Headers */,
>+				41E02CAC2127352D00C27CD6 /* bitrate_estimator.h in Headers */,
> 				5CDD85CB1E43B5C000621E92 /* video_receive_stream.h in Headers */,
> 				5C4B4C211E431F75002651C8 /* video_render_frames.h in Headers */,
> 				5CD2846C1E6A57F40094FDC8 /* video_rotation.h in Headers */,
> 				419C82BA1FE20DCD0040C30F /* video_send_stream.h in Headers */,
> 				5CDD85CE1E43B5C000621E92 /* video_send_stream.h in Headers */,
>+				41A08BB5212681C8001D5D7B /* video_stream_encoder_observer.h in Headers */,
> 				5CDD85D01E43B5C000621E92 /* video_stream_decoder.h in Headers */,
> 				419C84121FE249AB0040C30F /* video_timing.h in Headers */,
>+				419241622127385B00634FCF /* rw_lock_posix.h in Headers */,
> 				5C4B48FD1E42C1E3002651C8 /* videoadapter.h in Headers */,
> 				5C4B49001E42C1E3002651C8 /* videobroadcaster.h in Headers */,
> 				5C4B49031E42C1E3002651C8 /* videocapturer.h in Headers */,
>+				419241E021275A3000634FCF /* timestamp.h in Headers */,
> 				5C4B49041E42C1E3002651C8 /* videocapturerfactory.h in Headers */,
>+				419241572127376F00634FCF /* udptransport.h in Headers */,
> 				5CD285151E6A60570094FDC8 /* videocapturertracksource.h in Headers */,
> 				5C4B49071E42C1E3002651C8 /* videocommon.h in Headers */,
> 				41ECEAFE20646664009D5141 /* VideoProcessingSoftLink.h in Headers */,
>@@ -11414,6 +12894,7 @@
> 				5CD285171E6A60570094FDC8 /* videotrack.h in Headers */,
> 				5CD285191E6A60570094FDC8 /* videotracksource.h in Headers */,
> 				413A22BC1FE18E0700373E99 /* virtualsocketserver.h in Headers */,
>+				41F263B3212680A800274F59 /* arch.h in Headers */,
> 				5CDD87411E43BABE00621E92 /* voice_activity_detector.h in Headers */,
> 				5CDD84C81E43AF1300621E92 /* voice_detection_impl.h in Headers */,
> 				5CDD87421E43BABE00621E92 /* voice_gmm_tables.h in Headers */,
>@@ -11426,10 +12907,12 @@
> 				5C4B4C811E431F9C002651C8 /* wav_file.h in Headers */,
> 				5C4B4C841E431F9C002651C8 /* wav_header.h in Headers */,
> 				41ECEABC206403C2009D5141 /* WebKitUtilities.h in Headers */,
>+				419241AC212758D300634FCF /* sequence_buffer.h in Headers */,
> 				5CDD8CAB1E43C6F700621E92 /* webrtc_cng.h in Headers */,
> 				4145E4911EF88EF700FCF6E6 /* webrtc_libyuv.h in Headers */,
> 				41F9BFC72051DCE900ABF0B9 /* webrtc_vad.h in Headers */,
> 				5C4B488E1E42C1BA002651C8 /* webrtcmediaengine.h in Headers */,
>+				4102F6E121273416006AE8D7 /* video_stream_encoder_settings.h in Headers */,
> 				5CD2847A1E6A5D550094FDC8 /* webrtcsdp.h in Headers */,
> 				5CD2851B1E6A60570094FDC8 /* webrtcsessiondescriptionfactory.h in Headers */,
> 				5C4B48911E42C1BA002651C8 /* webrtcvideocapturer.h in Headers */,
>@@ -11651,7 +13134,7 @@
> 			);
> 			runOnlyForDeploymentPostprocessing = 0;
> 			shellPath = /bin/sh;
>-			shellScript = "PRIVATE_HEADERS_FOLDER_PATH=usr/local/include\n\nif [[ \"${DEPLOYMENT_LOCATION}\" == \"NO\" ]]; then\n    PRIVATE_HEADERS_PATH=\"${TARGET_BUILD_DIR%/}/${PRIVATE_HEADERS_FOLDER_PATH}\"\nelse\n    PRIVATE_HEADERS_PATH=\"${DSTROOT}${INSTALL_PATH_PREFIX%/}/${PRIVATE_HEADERS_FOLDER_PATH}\"\nfi;\n\nmkdir -p \"${PRIVATE_HEADERS_PATH}\"\n\nrsync -av --no-owner --no-group --prune-empty-dirs --exclude \".svn\" --exclude \"usr\" --include \"*/\" --include \"*.h\" --exclude \"*\" \"${SRCROOT}/Source/webrtc\" \"${PRIVATE_HEADERS_PATH}\"\n";
>+			shellScript = "PRIVATE_HEADERS_FOLDER_PATH=usr/local/include\n\nif [[ \"${DEPLOYMENT_LOCATION}\" == \"NO\" ]]; then\n    PRIVATE_HEADERS_PATH=\"${TARGET_BUILD_DIR%/}/${PRIVATE_HEADERS_FOLDER_PATH}\"\nelse\n    PRIVATE_HEADERS_PATH=\"${DSTROOT}${INSTALL_PATH_PREFIX%/}/${PRIVATE_HEADERS_FOLDER_PATH}\"\nfi;\n\nmkdir -p \"${PRIVATE_HEADERS_PATH}\"\n\nrsync -av --no-owner --no-group --prune-empty-dirs --exclude \".svn\" --exclude \"usr\" --include \"*/\" --include \"*.h\" --exclude \"*\" \"${SRCROOT}/Source/webrtc\" \"${PRIVATE_HEADERS_PATH}\"\n\nrsync -av --no-owner --no-group --prune-empty-dirs --exclude \".svn\" --exclude \"usr\" --include \"*/\" --include \"*.h\" --exclude \"*\" \"${SRCROOT}/Source/third_party/abseil-cpp/absl\" \"${PRIVATE_HEADERS_PATH}\"\n";
> 		};
> 		5CD286461E6E154E0094FDC8 /* Check for Weak VTables and Externals */ = {
> 			isa = PBXShellScriptBuildPhase;
>@@ -12227,6 +13710,7 @@
> 			isa = PBXSourcesBuildPhase;
> 			buildActionMask = 2147483647;
> 			files = (
>+				415F1FB121272FBA00064CBF /* gain_curve_applier.cc in Sources */,
> 				5CDD8B6C1E43C2B500621E92 /* abs_quant.c in Sources */,
> 				5CDD8B6A1E43C2B500621E92 /* abs_quant_loop.c in Sources */,
> 				5CDD89F81E43BFB300621E92 /* accelerate.cc in Sources */,
>@@ -12234,18 +13718,23 @@
> 				5CDD85091E43B1EA00621E92 /* acm_receiver.cc in Sources */,
> 				5CDD850B1E43B1EA00621E92 /* acm_resampler.cc in Sources */,
> 				5C4B48D71E42C1E3002651C8 /* adaptedvideotracksource.cc in Sources */,
>+				419241C52127593F00634FCF /* loss_rate_filter.cc in Sources */,
>+				419241542127376F00634FCF /* basicasyncresolverfactory.cc in Sources */,
> 				5CD285A41E6A63430094FDC8 /* adaptive_fir_filter.cc in Sources */,
> 				41F411C71EF97BAC00343C26 /* adm_helpers.cc in Sources */,
> 				5CD285A81E6A63430094FDC8 /* aec3_common.cc in Sources */,
> 				5CD285AA1E6A63430094FDC8 /* aec3_fft.cc in Sources */,
>+				41F263B72126810C00274F59 /* base64.cc in Sources */,
> 				5CDD86BC1E43B9C200621E92 /* aec_core.cc in Sources */,
> 				5CDD86BB1E43B9C200621E92 /* aec_core_sse2.cc in Sources */,
> 				413092131EF8D51100757C55 /* aec_dump.cc in Sources */,
> 				5CDD86BE1E43B9C200621E92 /* aec_resampler.cc in Sources */,
> 				5CD285A61E6A63430094FDC8 /* aec_state.cc in Sources */,
>+				415F1FC02127304C00064CBF /* echo_cancellation_proxy.cc in Sources */,
> 				5CDD85E71E43B81000621E92 /* aecm_core.cc in Sources */,
> 				5CDD85E41E43B81000621E92 /* aecm_core_c.cc in Sources */,
> 				5CDD87011E43BA7500621E92 /* agc.cc in Sources */,
>+				41F2639221267F5300274F59 /* g722_encode.c in Sources */,
> 				5CDD86FF1E43BA7500621E92 /* agc_manager_direct.cc in Sources */,
> 				5CDD8FD11E43CD6600621E92 /* aimd_rate_control.cc in Sources */,
> 				41F9BFCB2051DDA200ABF0B9 /* alr_experiment.cc in Sources */,
>@@ -12253,15 +13742,22 @@
> 				5CDD87E51E43BDA100621E92 /* apm_data_dumper.cc in Sources */,
> 				5CD284C21E6A5FFC0094FDC8 /* apm_helpers.cc in Sources */,
> 				5CDD893B1E43BF3A00621E92 /* app.cc in Sources */,
>+				419241642127385B00634FCF /* rw_lock_posix.cc in Sources */,
> 				5CDD87A91E43BC0500621E92 /* arith_routines.c in Sources */,
> 				41433D0A1F79B33400387B4D /* arith_routines.c in Sources */,
>+				4102F6AB212732B1006AE8D7 /* rtp_payload_params.cc in Sources */,
>+				41299B8C2127365100B3414B /* send_time_history.cc in Sources */,
> 				5CDD87A71E43BC0500621E92 /* arith_routines_hist.c in Sources */,
> 				41433D291F79B33400387B4D /* arith_routines_hist.c in Sources */,
>+				41E02C8B212734B900C27CD6 /* decoder_database.cc in Sources */,
>+				41F2638A21267F4000274F59 /* fft.c in Sources */,
> 				5CDD87A81E43BC0500621E92 /* arith_routines_logist.c in Sources */,
>+				41E02C88212734B900C27CD6 /* encoder_database.cc in Sources */,
> 				41433D0F1F79B33400387B4D /* arith_routines_logist.c in Sources */,
> 				413A238C1FE18E0700373E99 /* asyncinvoker.cc in Sources */,
> 				413A216D1FE18D6A00373E99 /* asyncpacketsocket.cc in Sources */,
> 				413A23551FE18E0700373E99 /* asyncresolverinterface.cc in Sources */,
>+				41299B8B2127365100B3414B /* pacer_controller.cc in Sources */,
> 				413A23961FE18E0800373E99 /* asyncsocket.cc in Sources */,
> 				5C63F9E01E4174F6002CA531 /* asyncstuntcpsocket.cc in Sources */,
> 				413A23AB1FE18E0800373E99 /* asynctcpsocket.cc in Sources */,
>@@ -12269,6 +13765,7 @@
> 				5CDD84A91E43AF1300621E92 /* audio_buffer.cc in Sources */,
> 				5CDD85101E43B1EA00621E92 /* audio_coding_module.cc in Sources */,
> 				5C4B4C591E431F9C002651C8 /* audio_converter.cc in Sources */,
>+				41DDB26E212679D200296D47 /* string_builder.cc in Sources */,
> 				5CD285441E6A61D20094FDC8 /* audio_decoder.cc in Sources */,
> 				419C83E91FE248350040C30F /* audio_decoder_g711.cc in Sources */,
> 				4140B8311E4E3396007409E6 /* audio_decoder_g722.cc in Sources */,
>@@ -12276,7 +13773,9 @@
> 				5CDD8B6E1E43C2B500621E92 /* audio_decoder_ilbc.cc in Sources */,
> 				419C84BF1FE319440040C30F /* audio_decoder_ilbc.cc in Sources */,
> 				5CDD87AB1E43BC0500621E92 /* audio_decoder_isac.cc in Sources */,
>+				4192419A2127586500634FCF /* kiss_fft.cc in Sources */,
> 				419C835C1FE246BB0040C30F /* audio_decoder_isac_fix.cc in Sources */,
>+				419241EE21275AFA00634FCF /* simulcast_rate_allocator.cc in Sources */,
> 				419C836A1FE246CA0040C30F /* audio_decoder_isac_float.cc in Sources */,
> 				41433CF91F79B33400387B4D /* audio_decoder_isacfix.cc in Sources */,
> 				419C83F21FE2488D0040C30F /* audio_decoder_L16.cc in Sources */,
>@@ -12285,6 +13784,8 @@
> 				4140B8221E4E3383007409E6 /* audio_decoder_pcm.cc in Sources */,
> 				5CDD8FFC1E43CE3A00621E92 /* audio_decoder_pcm16b.cc in Sources */,
> 				5C119FFB1E457400004F0987 /* audio_device_buffer.cc in Sources */,
>+				419241E221275A3000634FCF /* time_delta.cc in Sources */,
>+				41F2639421267F5300274F59 /* g722_decode.c in Sources */,
> 				5C11A01D1E457578004F0987 /* audio_device_dummy.cc in Sources */,
> 				5C119FFE1E457400004F0987 /* audio_device_generic.cc in Sources */,
> 				5C11A00B1E457448004F0987 /* audio_device_mac.cc in Sources */,
>@@ -12292,12 +13793,18 @@
> 				5CDD8CA71E43C6F700621E92 /* audio_encoder_cng.cc in Sources */,
> 				5CDD8FF31E43CDF400621E92 /* audio_encoder_copy_red.cc in Sources */,
> 				419C83E71FE248350040C30F /* audio_encoder_g711.cc in Sources */,
>+				41F2636221267ADF00274F59 /* sdp_video_format.cc in Sources */,
>+				419241EF21275AFA00634FCF /* simulcast_utility.cc in Sources */,
> 				4140B8331E4E3396007409E6 /* audio_encoder_g722.cc in Sources */,
> 				419C833E1FE246230040C30F /* audio_encoder_g722.cc in Sources */,
> 				5CDD8B701E43C2B500621E92 /* audio_encoder_ilbc.cc in Sources */,
> 				419C84BD1FE319440040C30F /* audio_encoder_ilbc.cc in Sources */,
> 				5CDD87AD1E43BC0500621E92 /* audio_encoder_isac.cc in Sources */,
> 				419C836C1FE246CA0040C30F /* audio_encoder_isac_fix.cc in Sources */,
>+				416D3BDB212731C200775F09 /* adaptive_agc.cc in Sources */,
>+				416D3BE0212731C200775F09 /* biquad_filter.cc in Sources */,
>+				41924179212738FB00634FCF /* rtc_event_log_impl.cc in Sources */,
>+				4102F6BA21273382006AE8D7 /* audio_receive_stream.cc in Sources */,
> 				419C836B1FE246CA0040C30F /* audio_encoder_isac_float.cc in Sources */,
> 				41433CFB1F79B33400387B4D /* audio_encoder_isacfix.cc in Sources */,
> 				419C83F11FE2488D0040C30F /* audio_encoder_L16.cc in Sources */,
>@@ -12305,10 +13812,12 @@
> 				419C82F51FE20EB50040C30F /* audio_encoder_opus.cc in Sources */,
> 				419C82F31FE20EB50040C30F /* audio_encoder_opus_config.cc in Sources */,
> 				4140B8201E4E3383007409E6 /* audio_encoder_pcm.cc in Sources */,
>+				41F2638321267B7700274F59 /* rtpreceiverinterface.cc in Sources */,
> 				5CDD8FFE1E43CE3A00621E92 /* audio_encoder_pcm16b.cc in Sources */,
> 				5CD285461E6A61D20094FDC8 /* audio_format.cc in Sources */,
> 				5CDD84191E439B2900621E92 /* audio_format_conversion.cc in Sources */,
> 				5CD284921E6A5F410094FDC8 /* audio_frame_manipulator.cc in Sources */,
>+				41D6B45C21273159008F9353 /* rtp_video_sender.cc in Sources */,
> 				5CD2854D1E6A62130094FDC8 /* audio_frame_operations.cc in Sources */,
> 				5CDD8A8F1E43C00F00621E92 /* audio_loop.cc in Sources */,
> 				5CD284951E6A5F410094FDC8 /* audio_mixer_impl.cc in Sources */,
>@@ -12316,10 +13825,12 @@
> 				5CDD8A011E43BFB300621E92 /* audio_multi_vector.cc in Sources */,
> 				419C830D1FE20F3C0040C30F /* audio_network_adaptor_config.cc in Sources */,
> 				5CDD8F791E43CBE000621E92 /* audio_network_adaptor_impl.cc in Sources */,
>+				419241C72127593F00634FCF /* bbr_network_controller.cc in Sources */,
> 				5CDD8FE91E43CDCA00621E92 /* audio_processing.cc in Sources */,
> 				5CDD84AB1E43AF1300621E92 /* audio_processing_impl.cc in Sources */,
> 				419C829D1FE20D1C0040C30F /* audio_processing_statistics.cc in Sources */,
> 				5CDD8ACC1E43C23900621E92 /* audio_receive_stream.cc in Sources */,
>+				416D3BED212731C200775F09 /* vector_float_frame.cc in Sources */,
> 				5C4B4C5C1E431F9C002651C8 /* audio_ring_buffer.cc in Sources */,
> 				5CDD8ACF1E43C23900621E92 /* audio_send_stream.cc in Sources */,
> 				5CD284B11E6A5F9F0094FDC8 /* audio_send_stream.cc in Sources */,
>@@ -12328,6 +13839,7 @@
> 				41F9BF9B2051C84C00ABF0B9 /* audio_transport_impl.cc in Sources */,
> 				5C4B4C5F1E431F9C002651C8 /* audio_util.cc in Sources */,
> 				5CDD8A041E43BFB300621E92 /* audio_vector.cc in Sources */,
>+				41E02CAE2127352D00C27CD6 /* bitrate_estimator.cc in Sources */,
> 				5CD284F01E6A60570094FDC8 /* audiotrack.cc in Sources */,
> 				5CDD8B721E43C2B500621E92 /* augmented_cb_corr.c in Sources */,
> 				5CDD86401E43B8B500621E92 /* auto_corr_to_refl_coef.c in Sources */,
>@@ -12335,13 +13847,16 @@
> 				5CDD8A071E43BFB300621E92 /* background_noise.cc in Sources */,
> 				41433CF71F79B33400387B4D /* bandwidth_estimator.c in Sources */,
> 				5CDD87AE1E43BC0500621E92 /* bandwidth_estimator.c in Sources */,
>+				41E02CCC212735B700C27CD6 /* bitrate_settings.cc in Sources */,
> 				5C63F9E21E4174F6002CA531 /* basicpacketsocketfactory.cc in Sources */,
>+				4102F69221273206006AE8D7 /* rtcstatstraversal.cc in Sources */,
> 				5C4B4D091E4322F6002651C8 /* basicportallocator.cc in Sources */,
> 				419C846F1FE255FA0040C30F /* bio_ssl.cc in Sources */,
> 				413A233B1FE18E0700373E99 /* bitbuffer.cc in Sources */,
> 				5C4B4C191E431F75002651C8 /* bitrate_adjuster.cc in Sources */,
> 				5CDD852F1E43B39C00621E92 /* bitrate_allocator.cc in Sources */,
> 				5CDD8F7D1E43CBE000621E92 /* bitrate_controller.cc in Sources */,
>+				4102F6BC21273382006AE8D7 /* flexfec_receive_stream.cc in Sources */,
> 				5CDD905D1E43D11200621E92 /* bitrate_controller_impl.cc in Sources */,
> 				5CDD8FA01E43CCBE00621E92 /* bitrate_prober.cc in Sources */,
> 				413A21751FE18D7100373E99 /* bitrateallocationstrategy.cc in Sources */,
>@@ -12349,17 +13864,22 @@
> 				5CDD86871E43B93900621E92 /* block_mean_calculator.cc in Sources */,
> 				5CD285B01E6A63430094FDC8 /* block_processor.cc in Sources */,
> 				5CD285AE1E6A63430094FDC8 /* block_processor_metrics.cc in Sources */,
>+				4102F69021273206006AE8D7 /* jseptransportcontroller.cc in Sources */,
> 				5C4B4C611E431F9C002651C8 /* blocker.cc in Sources */,
> 				5CDD8A0A1E43BFB300621E92 /* buffer_level_filter.cc in Sources */,
>+				41DDB264212679A300296D47 /* opensslcertificate.cc in Sources */,
> 				413A239B1FE18E0800373E99 /* bufferqueue.cc in Sources */,
> 				5CD285481E6A61D20094FDC8 /* builtin_audio_decoder_factory.cc in Sources */,
>+				41F2637321267B4A00274F59 /* spl_sqrt_floor.c in Sources */,
> 				413091FB1EF8CFF800757C55 /* builtin_audio_encoder_factory.cc in Sources */,
> 				5CDD8B741E43C2B500621E92 /* bw_expand.c in Sources */,
> 				5CDD8FD31E43CD6600621E92 /* bwe_defines.cc in Sources */,
> 				5CDD893E1E43BF3A00621E92 /* bye.cc in Sources */,
>+				4192423E21276B5A00634FCF /* VideoToolBoxEncoderFactory.cpp in Sources */,
> 				413A23F21FE18E0800373E99 /* bytebuffer.cc in Sources */,
> 				5CDD85341E43B39C00621E92 /* call.cc in Sources */,
> 				5CDD85121E43B1EA00621E92 /* call_statistics.cc in Sources */,
>+				419241442127372800634FCF /* pitch_search_internal.cc in Sources */,
> 				5CDD859C1E43B5C000621E92 /* call_stats.cc in Sources */,
> 				4145E4E41EF8CD7D00FCF6E6 /* callfactory.cc in Sources */,
> 				419C83121FE242800040C30F /* candidate.cc in Sources */,
>@@ -12371,10 +13891,12 @@
> 				5CDD8B801E43C2B500621E92 /* cb_search.c in Sources */,
> 				5CDD8B7E1E43C2B500621E92 /* cb_search_core.c in Sources */,
> 				5CDD8B821E43C2B500621E92 /* cb_update_best_index.c in Sources */,
>+				4102F6DB21273416006AE8D7 /* video_stream_decoder_create.cc in Sources */,
> 				5C4B4C641E431F9C002651C8 /* channel_buffer.cc in Sources */,
> 				5CDD8F801E43CBE000621E92 /* channel_controller.cc in Sources */,
> 				5CDD8B841E43C2B500621E92 /* chebyshev.c in Sources */,
> 				413A23731FE18E0700373E99 /* checks.cc in Sources */,
>+				419241AD212758D300634FCF /* spectral_features.cc in Sources */,
> 				5CDD86A11E43B99400621E92 /* circular_buffer.cc in Sources */,
> 				5C4B4CC61E4320A9002651C8 /* clock.cc in Sources */,
> 				5C4B48DB1E42C1E3002651C8 /* codec.cc in Sources */,
>@@ -12383,6 +13905,9 @@
> 				5CDD8A0D1E43BFB300621E92 /* comfort_noise.cc in Sources */,
> 				5CD285B41E6A63430094FDC8 /* comfort_noise_generator.cc in Sources */,
> 				5CDD89411E43BF3A00621E92 /* common_header.cc in Sources */,
>+				419241812127497100634FCF /* bad_variant_access.cc in Sources */,
>+				416D3BEC212731C200775F09 /* vad_with_level.cc in Sources */,
>+				41DDB26F212679D200296D47 /* audio_format_to_string.cc in Sources */,
> 				5CDD8B861E43C2B500621E92 /* comp_corr.c in Sources */,
> 				5CFD538D1E4BD3A300482908 /* compare_neon.cc in Sources */,
> 				5CFD538E1E4BD3A300482908 /* compare_neon64.cc in Sources */,
>@@ -12395,9 +13920,11 @@
> 				419C83FB1FE249370040C30F /* constants.cc in Sources */,
> 				5CDD8F851E43CBE000621E92 /* controller.cc in Sources */,
> 				5CDD8F831E43CBE000621E92 /* controller_manager.cc in Sources */,
>+				41E02C7D2127347400C27CD6 /* rtp_video_header.cc in Sources */,
> 				41F9BFAC2051C91900ABF0B9 /* convert_legacy_video_factory.cc in Sources */,
> 				5CDD86481E43B8B500621E92 /* copy_set_operations.c in Sources */,
> 				413A23151FE18E0700373E99 /* copyonwritebuffer.cc in Sources */,
>+				41F2636921267ADF00274F59 /* video_encoder_software_fallback_wrapper.cc in Sources */,
> 				5C4B4CCA1E4320A9002651C8 /* cpu_features.cc in Sources */,
> 				5C4B4CCB1E4320A9002651C8 /* cpu_info.cc in Sources */,
> 				5CDD87B11E43BC0500621E92 /* crc.c in Sources */,
>@@ -12408,7 +13935,9 @@
> 				5CDD864B1E43B8B500621E92 /* cross_correlation.c in Sources */,
> 				5CDD8A0F1E43BFB300621E92 /* cross_correlation.cc in Sources */,
> 				413A23221FE18E0700373E99 /* cryptstring.cc in Sources */,
>+				41924246212771DC00634FCF /* objc_video_frame.mm in Sources */,
> 				419C848E1FE2574D0040C30F /* custom_extensions.cc in Sources */,
>+				419241E821275A7600634FCF /* field_trial_units.cc in Sources */,
> 				419C848C1FE2574D0040C30F /* d1_both.cc in Sources */,
> 				419C848D1FE2574D0040C30F /* d1_lib.cc in Sources */,
> 				419C848F1FE2574D0040C30F /* d1_pkt.cc in Sources */,
>@@ -12416,7 +13945,9 @@
> 				5CD284F21E6A60570094FDC8 /* datachannel.cc in Sources */,
> 				5CDD8F871E43CBE000621E92 /* debug_dump_writer.cc in Sources */,
> 				419C83F91FE248F10040C30F /* decimator.cc in Sources */,
>+				4192416E2127389B00634FCF /* field_trial_parser.cc in Sources */,
> 				5CDD8A161E43BFB300621E92 /* decision_logic.cc in Sources */,
>+				4102F6A121273262006AE8D7 /* remix_resample.cc in Sources */,
> 				41433D051F79B33400387B4D /* decode.c in Sources */,
> 				5CDD87B41E43BC0500621E92 /* decode.c in Sources */,
> 				5CDD8B8F1E43C2B500621E92 /* decode.c in Sources */,
>@@ -12425,38 +13956,54 @@
> 				41433D0E1F79B33400387B4D /* decode_plc.c in Sources */,
> 				5CDD8B8D1E43C2B500621E92 /* decode_residual.c in Sources */,
> 				5CDD8A191E43BFB300621E92 /* decoder_database.cc in Sources */,
>+				419241E121275A3000634FCF /* data_rate.cc in Sources */,
> 				5CDD8B911E43C2B500621E92 /* decoder_interpolate_lsf.c in Sources */,
>+				419241CA2127593F00634FCF /* rtt_stats.cc in Sources */,
> 				5CDD83CA1E439A6F00621E92 /* decoding_state.cc in Sources */,
> 				5CD284971E6A5F410094FDC8 /* default_output_rate_calculator.cc in Sources */,
>+				41924245212771BB00634FCF /* objc_video_decoder_factory.mm in Sources */,
> 				5CD285391E6A61980094FDC8 /* default_video_bitrate_allocator.cc in Sources */,
> 				5CDD868D1E43B93900621E92 /* delay_estimator.cc in Sources */,
> 				5CDD868B1E43B93900621E92 /* delay_estimator_wrapper.cc in Sources */,
> 				5CDD8A1D1E43BFB300621E92 /* delay_manager.cc in Sources */,
> 				5CDD8A201E43BFB300621E92 /* delay_peak_detector.cc in Sources */,
>+				416D3BE7212731C200775F09 /* noise_level_estimator.cc in Sources */,
>+				4102F68F21273206006AE8D7 /* transportstats.cc in Sources */,
> 				5CDD86EE1E43BA6D00621E92 /* digital_agc.c in Sources */,
> 				5CDD864C1E43B8B500621E92 /* division_operations.c in Sources */,
> 				5CDD89471E43BF3A00621E92 /* dlrr.cc in Sources */,
> 				5CDD8B941E43C2B500621E92 /* do_plc.c in Sources */,
> 				412455451EF87C0F00F11809 /* dot_product_with_scale.cc in Sources */,
> 				5CDD86501E43B8B500621E92 /* downsample_fast.c in Sources */,
>+				4102F6BB21273382006AE8D7 /* degraded_call.cc in Sources */,
> 				4145E4D61EF8CC7600FCF6E6 /* downsampled_render_buffer.cc in Sources */,
> 				5CDD8A231E43BFB300621E92 /* dsp_helper.cc in Sources */,
> 				419C84881FE2574D0040C30F /* dtls_method.cc in Sources */,
> 				419C848A1FE2574D0040C30F /* dtls_record.cc in Sources */,
> 				419C82E41FE20E590040C30F /* dtlssrtptransport.cc in Sources */,
> 				419C83E11FE247F20040C30F /* dtlstransport.cc in Sources */,
>+				41DDB254212678E400296D47 /* platform_thread_types.cc in Sources */,
> 				419C83DE1FE247F20040C30F /* dtlstransportinternal.cc in Sources */,
>+				41E02CB02127352D00C27CD6 /* trendline_estimator.cc in Sources */,
> 				5CDD8A261E43BFB300621E92 /* dtmf_buffer.cc in Sources */,
>+				4102F6C9212733B7006AE8D7 /* video_quality_observer.cc in Sources */,
> 				5CDD88771E43BE3C00621E92 /* dtmf_queue.cc in Sources */,
> 				5CDD8A291E43BFB300621E92 /* dtmf_tone_generator.cc in Sources */,
> 				5CD284F41E6A60570094FDC8 /* dtmfsender.cc in Sources */,
>+				419241E321275A3000634FCF /* timestamp.cc in Sources */,
> 				5CDD8F8A1E43CBE000621E92 /* dtx_controller.cc in Sources */,
> 				5CDD86C11E43B9C200621E92 /* echo_cancellation.cc in Sources */,
> 				5CDD84AE1E43AF1300621E92 /* echo_cancellation_impl.cc in Sources */,
>+				4102F6A421273262006AE8D7 /* audio_level.cc in Sources */,
>+				419241D3212759A100634FCF /* fec_private_tables_random.cc in Sources */,
> 				5CD285B81E6A63430094FDC8 /* echo_canceller3.cc in Sources */,
>+				41F2639A21267F5E00274F59 /* pa_ringbuffer.c in Sources */,
> 				5CDD85EA1E43B81000621E92 /* echo_control_mobile.cc in Sources */,
>+				419241F121275B2B00634FCF /* simulcast_encoder_adapter.cc in Sources */,
>+				416D3BE8212731C200775F09 /* noise_spectrum_estimator.cc in Sources */,
> 				5CDD84B01E43AF1300621E92 /* echo_control_mobile_impl.cc in Sources */,
> 				5CD285BA1E6A63430094FDC8 /* echo_path_delay_estimator.cc in Sources */,
>+				419241C42127593F00634FCF /* bandwidth_sampler.cc in Sources */,
> 				5CD285BC1E6A63430094FDC8 /* echo_path_variability.cc in Sources */,
> 				5CD285C01E6A63430094FDC8 /* echo_remover.cc in Sources */,
> 				5CD285BE1E6A63430094FDC8 /* echo_remover_metrics.cc in Sources */,
>@@ -12466,10 +14013,12 @@
> 				5CDD87B51E43BC0500621E92 /* encode_lpc_swb.c in Sources */,
> 				5CDD8A951E43C00F00621E92 /* encode_neteq_input.cc in Sources */,
> 				5CDD83CC1E439A6F00621E92 /* encoded_frame.cc in Sources */,
>+				41A08BAB212681C8001D5D7B /* encoded_frame.cc in Sources */,
> 				5CDD859F1E43B5C000621E92 /* encoder_rtcp_feedback.cc in Sources */,
> 				5CDD86511E43B8B500621E92 /* energy.c in Sources */,
> 				5CDD8B981E43C2B500621E92 /* energy_inverse.c in Sources */,
> 				5CDD8B9A1E43C2B500621E92 /* enh_upsample.c in Sources */,
>+				41E02CBE2127356A00C27CD6 /* sslcertificate.cc in Sources */,
> 				5CDD8B9E1E43C2B500621E92 /* enhancer.c in Sources */,
> 				5CDD8B9C1E43C2B500621E92 /* enhancer_interface.c in Sources */,
> 				5CDD87B81E43BC0500621E92 /* entropy_coding.c in Sources */,
>@@ -12477,6 +14026,8 @@
> 				5CD285C21E6A63430094FDC8 /* erl_estimator.cc in Sources */,
> 				5CD285C41E6A63430094FDC8 /* erle_estimator.cc in Sources */,
> 				419C84921FE2591F0040C30F /* err_data.c in Sources */,
>+				4102F6DC21273416006AE8D7 /* color_space.cc in Sources */,
>+				41E02C7A2127347400C27CD6 /* rtp_generic_frame_descriptor.cc in Sources */,
> 				5C4B4CD21E4320A9002651C8 /* event.cc in Sources */,
> 				413A23B71FE18E0800373E99 /* event.cc in Sources */,
> 				5CD286141E6A66130094FDC8 /* event_log_writer.cc in Sources */,
>@@ -12486,6 +14037,7 @@
> 				5CDD8A2C1E43BFB300621E92 /* expand.cc in Sources */,
> 				5CDD894A1E43BF3A00621E92 /* extended_jitter_report.cc in Sources */,
> 				5CDD894D1E43BF3A00621E92 /* extended_reports.cc in Sources */,
>+				41E02C8D212734B900C27CD6 /* video_coding_defines.cc in Sources */,
> 				5CDD8A971E43C00F00621E92 /* fake_decode_from_file.cc in Sources */,
> 				413A23101FE18E0700373E99 /* fakeclock.cc in Sources */,
> 				413A23CC1FE18E0800373E99 /* fakesslidentity.cc in Sources */,
>@@ -12503,23 +14055,27 @@
> 				413A23B51FE18E0800373E99 /* fileutils.cc in Sources */,
> 				5CDD86551E43B8B500621E92 /* filter_ar.c in Sources */,
> 				5CDD86541E43B8B500621E92 /* filter_ar_fast_q12.c in Sources */,
>+				419241992127586500634FCF /* rnn_vad_weights.cc in Sources */,
> 				5CDD87BC1E43BC0500621E92 /* filter_functions.c in Sources */,
> 				5CDD86561E43B8B500621E92 /* filter_ma_fast_q12.c in Sources */,
> 				41433D271F79B33400387B4D /* filterbank_tables.c in Sources */,
> 				41433D061F79B33400387B4D /* filterbanks.c in Sources */,
> 				5CDD87BF1E43BC0500621E92 /* filterbanks.c in Sources */,
> 				5CDD8BA01E43C2B500621E92 /* filtered_cb_vecs.c in Sources */,
>+				4192413D2127372400634FCF /* fft_util.cc in Sources */,
> 				41433D041F79B33400387B4D /* filters.c in Sources */,
> 				5C11A0031E457400004F0987 /* fine_audio_buffer.cc in Sources */,
> 				5CDD89501E43BF3A00621E92 /* fir.cc in Sources */,
> 				419C831B1FE242B20040C30F /* fir_filter_c.cc in Sources */,
> 				419C831D1FE242B20040C30F /* fir_filter_factory.cc in Sources */,
> 				5C4B4C6A1E431F9C002651C8 /* fir_filter_sse.cc in Sources */,
>+				4192424121276F0200634FCF /* VideoToolBoxDecoderFactory.cpp in Sources */,
> 				413A217D1FE18D7800373E99 /* flags.cc in Sources */,
> 				5CDD887E1E43BE3C00621E92 /* flexfec_header_reader_writer.cc in Sources */,
> 				5CD284B91E6A5F9F0094FDC8 /* flexfec_receive_stream_impl.cc in Sources */,
> 				5CDD88831E43BE3C00621E92 /* flexfec_receiver.cc in Sources */,
> 				5CDD88851E43BE3C00621E92 /* flexfec_sender.cc in Sources */,
>+				41DDB2592126792800296D47 /* zero_memory.cc in Sources */,
> 				5CDD88881E43BE3C00621E92 /* forward_error_correction.cc in Sources */,
> 				5CDD88861E43BE3C00621E92 /* forward_error_correction_internal.cc in Sources */,
> 				5CD285C91E6A63430094FDC8 /* frame_blocker.cc in Sources */,
>@@ -12532,16 +14088,21 @@
> 				5CDD83D31E439A6F00621E92 /* frame_object.cc in Sources */,
> 				4140B8241E4E3383007409E6 /* g711_interface.c in Sources */,
> 				4140B8381E4E3396007409E6 /* g722_interface.c in Sources */,
>+				41F2635F21267ADF00274F59 /* builtin_video_decoder_factory.cc in Sources */,
> 				5CDD84B21E43AF1300621E92 /* gain_control_for_experimental_agc.cc in Sources */,
>+				4102F6A921273262006AE8D7 /* transport_feedback_packet_loss_tracker.cc in Sources */,
> 				5CDD84B41E43AF1300621E92 /* gain_control_impl.cc in Sources */,
> 				5CDD8BA41E43C2B500621E92 /* gain_dequant.c in Sources */,
> 				5CDD8BA61E43C2B500621E92 /* gain_quant.c in Sources */,
>+				4192413E2127372400634FCF /* lp_residual.cc in Sources */,
>+				41F2638D21267F4900274F59 /* g711.c in Sources */,
> 				5CDD83D51E439A6F00621E92 /* generic_decoder.cc in Sources */,
> 				5CDD83D71E439A6F00621E92 /* generic_encoder.cc in Sources */,
> 				5CDD8BA81E43C2B500621E92 /* get_cd_vec.c in Sources */,
> 				5CDD86571E43B8B500621E92 /* get_hanning_window.c in Sources */,
> 				5CDD8BAA1E43C2B500621E92 /* get_lsp_poly.c in Sources */,
> 				5CDD86581E43B8B500621E92 /* get_scaling_square.c in Sources */,
>+				419241C92127593F00634FCF /* bbr_factory.cc in Sources */,
> 				5CDD8BAC1E43C2B500621E92 /* get_sync_seq.c in Sources */,
> 				5CDD87291E43BABE00621E92 /* gmm.cc in Sources */,
> 				5CDD85581E43B42B00621E92 /* h264.cc in Sources */,
>@@ -12549,14 +14110,21 @@
> 				5CDD83561E43257200621E92 /* h264_common.cc in Sources */,
> 				4145E4BF1EF894F600FCF6E6 /* h264_profile_level_id.cc in Sources */,
> 				5CD285EA1E6A639F0094FDC8 /* h264_sprop_parameter_sets.cc in Sources */,
>+				4192418E212749C800634FCF /* transport_feedback_adapter.cc in Sources */,
> 				5CDD83D91E439A6F00621E92 /* h264_sps_pps_tracker.cc in Sources */,
>+				41F263BB2126811500274F59 /* sigslot.cc in Sources */,
> 				419C84691FE255FA0040C30F /* handshake.cc in Sources */,
>+				41299B912127367B00B3414B /* isac_vad.c in Sources */,
> 				419C84891FE2574D0040C30F /* handshake_client.cc in Sources */,
>+				41299B962127369100B3414B /* fake_network_pipe.cc in Sources */,
> 				419C848B1FE2574D0040C30F /* handshake_server.cc in Sources */,
> 				41ECEAD82064147C009D5141 /* helpers.cc in Sources */,
> 				413A236E1FE18E0700373E99 /* helpers.cc in Sources */,
>+				419241612127385B00634FCF /* rw_lock_wrapper.cc in Sources */,
>+				41E02CB22127352D00C27CD6 /* probe_bitrate_estimator.cc in Sources */,
> 				5CDD83DB1E439A6F00621E92 /* histogram.cc in Sources */,
> 				419C83511FE246650040C30F /* histogram_percentile_counter.cc in Sources */,
>+				41AF2D14212CD6AF00D033D5 /* RTCVideoDecoderH264.mm in Sources */,
> 				5CDD8BAE1E43C2B500621E92 /* hp_input.c in Sources */,
> 				5CDD8BB01E43C2B500621E92 /* hp_output.c in Sources */,
> 				413A23071FE18E0700373E99 /* httpbase.cc in Sources */,
>@@ -12564,6 +14132,7 @@
> 				5CDD90501E43D0E900621E92 /* i420.cc in Sources */,
> 				5CD284671E6A57F40094FDC8 /* i420_buffer.cc in Sources */,
> 				5C4B4C1B1E431F75002651C8 /* i420_buffer_pool.cc in Sources */,
>+				41924244212771B700634FCF /* objc_video_encoder_factory.mm in Sources */,
> 				413091F81EF8CFBD00757C55 /* iceserverparsing.cc in Sources */,
> 				419C83E01FE247F20040C30F /* icetransportinternal.cc in Sources */,
> 				413A23741FE18E0700373E99 /* ifaddrs_converter.cc in Sources */,
>@@ -12578,6 +14147,7 @@
> 				5CDD8A9A1E43C00F00621E92 /* input_audio_file.cc in Sources */,
> 				5CDD8FD51E43CD6600621E92 /* inter_arrival.cc in Sources */,
> 				5CDD83DD1E439A6F00621E92 /* inter_frame_delay.cc in Sources */,
>+				41F2637E21267B7700274F59 /* audio_options.cc in Sources */,
> 				5CD2852F1E6A61110094FDC8 /* internaldecoderfactory.cc in Sources */,
> 				5CD285311E6A61110094FDC8 /* internalencoderfactory.cc in Sources */,
> 				5CDD8BBF1E43C2B500621E92 /* interpolate.c in Sources */,
>@@ -12594,6 +14164,7 @@
> 				5CD284741E6A5D280094FDC8 /* jsepicecandidate.cc in Sources */,
> 				5CD284721E6A5D080094FDC8 /* jsepsessiondescription.cc in Sources */,
> 				41F9BFE32051DE1800ABF0B9 /* jseptransport.cc in Sources */,
>+				41F2639E2126800000274F59 /* aligned_malloc.cc in Sources */,
> 				413A23301FE18E0700373E99 /* json.cc in Sources */,
> 				5C4B4C701E431F9C002651C8 /* lapped_transform.cc in Sources */,
> 				5CDD87C41E43BC0500621E92 /* lattice.c in Sources */,
>@@ -12607,6 +14178,7 @@
> 				5CDD8C0F1E43C34600621E92 /* locked_bandwidth_info.cc in Sources */,
> 				413A22CA1FE18E0700373E99 /* logging.cc in Sources */,
> 				413A22C01FE18E0700373E99 /* logging_mac.mm in Sources */,
>+				419241862127499300634FCF /* packet_queue_interface.cc in Sources */,
> 				413A23BE1FE18E0800373E99 /* logsinks.cc in Sources */,
> 				5CDD87051E43BA7500621E92 /* loudness_histogram.cc in Sources */,
> 				5CD2855D1E6A62ED0094FDC8 /* low_cut_filter.cc in Sources */,
>@@ -12614,14 +14186,18 @@
> 				5CDD8BC11E43C2B500621E92 /* lpc_encode.c in Sources */,
> 				5CDD87C71E43BC0500621E92 /* lpc_gain_swb_tables.c in Sources */,
> 				41433CEE1F79B33400387B4D /* lpc_masking_model.c in Sources */,
>+				419241F521275C3200634FCF /* vp8_encoder_simulcast_proxy.cc in Sources */,
> 				5CDD87C91E43BC0500621E92 /* lpc_shape_swb12_tables.c in Sources */,
> 				5CDD87CB1E43BC0500621E92 /* lpc_shape_swb16_tables.c in Sources */,
> 				5CDD87CD1E43BC0500621E92 /* lpc_tables.c in Sources */,
> 				41433D1E1F79B33400387B4D /* lpc_tables.c in Sources */,
> 				5CDD865B1E43B8B500621E92 /* lpc_to_refl_coef.c in Sources */,
> 				5CDD8BC31E43C2B500621E92 /* lsf_check.c in Sources */,
>+				419241522127376F00634FCF /* regatheringcontroller.cc in Sources */,
>+				41E02CB62127352D00C27CD6 /* goog_cc_network_control.cc in Sources */,
> 				5CDD8BC51E43C2B500621E92 /* lsf_interpolate_to_poly_dec.c in Sources */,
> 				5CDD8BC71E43C2B500621E92 /* lsf_interpolate_to_poly_enc.c in Sources */,
>+				4102F6E021273416006AE8D7 /* i010_buffer.cc in Sources */,
> 				5CDD8BC91E43C2B500621E92 /* lsf_to_lsp.c in Sources */,
> 				5CDD8BCB1E43C2B500621E92 /* lsf_to_poly.c in Sources */,
> 				5CDD8BCD1E43C2B500621E92 /* lsp_to_lsf.c in Sources */,
>@@ -12629,6 +14205,7 @@
> 				413A23381FE18E0700373E99 /* macutils.cc in Sources */,
> 				5CD285CB1E6A63430094FDC8 /* main_filter_update_gain.cc in Sources */,
> 				5CD285CF1E6A63430094FDC8 /* matched_filter.cc in Sources */,
>+				416D3BDE212731C200775F09 /* adaptive_mode_level_estimator.cc in Sources */,
> 				5CD285CD1E6A63430094FDC8 /* matched_filter_lag_aggregator.cc in Sources */,
> 				41F9BFD42051DDE500ABF0B9 /* matrix_buffer.cc in Sources */,
> 				5CDD86A41E43B99400621E92 /* mean_variance_estimator.cc in Sources */,
>@@ -12667,17 +14244,24 @@
> 				413A23E61FE18E0800373E99 /* network.cc in Sources */,
> 				413A238B1FE18E0700373E99 /* networkmonitor.cc in Sources */,
> 				5CDD85FE1E43B84E00621E92 /* noise_suppression.c in Sources */,
>+				41A08BD421272EFA001D5D7B /* round_robin_packet_queue.cc in Sources */,
> 				5CDD84E91E43B0B600621E92 /* noise_suppression_impl.cc in Sources */,
> 				5CDD85FC1E43B84E00621E92 /* noise_suppression_x.c in Sources */,
> 				413A235F1FE18E0700373E99 /* noop.cc in Sources */,
> 				413A23EE1FE18E0800373E99 /* noop.mm in Sources */,
>+				419241702127389B00634FCF /* quality_scaling_experiment.cc in Sources */,
> 				5CDD8A3D1E43BFB300621E92 /* normal.cc in Sources */,
>+				419241A02127588300634FCF /* audio_codec_pair_id.cc in Sources */,
> 				5CDD86A71E43B99400621E92 /* normalized_covariance_estimator.cc in Sources */,
> 				5CDD86001E43B84E00621E92 /* ns_core.c in Sources */,
>+				4102F69321273206006AE8D7 /* rtpparametersconversion.cc in Sources */,
>+				415F1FCE212730F000064CBF /* audio_frame.cc in Sources */,
>+				415F1FA621272FBA00064CBF /* adaptive_digital_gain_applier.cc in Sources */,
> 				41ECEAC220640F28009D5141 /* 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 */,
>+				4192423D212768BD00634FCF /* objc_frame_buffer.mm in Sources */,
> 				419C83B31FE2472E0040C30F /* null_audio_poller.cc in Sources */,
> 				413A23D21FE18E0800373E99 /* nullsocketserver.cc in Sources */,
> 				5CDD86941E43B93900621E92 /* ooura_fft.cc in Sources */,
>@@ -12697,6 +14281,7 @@
> 				5CDD8BD31E43C2B500621E92 /* pack_bits.c in Sources */,
> 				5CDD8A421E43BFB300621E92 /* packet.cc in Sources */,
> 				5CDD83EE1E439A6F00621E92 /* packet.cc in Sources */,
>+				419242422127718D00634FCF /* video_encoder_factory.mm in Sources */,
> 				5CDD8AAF1E43C00F00621E92 /* packet.cc in Sources */,
> 				5CDD8A401E43BFB300621E92 /* packet_buffer.cc in Sources */,
> 				5CDD83EC1E439A6F00621E92 /* packet_buffer.cc in Sources */,
>@@ -12704,9 +14289,11 @@
> 				419C832C1FE245CD0040C30F /* packet_queue.cc in Sources */,
> 				5CDD8FA41E43CCBE00621E92 /* packet_router.cc in Sources */,
> 				5CDD8AAC1E43C00F00621E92 /* packet_source.cc in Sources */,
>+				41AF2D13212CD6AC00D033D5 /* RTCVideoCodecH264.mm in Sources */,
> 				412455561EF887FB00F11809 /* packetlossestimator.cc in Sources */,
> 				419C83F51FE248CA0040C30F /* packetsocketfactory.cc in Sources */,
> 				419C83DA1FE247F20040C30F /* packettransportinternal.cc in Sources */,
>+				416D3BE3212731C200775F09 /* fixed_gain_controller.cc in Sources */,
> 				413A233C1FE18E0700373E99 /* pathutils.cc in Sources */,
> 				5C4B48831E42C1BA002651C8 /* payload_type_mapper.cc in Sources */,
> 				5CDD90001E43CE3A00621E92 /* pcm16b.c in Sources */,
>@@ -12718,6 +14305,7 @@
> 				5CDD87D01E43BC0500621E92 /* pitch_estimator.c in Sources */,
> 				41433CF61F79B33400387B4D /* pitch_estimator_c.c in Sources */,
> 				5CDD87D21E43BC0500621E92 /* pitch_filter.c in Sources */,
>+				41F2636321267ADF00274F59 /* video_codec.cc in Sources */,
> 				41433D111F79B33400387B4D /* pitch_filter.c in Sources */,
> 				41433CF41F79B33400387B4D /* pitch_filter_c.c in Sources */,
> 				41433D101F79B33400387B4D /* pitch_gain_tables.c in Sources */,
>@@ -12742,6 +14330,7 @@
> 				419C83131FE242800040C30F /* proxy.cc in Sources */,
> 				413A23C21FE18E0800373E99 /* proxyinfo.cc in Sources */,
> 				413A22A71FE18E0600373E99 /* proxyserver.cc in Sources */,
>+				4192418C212749C800634FCF /* send_side_congestion_controller.cc in Sources */,
> 				5C63F9FE1E4174F6002CA531 /* pseudotcp.cc in Sources */,
> 				5CDD89581E43BF3A00621E92 /* psfb.cc in Sources */,
> 				5CDD87531E43BAF500621E92 /* push_resampler.cc in Sources */,
>@@ -12749,13 +14338,17 @@
> 				413092091EF8D11200757C55 /* qp_parser.cc in Sources */,
> 				5CDD83811E439A3500621E92 /* quality_scaler.cc in Sources */,
> 				5CD2862A1E6A66C80094FDC8 /* quality_threshold.cc in Sources */,
>+				41F2636521267ADF00274F59 /* video_decoder_software_fallback_wrapper.cc in Sources */,
> 				413A22FB1FE18E0700373E99 /* race_checker.cc in Sources */,
> 				413A23CB1FE18E0800373E99 /* random.cc in Sources */,
> 				5CDD8A4A1E43BFB300621E92 /* random_vector.cc in Sources */,
>+				4102F6AE212732E7006AE8D7 /* expand_uma_logger.cc in Sources */,
> 				5CDD865F1E43B8B500621E92 /* randomization_functions.c in Sources */,
> 				5CDD895B1E43BF3A00621E92 /* rapid_resync_request.cc in Sources */,
> 				413A238D1FE18E0700373E99 /* rate_limiter.cc in Sources */,
> 				413A23581FE18E0700373E99 /* rate_statistics.cc in Sources */,
>+				4102F6E321273416006AE8D7 /* video_bitrate_allocation.cc in Sources */,
>+				41E02CC52127358800C27CD6 /* video_encoder_config.cc in Sources */,
> 				413A23621FE18E0700373E99 /* ratetracker.cc in Sources */,
> 				5CDD86611E43B8B500621E92 /* real_fft.c in Sources */,
> 				5C4B4C771E431F9C002651C8 /* real_fourier.cc in Sources */,
>@@ -12766,17 +14359,22 @@
> 				5CDD83F41E439A6F00621E92 /* receiver.cc in Sources */,
> 				5CDD895E1E43BF3A00621E92 /* receiver_report.cc in Sources */,
> 				5CDD8A4D1E43BFB300621E92 /* red_payload_splitter.cc in Sources */,
>+				41DDB265212679A300296D47 /* opensslutility.cc in Sources */,
> 				5CDD8BD91E43C2B500621E92 /* refiner.c in Sources */,
>+				419241902127581000634FCF /* fec_private_tables_bursty.cc in Sources */,
> 				5CDD86621E43B8B500621E92 /* refl_coef_to_lpc.c in Sources */,
> 				5C63FA011E4174F6002CA531 /* relayport.cc in Sources */,
> 				5C63FA041E4174F6002CA531 /* relayserver.cc in Sources */,
> 				5CDD89611E43BF3A00621E92 /* remb.cc in Sources */,
>+				41E02CC42127358800C27CD6 /* builtin_video_encoder_factory.cc in Sources */,
> 				5CDD8FDB1E43CD6600621E92 /* remote_bitrate_estimator_abs_send_time.cc in Sources */,
> 				5CDD8FDD1E43CD6600621E92 /* remote_bitrate_estimator_single_stream.cc in Sources */,
> 				5CDD8FE11E43CD6600621E92 /* remote_estimator_proxy.cc in Sources */,
>+				41DDB2562126790A00296D47 /* socket.cc in Sources */,
> 				5CDD88951E43BE3C00621E92 /* remote_ntp_time_estimator.cc in Sources */,
> 				5CD285051E6A60570094FDC8 /* remoteaudiosource.cc in Sources */,
> 				4145E4D91EF8CC9B00FCF6E6 /* render_buffer.cc in Sources */,
>+				419241C12127593F00634FCF /* data_transfer_tracker.cc in Sources */,
> 				5CD285D51E6A63430094FDC8 /* render_delay_buffer.cc in Sources */,
> 				5CD285D91E6A63430094FDC8 /* render_delay_controller.cc in Sources */,
> 				5CD285D71E6A63430094FDC8 /* render_delay_controller_metrics.cc in Sources */,
>@@ -12803,17 +14401,20 @@
> 				41F9BFAA2051C8E200ABF0B9 /* rtc_event_alr_state.cc in Sources */,
> 				419C83A21FE246F90040C30F /* rtc_event_audio_network_adaptation.cc in Sources */,
> 				419C83951FE246F90040C30F /* rtc_event_audio_playout.cc in Sources */,
>+				415F1FAC21272FBA00064CBF /* down_sampler.cc in Sources */,
> 				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 */,
> 				419C838D1FE246F90040C30F /* rtc_event_bwe_update_loss_based.cc in Sources */,
> 				5CDD904B1E43CF6100621E92 /* rtc_event_log.cc in Sources */,
>+				41F2636721267ADF00274F59 /* video_decoder.cc in Sources */,
> 				41F411B91EF8DA9600343C26 /* rtc_event_log_factory.cc in Sources */,
> 				419C84241FE24BA60040C30F /* rtc_event_log_output_file.cc in Sources */,
> 				419C838F1FE246F90040C30F /* rtc_event_probe_cluster_created.cc in Sources */,
> 				419C83991FE246F90040C30F /* rtc_event_probe_result_failure.cc in Sources */,
> 				419C83921FE246F90040C30F /* rtc_event_probe_result_success.cc in Sources */,
> 				419C83911FE246F90040C30F /* rtc_event_rtcp_packet_incoming.cc in Sources */,
>+				41E02C7C2127347400C27CD6 /* rtp_packet_to_send.cc in Sources */,
> 				419C839C1FE246F90040C30F /* rtc_event_rtcp_packet_outgoing.cc in Sources */,
> 				419C83241FE245280040C30F /* rtc_event_rtp_packet_incoming.cc in Sources */,
> 				419C83AA1FE247080040C30F /* rtc_event_rtp_packet_outgoing.cc in Sources */,
>@@ -12826,27 +14427,35 @@
> 				41ECEAC820641328009D5141 /* RTCEncodedImage.mm in Sources */,
> 				41F411B01EF8DA0100343C26 /* rtcerror.cc in Sources */,
> 				41ECEACF20641370009D5141 /* RTCI420Buffer.mm in Sources */,
>+				41E02CA92127352D00C27CD6 /* goog_cc_factory.cc in Sources */,
> 				5CDD8A4F1E43BFB300621E92 /* rtcp.cc in Sources */,
>+				419241DF21275A3000634FCF /* data_size.cc in Sources */,
> 				419C82B91FE20DCD0040C30F /* rtcp_demuxer.cc in Sources */,
> 				5CD285FA1E6A64520094FDC8 /* rtcp_nack_stats.cc in Sources */,
>+				415F1FB221272FBA00064CBF /* interpolated_gain_curve.cc in Sources */,
> 				5CDD88971E43BE3C00621E92 /* rtcp_packet.cc in Sources */,
> 				5CDD889A1E43BE3C00621E92 /* rtcp_receiver.cc in Sources */,
>+				415F1FAA21272FBA00064CBF /* compute_interpolated_gain_curve.cc in Sources */,
> 				5CDD889D1E43BE3C00621E92 /* rtcp_sender.cc in Sources */,
> 				419C82FF1FE20F020040C30F /* rtcp_transceiver.cc in Sources */,
>+				41924177212738C200634FCF /* transport.cc in Sources */,
> 				419C83021FE20F020040C30F /* rtcp_transceiver_config.cc in Sources */,
> 				419C83001FE20F020040C30F /* rtcp_transceiver_impl.cc in Sources */,
>+				41D6B45921273159008F9353 /* rtp_bitrate_configurator.cc in Sources */,
> 				41ECEADA206414A6009D5141 /* RTCRtpFragmentationHeader.mm in Sources */,
> 				5C63FA781E417AED002CA531 /* rtcstats.cc in Sources */,
> 				5C63FA771E417AED002CA531 /* rtcstats_objects.cc in Sources */,
>+				415F1FA821272FBA00064CBF /* adaptive_mode_level_estimator_agc.cc in Sources */,
> 				5CD285071E6A60570094FDC8 /* rtcstatscollector.cc in Sources */,
> 				5C63FA791E417AED002CA531 /* rtcstatsreport.cc in Sources */,
> 				41ECEAB720630108009D5141 /* RTCVideoCodec.mm in Sources */,
>-				41ECEAB820630108009D5141 /* RTCVideoCodecH264.mm in Sources */,
>-				41ECEAC3206411E2009D5141 /* RTCVideoDecoderH264.mm in Sources */,
>+				41F2636E21267B3E00274F59 /* fft4g.c in Sources */,
> 				41ECEAC4206411E5009D5141 /* RTCVideoEncoderH264.mm in Sources */,
> 				41ECEAEA20641812009D5141 /* RTCVideoEncoderSettings.mm in Sources */,
> 				41ECEACB2064134F009D5141 /* RTCVideoFrame.mm in Sources */,
>+				41F2638021267B7700274F59 /* datachannelinterface.cc in Sources */,
> 				41ECEAE4206416FF009D5141 /* RTCWrappedNativeVideoDecoder.mm in Sources */,
>+				419241B02127590200634FCF /* rtp_generic_frame_descriptor_extension.cc in Sources */,
> 				41ECEAE2206416FF009D5141 /* RTCWrappedNativeVideoEncoder.mm in Sources */,
> 				419C82B41FE20DCD0040C30F /* rtp_config.cc in Sources */,
> 				413092021EF8D0A600757C55 /* rtp_demuxer.cc in Sources */,
>@@ -12856,9 +14465,13 @@
> 				5CDD88AB1E43BE3C00621E92 /* rtp_format_vp8.cc in Sources */,
> 				5CDD88AE1E43BE3C00621E92 /* rtp_format_vp9.cc in Sources */,
> 				5CDD83F61E439A6F00621E92 /* rtp_frame_reference_finder.cc in Sources */,
>+				41E02CCD212735B700C27CD6 /* network_types.cc in Sources */,
> 				5CDD8AB81E43C00F00621E92 /* rtp_generator.cc in Sources */,
>+				416D3BEB212731C200775F09 /* signal_classifier.cc in Sources */,
> 				4145E4DC1EF8CCEF00FCF6E6 /* rtp_header_extension_map.cc in Sources */,
> 				5CDD88B51E43BE3C00621E92 /* rtp_header_extensions.cc in Sources */,
>+				41A08BC8212695DE001D5D7B /* bad_optional_access.cc in Sources */,
>+				419241A8212758D300634FCF /* spectral_features_internal.cc in Sources */,
> 				5CDD88B71E43BE3C00621E92 /* rtp_header_parser.cc in Sources */,
> 				41F9BF8B2051C80100ABF0B9 /* rtp_headers.cc in Sources */,
> 				5CDD88BE1E43BE3D00621E92 /* rtp_packet.cc in Sources */,
>@@ -12871,6 +14484,7 @@
> 				5CDD88C81E43BE3D00621E92 /* rtp_receiver_video.cc in Sources */,
> 				419C83CD1FE247B40040C30F /* rtp_rtcp_defines.cc in Sources */,
> 				419C82BF1FE20DCD0040C30F /* rtp_rtcp_demuxer_helper.cc in Sources */,
>+				4102F6A821273262006AE8D7 /* channel.cc in Sources */,
> 				5CDD88CC1E43BE3D00621E92 /* rtp_rtcp_impl.cc in Sources */,
> 				5CDD88D31E43BE3D00621E92 /* rtp_sender.cc in Sources */,
> 				5CDD88CE1E43BE3D00621E92 /* rtp_sender_audio.cc in Sources */,
>@@ -12879,7 +14493,9 @@
> 				5CDD85B21E43B5C000621E92 /* rtp_streams_synchronizer.cc in Sources */,
> 				5CD286271E6A669D0094FDC8 /* rtp_to_ntp_estimator.cc in Sources */,
> 				4130921B1EF8D63F00757C55 /* rtp_transport_controller_send.cc in Sources */,
>+				41E02CD42127360700C27CD6 /* sample_counter.cc in Sources */,
> 				5CDD88D51E43BE3D00621E92 /* rtp_utility.cc in Sources */,
>+				415F1FAF21272FBA00064CBF /* gain_applier.cc in Sources */,
> 				413091F41EF8CF9200757C55 /* rtp_video_stream_receiver.cc in Sources */,
> 				5C4B48EB1E42C1E3002651C8 /* rtpdataengine.cc in Sources */,
> 				5CDD896C1E43BF3A00621E92 /* rtpfb.cc in Sources */,
>@@ -12891,29 +14507,36 @@
> 				4145E4CC1EF8CB8B00FCF6E6 /* rtptransport.cc in Sources */,
> 				5C4B48F11E42C1E3002651C8 /* rtputils.cc in Sources */,
> 				5CDD83F81E439A6F00621E92 /* rtt_filter.cc in Sources */,
>+				41E02CB52127352D00C27CD6 /* alr_detector.cc in Sources */,
> 				419C82B71FE20DCD0040C30F /* rtx_receive_stream.cc in Sources */,
> 				41F9BFE62051DECD00ABF0B9 /* runtime_enabled_features_default.cc in Sources */,
>+				41D6B45721273159008F9353 /* receive_time_calculator.cc in Sources */,
> 				419C84701FE255FA0040C30F /* s3_both.cc in Sources */,
> 				419C84771FE255FA0040C30F /* s3_lib.cc in Sources */,
> 				419C84731FE255FA0040C30F /* s3_pkt.cc in Sources */,
> 				5CFD53931E4BD3A300482908 /* scale_neon.cc in Sources */,
> 				5CFD53941E4BD3A300482908 /* scale_neon64.cc in Sources */,
> 				41F9BFB52051C93600ABF0B9 /* scopedvideodecoder.cc in Sources */,
>+				4102F6CB212733B7006AE8D7 /* video_stream_decoder_impl.cc in Sources */,
> 				41F9BFB92051C93600ABF0B9 /* scopedvideoencoder.cc in Sources */,
> 				5CD2863C1E6A681C0094FDC8 /* sctptransport.cc in Sources */,
> 				5CD2850D1E6A60570094FDC8 /* sctputils.cc in Sources */,
>+				41E02CB42127352D00C27CD6 /* probe_controller.cc in Sources */,
> 				5CDD896F1E43BF3A00621E92 /* sdes.cc in Sources */,
> 				419C82DC1FE20E590040C30F /* sdputils.cc in Sources */,
>+				41E02CA82127352D00C27CD6 /* acknowledged_bitrate_estimator.cc in Sources */,
> 				5CDD85B61E43B5C000621E92 /* send_delay_stats.cc in Sources */,
> 				5CDD90611E43D11200621E92 /* send_side_bandwidth_estimation.cc in Sources */,
> 				413092151EF8D5EE00757C55 /* send_side_congestion_controller.cc in Sources */,
> 				5CDD85B91E43B5C000621E92 /* send_statistics_proxy.cc in Sources */,
> 				5CDD89721E43BF3A00621E92 /* sender_report.cc in Sources */,
> 				413A239D1FE18E0800373E99 /* sequenced_task_checker_impl.cc in Sources */,
>+				419241412127372400634FCF /* pitch_search.cc in Sources */,
> 				5CDD83FB1E439A6F00621E92 /* session_info.cc in Sources */,
> 				41F9BFE42051DE1800ABF0B9 /* sessiondescription.cc in Sources */,
> 				5CD285DF1E6A63430094FDC8 /* shadow_filter_update_gain.cc in Sources */,
> 				413A235B1FE18E0700373E99 /* signalthread.cc in Sources */,
>+				416D3BE9212731C200775F09 /* saturation_protector.cc in Sources */,
> 				5CDD8BDB1E43C2B500621E92 /* simple_interpolate_lsf.c in Sources */,
> 				5CDD8BDD1E43C2B500621E92 /* simple_lpc_analysis.c in Sources */,
> 				5CDD8BDF1E43C2B500621E92 /* simple_lsf_dequant.c in Sources */,
>@@ -12923,6 +14546,7 @@
> 				5CDD87E11E43BD7000621E92 /* sinc_resampler_sse.cc in Sources */,
> 				5CDD875E1E43BAF500621E92 /* sinusoidal_linear_chirp_source.cc in Sources */,
> 				5C4B4CE51E4320A9002651C8 /* sleep.cc in Sources */,
>+				419242432127719300634FCF /* video_decoder_factory.mm in Sources */,
> 				5CDD8BE51E43C2B500621E92 /* smooth.c in Sources */,
> 				5CDD8BE31E43C2B500621E92 /* smooth_out_data.c in Sources */,
> 				5CD285F61E6A63F60094FDC8 /* smoothing_filter.cc in Sources */,
>@@ -12932,9 +14556,13 @@
> 				413A22D31FE18E0700373E99 /* socketstream.cc in Sources */,
> 				5CDD8BE71E43C2B500621E92 /* sort_sq.c in Sources */,
> 				5C4B4C7D1E431F9C002651C8 /* sparse_fir_filter.cc in Sources */,
>+				415F1FC62127308F00064CBF /* audio_state.cc in Sources */,
> 				41433D021F79B33400387B4D /* spectrum_ar_model_tables.c in Sources */,
> 				5CDD87D81E43BC0600621E92 /* spectrum_ar_model_tables.c in Sources */,
> 				5CDD866B1E43B8B500621E92 /* spl_init.c in Sources */,
>+				4102F6A621273262006AE8D7 /* channel_proxy.cc in Sources */,
>+				41E02CC02127356A00C27CD6 /* data_rate_limiter.cc in Sources */,
>+				41F2638521267B7700274F59 /* rtptransceiverinterface.cc in Sources */,
> 				5CDD866C1E43B8B500621E92 /* spl_inl.c in Sources */,
> 				5CDD86701E43B8B500621E92 /* spl_sqrt.c in Sources */,
> 				5CDD8BE91E43C2B500621E92 /* split_vq.c in Sources */,
>@@ -12942,17 +14570,22 @@
> 				5CDD84C11E43AF1300621E92 /* splitting_filter.cc in Sources */,
> 				5CDD835F1E43257200621E92 /* sps_parser.cc in Sources */,
> 				5CDD83621E43257200621E92 /* sps_vui_rewriter.cc in Sources */,
>+				4102F6DF21273416006AE8D7 /* video_stream_encoder_create.cc in Sources */,
> 				5CDD86721E43B8B500621E92 /* sqrt_of_one_minus_x_squared.c in Sources */,
> 				419C82DE1FE20E590040C30F /* srtpsession.cc in Sources */,
> 				419C82DF1FE20E590040C30F /* srtptransport.cc in Sources */,
> 				419C845E1FE255FA0040C30F /* ssl_aead_ctx.cc in Sources */,
> 				419C84741FE255FA0040C30F /* ssl_asn1.cc in Sources */,
>+				41A08BC021269553001D5D7B /* rtc_event_ice_candidate_pair.cc in Sources */,
>+				419241462127372800634FCF /* rnn.cc in Sources */,
> 				419C84601FE255FA0040C30F /* ssl_buffer.cc in Sources */,
> 				419C84751FE255FA0040C30F /* ssl_cert.cc in Sources */,
> 				419C84651FE255FA0040C30F /* ssl_cipher.cc in Sources */,
> 				419C84721FE255FA0040C30F /* ssl_file.cc in Sources */,
> 				419C84781FE255FA0040C30F /* ssl_key_share.cc in Sources */,
>+				41D6B45621273159008F9353 /* packet_receiver.cc in Sources */,
> 				419C84761FE255FA0040C30F /* ssl_lib.cc in Sources */,
>+				41A08BB0212681C8001D5D7B /* video_source_interface.cc in Sources */,
> 				419C846C1FE255FA0040C30F /* ssl_privkey.cc in Sources */,
> 				419C845F1FE255FA0040C30F /* ssl_session.cc in Sources */,
> 				419C84611FE255FA0040C30F /* ssl_stat.cc in Sources */,
>@@ -12963,6 +14596,7 @@
> 				413A23F11FE18E0800373E99 /* sslfingerprint.cc in Sources */,
> 				419C84161FE24A8D0040C30F /* sslidentity.cc in Sources */,
> 				419C84171FE24AA30040C30F /* sslstreamadapter.cc in Sources */,
>+				41F2637D21267B7700274F59 /* peerconnectioninterface.cc in Sources */,
> 				5CDD87361E43BABE00621E92 /* standalone_vad.cc in Sources */,
> 				5CDD8BEB1E43C2B500621E92 /* state_construct.c in Sources */,
> 				5CDD8BED1E43C2B500621E92 /* state_search.c in Sources */,
>@@ -12972,8 +14606,10 @@
> 				41DDB25221265BE900296D47 /* optional.cc in Sources */,
> 				5C63F9721E41737B002CA531 /* statstypes.cc in Sources */,
> 				413A23D31FE18E0800373E99 /* stream.cc in Sources */,
>+				4192413A2127372400634FCF /* features_extraction.cc in Sources */,
> 				5CDD85BF1E43B5C000621E92 /* stream_synchronization.cc in Sources */,
> 				5C4B48F41E42C1E3002651C8 /* streamparams.cc in Sources */,
>+				41DDB262212679A300296D47 /* opensslsessioncache.cc in Sources */,
> 				413A23D61FE18E0800373E99 /* string_to_number.cc in Sources */,
> 				413A22B51FE18E0700373E99 /* stringencode.cc in Sources */,
> 				413A23131FE18E0700373E99 /* stringutils.cc in Sources */,
>@@ -12987,6 +14623,7 @@
> 				5CDD8BEF1E43C2B500621E92 /* swap_bytes.c in Sources */,
> 				5CDD8A541E43BFB300621E92 /* sync_buffer.cc in Sources */,
> 				5CD284BE1E6A5F9F0094FDC8 /* syncable.cc in Sources */,
>+				4102F6B321273331006AE8D7 /* mediachannel.cc in Sources */,
> 				419C846D1FE255FA0040C30F /* t1_enc.cc in Sources */,
> 				419C84791FE255FA0040C30F /* t1_lib.cc in Sources */,
> 				5CDD89781E43BF3A00621E92 /* target_bitrate.cc in Sources */,
>@@ -12995,6 +14632,7 @@
> 				5C63FA171E4174F6002CA531 /* tcpport.cc in Sources */,
> 				419C84341FE24E800040C30F /* temporal_layers.cc in Sources */,
> 				413A22A51FE18E0600373E99 /* testechoserver.cc in Sources */,
>+				4192417D2127392300634FCF /* timestamp_extrapolator.cc in Sources */,
> 				413A22E51FE18E0700373E99 /* thread.cc in Sources */,
> 				413A23501FE18E0700373E99 /* thread_checker_impl.cc in Sources */,
> 				413A23DD1FE18E0800373E99 /* thread_darwin.mm in Sources */,
>@@ -13002,14 +14640,19 @@
> 				5CDD8A571E43BFB300621E92 /* tick_timer.cc in Sources */,
> 				419C83351FE245EA0040C30F /* time_interval.cc in Sources */,
> 				5CDD8A5A1E43BFB300621E92 /* time_stretch.cc in Sources */,
>+				41A08BBD21269553001D5D7B /* rtc_event_ice_candidate_pair_config.cc in Sources */,
> 				5CDD88DA1E43BE3D00621E92 /* time_util.cc in Sources */,
> 				5CDD83FD1E439A6F00621E92 /* timestamp_map.cc in Sources */,
> 				5CDD8A5D1E43BFB300621E92 /* timestamp_scaler.cc in Sources */,
> 				413A237B1FE18E0700373E99 /* timestampaligner.cc in Sources */,
>+				415F1FAD21272FBA00064CBF /* fixed_digital_level_estimator.cc in Sources */,
>+				415F1FBE2127304C00064CBF /* echo_control_mobile_proxy.cc in Sources */,
> 				413A23C31FE18E0800373E99 /* timeutils.cc in Sources */,
> 				5CDD83FF1E439A6F00621E92 /* timing.cc in Sources */,
> 				419C84681FE255FA0040C30F /* tls13_both.cc in Sources */,
> 				419C84671FE255FA0040C30F /* tls13_client.cc in Sources */,
>+				41E02CA52127352D00C27CD6 /* delay_based_bwe.cc in Sources */,
>+				41E02C8A212734B900C27CD6 /* fec_controller_default.cc in Sources */,
> 				419C84631FE255FA0040C30F /* tls13_enc.cc in Sources */,
> 				419C845D1FE255FA0040C30F /* tls13_server.cc in Sources */,
> 				419C846B1FE255FA0040C30F /* tls_method.cc in Sources */,
>@@ -13018,10 +14661,14 @@
> 				5CDD897D1E43BF3A00621E92 /* tmmbn.cc in Sources */,
> 				5CDD89801E43BF3A00621E92 /* tmmbr.cc in Sources */,
> 				5CDD88DC1E43BE3D00621E92 /* tmmbr_help.cc in Sources */,
>+				41F2638421267B7700274F59 /* jsepicecandidate.cc in Sources */,
> 				5CD285121E6A60570094FDC8 /* trackmediainfomap.cc in Sources */,
> 				41433D001F79B33400387B4D /* transform.c in Sources */,
>+				41E02C782127347400C27CD6 /* contributing_sources.cc in Sources */,
> 				5CDD87DB1E43BC0600621E92 /* transform.c in Sources */,
> 				41433D161F79B33400387B4D /* transform_tables.c in Sources */,
>+				41F263AC212680A800274F59 /* file_wrapper.cc in Sources */,
>+				41A08BCC21269620001D5D7B /* raw_logging.cc in Sources */,
> 				5CDD8C911E43C66000621E92 /* transient_detector.cc in Sources */,
> 				5CDD8C951E43C66000621E92 /* transient_suppressor.cc in Sources */,
> 				5CDD85C11E43B5C000621E92 /* transport_adapter.cc in Sources */,
>@@ -13030,15 +14677,18 @@
> 				5C63FA251E4174F6002CA531 /* transportdescription.cc in Sources */,
> 				5C63FA281E4174F6002CA531 /* transportdescriptionfactory.cc in Sources */,
> 				5C63FA2C1E4174F6002CA531 /* turnport.cc in Sources */,
>+				4102F6CC212733B7006AE8D7 /* video_send_stream_impl.cc in Sources */,
> 				41F9BFEA2051DEEA00ABF0B9 /* turnportfactory.cc in Sources */,
> 				5C63FA2F1E4174F6002CA531 /* turnserver.cc in Sources */,
> 				5C4B48F91E42C1E3002651C8 /* turnutils.cc in Sources */,
>+				41A08BD021272EE2001D5D7B /* gain_controller2.cc 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 */,
> 				5CDD8BF11E43C2B500621E92 /* unpack_bits.c in Sources */,
>+				41A08BD821272F1D001D5D7B /* congestion_window_pushback_controller.cc in Sources */,
> 				5CDD87DE1E43BC2E00621E92 /* utility.c in Sources */,
> 				5CDD87081E43BA7500621E92 /* utility.cc in Sources */,
> 				5CDD86E31E43BA2800621E92 /* vad.cc in Sources */,
>@@ -13046,6 +14696,7 @@
> 				5CDD873D1E43BABE00621E92 /* vad_circular_buffer.cc in Sources */,
> 				5CDD86D61E43BA2800621E92 /* vad_core.c in Sources */,
> 				5CDD86D91E43BA2800621E92 /* vad_filterbank.c in Sources */,
>+				419241712127389B00634FCF /* congestion_controller_experiment.cc in Sources */,
> 				5CDD86DC1E43BA2800621E92 /* vad_gmm.c in Sources */,
> 				5CDD86DF1E43BA2800621E92 /* vad_sp.c in Sources */,
> 				41F9BFD52051DDE500ABF0B9 /* vector_buffer.cc in Sources */,
>@@ -13061,8 +14712,11 @@
> 				419C82BE1FE20DCD0040C30F /* video_receive_stream.cc in Sources */,
> 				5CDD85CA1E43B5C000621E92 /* video_receive_stream.cc in Sources */,
> 				5CDD84031E439A6F00621E92 /* video_receiver.cc in Sources */,
>+				41E02CB12127352D00C27CD6 /* median_slope_estimator.cc in Sources */,
> 				5C4B4C201E431F75002651C8 /* video_render_frames.cc in Sources */,
>+				416D3BE6212731C200775F09 /* limiter.cc in Sources */,
> 				419C82B31FE20DCD0040C30F /* video_send_stream.cc in Sources */,
>+				41A08BC421269577001D5D7B /* icelogger.cc in Sources */,
> 				5CDD85CD1E43B5C000621E92 /* video_send_stream.cc in Sources */,
> 				5CDD84041E439A6F00621E92 /* video_sender.cc in Sources */,
> 				5CDD85CF1E43B5C000621E92 /* video_stream_decoder.cc in Sources */,
>@@ -13075,11 +14729,13 @@
> 				5C4B49061E42C1E3002651C8 /* videocommon.cc in Sources */,
> 				41ECEAFD20646664009D5141 /* VideoProcessingSoftLink.cpp in Sources */,
> 				5C4B490B1E42C1E3002651C8 /* videosourcebase.cc in Sources */,
>+				419241D02127597D00634FCF /* simulated_network.cc in Sources */,
> 				5CD285161E6A60570094FDC8 /* videotrack.cc in Sources */,
> 				5CD285181E6A60570094FDC8 /* videotracksource.cc in Sources */,
> 				413A23121FE18E0700373E99 /* virtualsocketserver.cc in Sources */,
> 				5CDD87401E43BABE00621E92 /* voice_activity_detector.cc in Sources */,
> 				5CDD84C71E43AF1300621E92 /* voice_detection_impl.cc in Sources */,
>+				41D6B45321273159008F9353 /* call_config.cc in Sources */,
> 				5CFD53821E4BA4F500482908 /* voice_processing_audio_unit.mm in Sources */,
> 				5CDD89861E43BF3A00621E92 /* voip_metric.cc in Sources */,
> 				5CDD83851E439A3500621E92 /* vp8_header_parser.cc in Sources */,
>diff --git a/Source/WTF/ChangeLog b/Source/WTF/ChangeLog
>index d4eaa08ad7f..d43fd9652a9 100644
>--- a/Source/WTF/ChangeLog
>+++ b/Source/WTF/ChangeLog
>@@ -1,3 +1,13 @@
>+2018-08-20  Youenn Fablet  <youenn@apple.com> and Alejandro G. Castro  <alex@igalia.com>
>+
>+        Update libwebrtc up to 984f1a80c0
>+        https://bugs.webkit.org/show_bug.cgi?id=188745
>+        <rdar://problem/43539177>
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        * wtf/DisallowCType.h:
>+
> 2018-08-21  Antti Koivisto  <antti@apple.com>
> 
>         Allow creating WeakPtrs to const objects
>diff --git a/Source/WTF/wtf/DisallowCType.h b/Source/WTF/wtf/DisallowCType.h
>index 30544e729fd..386eb328e03 100644
>--- a/Source/WTF/wtf/DisallowCType.h
>+++ b/Source/WTF/wtf/DisallowCType.h
>@@ -40,7 +40,7 @@
> // are used from wx headers. On GTK+ for Mac many GTK+ files include <libintl.h>
> // or <glib/gi18n-lib.h>, which in turn include <xlocale/_ctype.h> which uses
> // isacii(). 
>-#if !(OS(DARWIN) && PLATFORM(GTK)) && !defined(_LIBCPP_VERSION) && defined(__GLIBC__)
>+#if !(OS(DARWIN) && PLATFORM(GTK)) && !PLATFORM(WPE) && !PLATFORM(GTK) && !defined(_LIBCPP_VERSION) && defined(__GLIBC__)
> 
> #include <ctype.h>
> 
>diff --git a/Source/WebCore/CMakeLists.txt b/Source/WebCore/CMakeLists.txt
>index 2f04d538df0..a9c4a5f4347 100644
>--- a/Source/WebCore/CMakeLists.txt
>+++ b/Source/WebCore/CMakeLists.txt
>@@ -1554,7 +1554,8 @@ endif ()
> 
> if (USE_LIBWEBRTC)
>   list(APPEND WebCore_SYSTEM_INCLUDE_DIRECTORIES "${THIRDPARTY_DIR}/libwebrtc/Source/"
>-      "${THIRDPARTY_DIR}/libwebrtc/Source/webrtc")
>+      "${THIRDPARTY_DIR}/libwebrtc/Source/webrtc"
>+      "${THIRDPARTY_DIR}/libwebrtc/Source/third_party/abseil-cpp")
>   list(APPEND WebCore_LIBRARIES webrtc)
>   list(APPEND WebCore_SOURCES
>       Modules/mediastream/libwebrtc/LibWebRTCDataChannelHandler.cpp
>diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
>index 2ff909029eb..aceaa79f9b0 100644
>--- a/Source/WebCore/ChangeLog
>+++ b/Source/WebCore/ChangeLog
>@@ -1,3 +1,60 @@
>+2018-08-20  Youenn Fablet  <youenn@apple.com>
>+
>+        Update libwebrtc up to 984f1a80c0
>+        https://bugs.webkit.org/show_bug.cgi?id=188745
>+        <rdar://problem/43539177>
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Updated implementation according new webrtc backend.
>+        Instead of modifying libwebrtc header files, we disable unused parameter warning for such headers included in WebCore.
>+        WebCore implementation is updated according new webrtc API, mostly AddRef/Release.
>+        RealtimeOutgoingVideoSource now uses a pixel convolver to convert RGB pixel buffers as I420.
>+
>+        Covered by existing tests.
>+
>+        * CMakeLists.txt:
>+        * Modules/mediastream/libwebrtc/LibWebRTCDataChannelHandler.h:
>+        * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
>+        (WebCore::LibWebRTCMediaEndpoint::addTrack):
>+        (WebCore::LibWebRTCMediaEndpoint::mediaStreamFromRTCStream):
>+        (WebCore::fillEncodingParameters):
>+        (WebCore::fillRtpParameters):
>+        * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
>+        (WebCore::LibWebRTCMediaEndpoint::Release const):
>+        * WebCore.xcodeproj/project.pbxproj:
>+        * platform/mediastream/RealtimeIncomingAudioSource.h:
>+        * platform/mediastream/RealtimeIncomingVideoSource.h:
>+        * platform/mediastream/RealtimeMediaSource.cpp:
>+        (WebCore::RealtimeMediaSource::supportsSizeAndFrameRate):
>+        (WebCore::RealtimeMediaSource::supportsConstraint const):
>+        (WebCore::RealtimeMediaSource::supportsConstraints):
>+        * platform/mediastream/RealtimeOutgoingAudioSource.h:
>+        * platform/mediastream/RealtimeOutgoingVideoSource.h:
>+        * platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp:
>+        (WebCore::GStreamerVideoDecoder::newSampleCallback):
>+        * platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp:
>+        (WebCore::GStreamerVideoEncoder::GStreamerVideoEncoder):
>+        (WebCore::GStreamerVideoEncoder::newSampleCallback):
>+        * platform/mediastream/libwebrtc/LibWebRTCProvider.h:
>+        * platform/mediastream/libwebrtc/LibWebRTCAudioModule.h:
>+        * platform/mediastream/libwebrtc/LibWebRTCProvider.cpp:
>+        * platform/mediastream/libwebrtc/LibWebRTCProvider.h:
>+        * platform/mediastream/mac/MockRealtimeVideoSourceMac.mm:
>+        (WebCore::MockRealtimeVideoSourceMac::CMSampleBufferFromPixelBuffer):
>+        * platform/mediastream/mac/RealtimeIncomingVideoSourceCocoa.mm:
>+        * platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.cpp:
>+        (WebCore::ConvertToI420):
>+        (WebCore::RealtimeOutgoingVideoSourceCocoa::sampleBufferUpdated):
>+        * platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.h:
>+        * platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.mm: Added.
>+        (WebCore::RealtimeOutgoingVideoSourceCocoa::convertToYUV):
>+        * testing/MockLibWebRTCPeerConnection.cpp:
>+        (WebCore::MockLibWebRTCPeerConnection::AddTrack):
>+        * testing/MockLibWebRTCPeerConnection.h:
>+        (WebCore::MockRtpSender::GetParameters):
>+        (WebCore::MockRtpSender::SetParameters):
>+
> 2018-08-21  Brent Fulgham  <bfulgham@apple.com>
> 
>         Remove experimental affiliated domain code now that StorageAccess API is available
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCDataChannelHandler.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCDataChannelHandler.h
>index 899c4ee1ad9..50bf77624a4 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCDataChannelHandler.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCDataChannelHandler.h
>@@ -28,8 +28,14 @@
> 
> #include "LibWebRTCMacros.h"
> #include "RTCDataChannelHandler.h"
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/datachannelinterface.h>
> 
>+#pragma clang diagnostic pop
>+
> namespace WebCore {
> 
> class RTCDataChannelHandlerClient;
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>index d240ab0e5e7..743d18f0cb5 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>@@ -197,27 +197,31 @@ void LibWebRTCMediaEndpoint::addTrack(RTCRtpSender& sender, MediaStreamTrack& tr
> {
>     ASSERT(m_backend);
> 
>-    std::vector<webrtc::MediaStreamInterface*> mediaStreams;
>-    rtc::scoped_refptr<webrtc::MediaStreamInterface> mediaStream = nullptr;
>-    if (mediaStreamIds.size()) {
>-        // libwebrtc is only using the first one if any.
>-        mediaStream = m_peerConnectionFactory.CreateLocalMediaStream(mediaStreamIds[0].utf8().data());
>-        mediaStreams.push_back(mediaStream.get());
>+    String mediaStreamId = mediaStreamIds.isEmpty() ? createCanonicalUUIDString() : mediaStreamIds[0];
>+    rtc::scoped_refptr<webrtc::MediaStreamInterface> mediaStream = m_localStreams.get(mediaStreamId);
>+    if (!mediaStream) {
>+        mediaStream = m_peerConnectionFactory.CreateLocalMediaStream(mediaStreamId.utf8().data());
>+        m_backend->AddStream(mediaStream);
>+        m_localStreams.add(mediaStreamId, mediaStream);
>     }
>-    
>+
>+    std::vector<std::string> ids;
>+    for (auto& id : mediaStreamIds)
>+        ids.push_back(id.utf8().data());
>+
>     switch (track.privateTrack().type()) {
>     case RealtimeMediaSource::Type::Audio: {
>         auto trackSource = RealtimeOutgoingAudioSource::create(track.privateTrack());
>         auto audioTrack = m_peerConnectionFactory.CreateAudioTrack(track.id().utf8().data(), trackSource.ptr());
>         m_peerConnectionBackend.addAudioSource(WTFMove(trackSource));
>-        m_senders.add(&sender, m_backend->AddTrack(audioTrack.get(), WTFMove(mediaStreams)));
>+        m_senders.add(&sender, m_backend->AddTrack(audioTrack.get(), WTFMove(ids)).MoveValue());
>         return;
>     }
>     case RealtimeMediaSource::Type::Video: {
>         auto videoSource = RealtimeOutgoingVideoSource::create(track.privateTrack());
>         auto videoTrack = m_peerConnectionFactory.CreateVideoTrack(track.id().utf8().data(), videoSource.ptr());
>         m_peerConnectionBackend.addVideoSource(WTFMove(videoSource));
>-        m_senders.add(&sender, m_backend->AddTrack(videoTrack.get(), WTFMove(mediaStreams)));
>+        m_senders.add(&sender, m_backend->AddTrack(videoTrack.get(), WTFMove(ids)).MoveValue());
>         return;
>     }
>     case RealtimeMediaSource::Type::None:
>@@ -619,7 +623,7 @@ void LibWebRTCMediaEndpoint::OnSignalingChange(webrtc::PeerConnectionInterface::
> MediaStream& LibWebRTCMediaEndpoint::mediaStreamFromRTCStream(webrtc::MediaStreamInterface& rtcStream)
> {
>     auto mediaStream = m_streams.ensure(&rtcStream, [&rtcStream, this] {
>-        auto label = rtcStream.label();
>+        auto label = rtcStream.id();
>         auto stream = MediaStream::create(*m_peerConnectionBackend.connection().scriptExecutionContext(), MediaStreamPrivate::create({ }, fromStdString(label)));
>         auto streamPointer = stream.ptr();
>         m_peerConnectionBackend.addRemoteStream(WTFMove(stream));
>@@ -971,7 +975,8 @@ static inline RTCRtpParameters::EncodingParameters fillEncodingParameters(const
>     if (rtcParameters.max_framerate)
>         parameters.maxFramerate = *rtcParameters.max_framerate;
>     parameters.rid = fromStdString(rtcParameters.rid);
>-    parameters.scaleResolutionDownBy = rtcParameters.scale_resolution_down_by;
>+    if (rtcParameters.scale_resolution_down_by)
>+        parameters.scaleResolutionDownBy = *rtcParameters.scale_resolution_down_by;
> 
>     return parameters;
> }
>@@ -1013,6 +1018,8 @@ static RTCRtpParameters fillRtpParameters(const webrtc::RtpParameters rtcParamet
>         parameters.codecs.append(fillCodecParameters(codec));
> 
>     switch (rtcParameters.degradation_preference) {
>+    // FIXME: Support DegradationPreference::DISABLED.
>+    case webrtc::DegradationPreference::DISABLED:
>     case webrtc::DegradationPreference::MAINTAIN_FRAMERATE:
>         parameters.degradationPreference = RTCRtpParameters::DegradationPreference::MaintainFramerate;
>         break;
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>index 297b60c185e..d409a1b23dd 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>@@ -32,10 +32,17 @@
> #include "RealtimeOutgoingAudioSource.h"
> #include "RealtimeOutgoingVideoSource.h"
> #include <Timer.h>
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/jsep.h>
> #include <webrtc/api/peerconnectioninterface.h>
> #include <webrtc/pc/peerconnectionfactory.h>
> #include <webrtc/pc/rtcstatscollector.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <wtf/LoggerHelper.h>
> #include <wtf/ThreadSafeRefCounted.h>
> 
>@@ -129,8 +136,10 @@ private:
>     void AddRef() const { ref(); }
>     rtc::RefCountReleaseStatus Release() const
>     {
>+        auto result = refCount() - 1;
>         deref();
>-        return refCount() ? rtc::RefCountReleaseStatus::kDroppedLastRef : rtc::RefCountReleaseStatus::kOtherRefsRemained;
>+        return result ? rtc::RefCountReleaseStatus::kOtherRefsRemained
>+        : rtc::RefCountReleaseStatus::kDroppedLastRef;
>     }
> 
>     bool shouldOfferAllowToReceiveAudio() const;
>@@ -214,6 +223,8 @@ private:
>     bool m_isInitiator { false };
>     Timer m_statsLogTimer;
> 
>+    HashMap<String, rtc::scoped_refptr<webrtc::MediaStreamInterface>> m_localStreams;
>+
> #if !RELEASE_LOG_DISABLED
>     int64_t m_statsFirstDeliveredTimestamp { 0 };
>     Ref<const Logger> m_logger;
>diff --git a/Source/WebCore/WebCore.xcodeproj/project.pbxproj b/Source/WebCore/WebCore.xcodeproj/project.pbxproj
>index ee295f9763f..dc89e2ae35a 100644
>--- a/Source/WebCore/WebCore.xcodeproj/project.pbxproj
>+++ b/Source/WebCore/WebCore.xcodeproj/project.pbxproj
>@@ -1106,6 +1106,7 @@
> 		41885B9311B6FDA6003383BB /* FormSubmission.h in Headers */ = {isa = PBXBuildFile; fileRef = 41885B9111B6FDA6003383BB /* FormSubmission.h */; settings = {ATTRIBUTES = (Private, ); }; };
> 		418A06D0133C04D500CD379C /* EventDispatcher.h in Headers */ = {isa = PBXBuildFile; fileRef = 418A06CE133C04D500CD379C /* EventDispatcher.h */; };
> 		418F88050FF957AF0080F045 /* JSAbstractWorker.h in Headers */ = {isa = PBXBuildFile; fileRef = 418F88030FF957AE0080F045 /* JSAbstractWorker.h */; };
>+		419242492127B93E00634FCF /* RealtimeOutgoingVideoSourceCocoa.mm in Sources */ = {isa = PBXBuildFile; fileRef = 419242472127B7CC00634FCF /* RealtimeOutgoingVideoSourceCocoa.mm */; };
> 		419ACF921F97E7DA009F1A83 /* ServiceWorkerFetch.h in Headers */ = {isa = PBXBuildFile; fileRef = 419ACF8E1F97E7D5009F1A83 /* ServiceWorkerFetch.h */; settings = {ATTRIBUTES = (Private, ); }; };
> 		419BC2DF1685329900D64D6D /* VisitedLinkState.h in Headers */ = {isa = PBXBuildFile; fileRef = 419BC2DD1685329900D64D6D /* VisitedLinkState.h */; };
> 		419BE7591BC7F42B00E1C85B /* WebCoreBuiltinNames.h in Headers */ = {isa = PBXBuildFile; fileRef = 419BE7521BC7F3DB00E1C85B /* WebCoreBuiltinNames.h */; };
>@@ -7291,6 +7292,7 @@
> 		418C395F1C8F0AAB0051C8A3 /* ReadableStreamDefaultController.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ReadableStreamDefaultController.h; sourceTree = "<group>"; };
> 		418F88020FF957AE0080F045 /* JSAbstractWorker.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = JSAbstractWorker.cpp; sourceTree = "<group>"; };
> 		418F88030FF957AE0080F045 /* JSAbstractWorker.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = JSAbstractWorker.h; sourceTree = "<group>"; };
>+		419242472127B7CC00634FCF /* RealtimeOutgoingVideoSourceCocoa.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = RealtimeOutgoingVideoSourceCocoa.mm; sourceTree = "<group>"; };
> 		41945694203502A5004BA277 /* ApplicationCacheResourceLoader.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = ApplicationCacheResourceLoader.cpp; sourceTree = "<group>"; };
> 		41945696203502A6004BA277 /* ApplicationCacheResourceLoader.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ApplicationCacheResourceLoader.h; sourceTree = "<group>"; };
> 		419ACF8E1F97E7D5009F1A83 /* ServiceWorkerFetch.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ServiceWorkerFetch.h; sourceTree = "<group>"; };
>@@ -15335,6 +15337,7 @@
> 				41103AA71E39790A00769F14 /* RealtimeOutgoingAudioSourceCocoa.cpp */,
> 				41103AA81E39790A00769F14 /* RealtimeOutgoingAudioSourceCocoa.h */,
> 				5CDD833B1E4324BB00621B83 /* RealtimeOutgoingVideoSourceCocoa.cpp */,
>+				419242472127B7CC00634FCF /* RealtimeOutgoingVideoSourceCocoa.mm */,
> 				5CDD833C1E4324BB00621B83 /* RealtimeOutgoingVideoSourceCocoa.h */,
> 				070A9F5E1FFECC70003DF649 /* ScreenDisplayCaptureSourceMac.h */,
> 				070A9F601FFECC71003DF649 /* ScreenDisplayCaptureSourceMac.mm */,
>@@ -31749,6 +31752,7 @@
> 				538EC8A11F993F9D004D22A8 /* UnifiedSource48.cpp in Sources */,
> 				DE5F85991FA1ABF4006DB63A /* UnifiedSource49-mm.mm in Sources */,
> 				538EC8A21F993F9D004D22A8 /* UnifiedSource49.cpp in Sources */,
>+				419242492127B93E00634FCF /* RealtimeOutgoingVideoSourceCocoa.mm in Sources */,
> 				DE5F859A1FA1ABF4006DB63A /* UnifiedSource50-mm.mm in Sources */,
> 				538EC8A31F993F9D004D22A8 /* UnifiedSource50.cpp in Sources */,
> 				DE5F86501FA2AF24006DB63A /* UnifiedSource51-mm.mm in Sources */,
>diff --git a/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.h b/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.h
>index 4c363b62daa..cbf5730ac94 100644
>--- a/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.h
>+++ b/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.h
>@@ -34,7 +34,14 @@
> 
> #include "LibWebRTCMacros.h"
> #include "RealtimeMediaSource.h"
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/mediastreaminterface.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <wtf/RetainPtr.h>
> 
> namespace WebCore {
>diff --git a/Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.h b/Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.h
>index 85fbfa7e122..4d7cad29a1a 100644
>--- a/Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.h
>+++ b/Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.h
>@@ -34,7 +34,14 @@
> 
> #include "LibWebRTCMacros.h"
> #include "RealtimeMediaSource.h"
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/mediastreaminterface.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <wtf/RetainPtr.h>
> 
> namespace WebCore {
>diff --git a/Source/WebCore/platform/mediastream/RealtimeMediaSource.cpp b/Source/WebCore/platform/mediastream/RealtimeMediaSource.cpp
>index 8a4609a2d26..bfe4d6de1c8 100644
>--- a/Source/WebCore/platform/mediastream/RealtimeMediaSource.cpp
>+++ b/Source/WebCore/platform/mediastream/RealtimeMediaSource.cpp
>@@ -286,7 +286,7 @@ bool RealtimeMediaSource::supportsSizeAndFrameRate(std::optional<IntConstraint>
>             badConstraint = frameRateConstraint->name();
>         return false;
>     }
>-
>+    
>     return true;
> }
> 
>@@ -754,7 +754,7 @@ bool RealtimeMediaSource::supportsConstraint(const MediaConstraint& constraint)
>         // Unknown (or unsupported) constraints should be ignored.
>         break;
>     }
>-
>+    
>     return false;
> }
> 
>@@ -765,7 +765,7 @@ bool RealtimeMediaSource::supportsConstraints(const MediaConstraints& constraint
>     FlattenedConstraint candidates;
>     if (!selectSettings(constraints, candidates, invalidConstraint, SelectType::ForSupportsConstraints))
>         return false;
>-
>+    
>     return true;
> }
> 
>diff --git a/Source/WebCore/platform/mediastream/RealtimeOutgoingAudioSource.h b/Source/WebCore/platform/mediastream/RealtimeOutgoingAudioSource.h
>index 76a61f8f644..ce2e5b1d9e0 100644
>--- a/Source/WebCore/platform/mediastream/RealtimeOutgoingAudioSource.h
>+++ b/Source/WebCore/platform/mediastream/RealtimeOutgoingAudioSource.h
>@@ -34,7 +34,14 @@
> #include "LibWebRTCMacros.h"
> #include "MediaStreamTrackPrivate.h"
> #include "Timer.h"
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/mediastreaminterface.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <wtf/ThreadSafeRefCounted.h>
> 
> namespace webrtc {
>diff --git a/Source/WebCore/platform/mediastream/RealtimeOutgoingVideoSource.h b/Source/WebCore/platform/mediastream/RealtimeOutgoingVideoSource.h
>index 029d1478274..c1aa9d11395 100644
>--- a/Source/WebCore/platform/mediastream/RealtimeOutgoingVideoSource.h
>+++ b/Source/WebCore/platform/mediastream/RealtimeOutgoingVideoSource.h
>@@ -33,10 +33,15 @@
> #include "LibWebRTCMacros.h"
> #include "MediaStreamTrackPrivate.h"
> #include <Timer.h>
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/mediastreaminterface.h>
>-#include <webrtc/api/optional.h>
> #include <webrtc/common_video/include/i420_buffer_pool.h>
>-#include <webrtc/api/videosinkinterface.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <wtf/Optional.h>
> #include <wtf/ThreadSafeRefCounted.h>
> 
>@@ -54,9 +59,8 @@ public:
>     void AddRef() const final { ref(); }
>     rtc::RefCountReleaseStatus Release() const final
>     {
>-        auto result = hasOneRef() ? rtc::RefCountReleaseStatus::kDroppedLastRef : rtc::RefCountReleaseStatus::kOtherRefsRemained;
>         deref();
>-        return result;
>+        return rtc::RefCountReleaseStatus::kOtherRefsRemained;
>     }
> 
>     void setApplyRotation(bool shouldApplyRotation) { m_shouldApplyRotation = shouldApplyRotation; }
>@@ -88,7 +92,7 @@ private:
> 
>     // VideoTrackSourceInterface API
>     bool is_screencast() const final { return false; }
>-    rtc::Optional<bool> needs_denoising() const final { return rtc::Optional<bool>(); }
>+    absl::optional<bool> needs_denoising() const final { return absl::optional<bool>(); }
>     bool GetStats(Stats*) final { return false; };
> 
>     // MediaSourceInterface API
>diff --git a/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp b/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp
>index 0ec2f754e7c..096ded08022 100644
>--- a/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp
>+++ b/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp
>@@ -143,7 +143,6 @@ public:
> 
>     int32_t Decode(const webrtc::EncodedImage& inputImage,
>         bool,
>-        const webrtc::RTPFragmentationHeader*,
>         const webrtc::CodecSpecificInfo*,
>         int64_t renderTimeMs) override
>     {
>@@ -233,7 +232,7 @@ public:
>         GST_LOG_OBJECT(pipeline(), "Output decoded frame! %d -> %" GST_PTR_FORMAT,
>             frame->timestamp(), buffer);
> 
>-        m_imageReadyCb->Decoded(*frame.get(), rtc::Optional<int32_t>(), rtc::Optional<uint8_t>());
>+        m_imageReadyCb->Decoded(*frame.get(), absl::optional<int32_t>(), absl::optional<uint8_t>());
> 
>         return GST_FLOW_OK;
>     }
>diff --git a/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp b/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp
>index 21649c25664..8037b93b79b 100644
>--- a/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp
>+++ b/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp
>@@ -61,15 +61,13 @@ static GRefPtr<GRegex> bitrateKBitPerSec;
> class GStreamerVideoEncoder : public webrtc::VideoEncoder {
> public:
>     GStreamerVideoEncoder(const webrtc::SdpVideoFormat&)
>-        : m_pictureId(0)
>-        , m_firstFramePts(GST_CLOCK_TIME_NONE)
>+        : m_firstFramePts(GST_CLOCK_TIME_NONE)
>         , m_restrictionCaps(adoptGRef(gst_caps_new_empty_simple("video/x-raw")))
>         , m_bitrateSetter(nullptr)
>     {
>     }
>     GStreamerVideoEncoder()
>-        : m_pictureId(0)
>-        , m_firstFramePts(GST_CLOCK_TIME_NONE)
>+        : m_firstFramePts(GST_CLOCK_TIME_NONE)
>         , m_restrictionCaps(adoptGRef(gst_caps_new_empty_simple("video/x-raw")))
>         , m_bitrateSetter(nullptr)
>     {
>@@ -241,7 +239,6 @@ public:
>         PopulateCodecSpecific(&codecSpecifiInfos, buffer);
> 
>         webrtc::EncodedImageCallback::Result result = m_imageReadyCb->OnEncodedImage(frame, &codecSpecifiInfos, fragmentationInfo);
>-        m_pictureId = (m_pictureId + 1) & 0x7FFF;
>         if (result.error != webrtc::EncodedImageCallback::Result::OK) {
>             GST_ELEMENT_ERROR(m_pipeline.get(), LIBRARY, FAILED, (nullptr),
>                 ("Encode callback failed: %d", result.error));
>@@ -391,9 +388,6 @@ public:
>         m_restrictionCaps = caps;
>     }
> 
>-protected:
>-    int16_t m_pictureId;
>-
> private:
>     static GstFlowReturn newSampleCallbackTramp(GstElement* sink, GStreamerVideoEncoder* enc)
>     {
>@@ -532,12 +526,10 @@ public:
>         codecSpecifiInfos->codec_name = ImplementationName();
>         webrtc::CodecSpecificInfoVP8* vp8Info = &(codecSpecifiInfos->codecSpecific.VP8);
>         vp8Info->temporalIdx = 0;
>-        vp8Info->pictureId = m_pictureId;
> 
>         vp8Info->simulcastIdx = 0;
>         vp8Info->keyIdx = webrtc::kNoKeyIdx;
>         vp8Info->nonReference = GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DELTA_UNIT);
>-        vp8Info->tl0PicIdx = webrtc::kNoTl0PicIdx;
>     }
> };
> 
>diff --git a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h
>index 9875c4fa61d..71ca2ab5dba 100644
>--- a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h
>+++ b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h
>@@ -28,9 +28,15 @@
> #if USE(LIBWEBRTC)
> 
> #include "LibWebRTCMacros.h"
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
>+#include <webrtc/modules/audio_device/include/audio_device.h>
> #include <webrtc/rtc_base/messagehandler.h>
> #include <webrtc/rtc_base/thread.h>
>-#include <webrtc/modules/audio_device/include/audio_device.h>
>+
>+#pragma clang diagnostic pop
> 
> namespace WebCore {
> 
>@@ -47,7 +53,7 @@ private:
>         return value;
>     }
> 
>-    void AddRef() const final { return; }
>+    void AddRef() const final { }
>     rtc::RefCountReleaseStatus Release() const final { return rtc::RefCountReleaseStatus::kOtherRefsRemained; }
>     void OnMessage(rtc::Message*);
> 
>diff --git a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCMacros.h b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCMacros.h
>index 938e6631f8d..ddf31f8f640 100644
>--- a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCMacros.h
>+++ b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCMacros.h
>@@ -36,6 +36,7 @@
> 
> #if PLATFORM(COCOA)
> #define WEBRTC_MAC
>+#define ABSL_ALLOCATOR_NOTHROW 1
> #endif
> 
> #define WEBRTC_POSIX 1
>diff --git a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp
>index f2363ba0ce4..b4bd0a45138 100644
>--- a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp
>+++ b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp
>@@ -30,6 +30,10 @@
> #include "LibWebRTCAudioModule.h"
> #include "Logging.h"
> #include <dlfcn.h>
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/audio_codecs/builtin_audio_decoder_factory.h>
> #include <webrtc/api/audio_codecs/builtin_audio_encoder_factory.h>
> #include <webrtc/api/peerconnectionfactoryproxy.h>
>@@ -38,6 +42,9 @@
> #include <webrtc/p2p/client/basicportallocator.h>
> #include <webrtc/pc/peerconnectionfactory.h>
> #include <webrtc/rtc_base/physicalsocketserver.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <wtf/Function.h>
> #include <wtf/NeverDestroyed.h>
> #endif
>diff --git a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h
>index 0b4b9bd5b3a..d8c4069b29b 100644
>--- a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h
>+++ b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h
>@@ -35,12 +35,15 @@
> 
> #if USE(LIBWEBRTC)
> 
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/peerconnectioninterface.h>
>-#include <webrtc/api/video_codecs/video_decoder_factory.h>
> #include <webrtc/api/video_codecs/video_encoder_factory.h>
>+#include <webrtc/api/video_codecs/video_decoder_factory.h>
> #include <webrtc/rtc_base/scoped_ref_ptr.h>
>-#include <webrtc/media/engine/webrtcvideodecoderfactory.h>
>-#include <webrtc/media/engine/webrtcvideoencoderfactory.h>
>+
>+#pragma clang diagnostic pop
> 
> namespace rtc {
> class NetworkManager;
>diff --git a/Source/WebCore/platform/mediastream/mac/MockRealtimeVideoSourceMac.mm b/Source/WebCore/platform/mediastream/mac/MockRealtimeVideoSourceMac.mm
>index 2ad4d0133e9..7e7ca7d72e6 100644
>--- a/Source/WebCore/platform/mediastream/mac/MockRealtimeVideoSourceMac.mm
>+++ b/Source/WebCore/platform/mediastream/mac/MockRealtimeVideoSourceMac.mm
>@@ -70,7 +70,7 @@ RetainPtr<CMSampleBufferRef> MockRealtimeVideoSourceMac::CMSampleBufferFromPixel
> {
>     if (!pixelBuffer)
>         return nullptr;
>-
>+    
>     if (!m_pixelBufferConformer)
>         m_pixelBufferConformer = std::make_unique<PixelBufferConformerCV>((__bridge CFDictionaryRef)@{ (__bridge NSString *)kCVPixelBufferPixelFormatTypeKey: @(kCVPixelFormatType_420YpCbCr8Planar) });
> 
>diff --git a/Source/WebCore/platform/mediastream/mac/RealtimeIncomingVideoSourceCocoa.mm b/Source/WebCore/platform/mediastream/mac/RealtimeIncomingVideoSourceCocoa.mm
>index 9e201eb58f9..4b425f9ba62 100644
>--- a/Source/WebCore/platform/mediastream/mac/RealtimeIncomingVideoSourceCocoa.mm
>+++ b/Source/WebCore/platform/mediastream/mac/RealtimeIncomingVideoSourceCocoa.mm
>@@ -33,7 +33,14 @@
> #include "Logging.h"
> #include "MediaSampleAVFObjC.h"
> #include <pal/cf/CoreMediaSoftLink.h>
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/sdk/WebKit/WebKitUtilities.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <wtf/cf/TypeCastsCF.h>
> 
> #include "CoreVideoSoftLink.h"
>diff --git a/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.cpp b/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.cpp
>index 439a1bf1579..cf25fd83f1f 100644
>--- a/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.cpp
>+++ b/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.cpp
>@@ -29,10 +29,17 @@
> #if USE(LIBWEBRTC)
> 
> #include "Logging.h"
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/video/i420_buffer.h>
> #include <webrtc/common_video/libyuv/include/webrtc_libyuv.h>
> #include <webrtc/sdk/WebKit/WebKitUtilities.h>
> #include <webrtc/sdk/objc/Framework/Classes/Video/corevideo_frame_buffer.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <pal/cf/CoreMediaSoftLink.h>
> #include "CoreVideoSoftLink.h"
> 
>@@ -50,7 +57,8 @@ typedef enum RotationMode {
>   kRotateCounterClockwise = 270,
> } RotationModeEnum;
> 
>-int ConvertToI420(const uint8_t* src_frame,
>+int ConvertToI420(webrtc::VideoType src_video_type,
>+                  const uint8_t* src_frame,
>                   size_t src_size,
>                   uint8_t* dst_y,
>                   int dst_stride_y,
>@@ -64,8 +72,7 @@ int ConvertToI420(const uint8_t* src_frame,
>                   int src_height,
>                   int crop_width,
>                   int crop_height,
>-                  RotationMode rotation,
>-                  uint32_t format);
>+                  RotationMode rotation);
> }
> }
> 
>@@ -103,7 +110,7 @@ static inline int ConvertToI420(webrtc::VideoType src_video_type,
>   if (rotation == libyuv::kRotate90 || rotation == libyuv::kRotate270) {
>     std::swap(dst_width, dst_height);
>   }
>-  return libyuv::ConvertToI420(
>+  return libyuv::ConvertToI420(src_video_type,
>       src_frame, sample_size,
>       dst_buffer->MutableDataY(), dst_buffer->StrideY(),
>       dst_buffer->MutableDataU(), dst_buffer->StrideU(),
>@@ -111,8 +118,7 @@ static inline int ConvertToI420(webrtc::VideoType src_video_type,
>       crop_x, crop_y,
>       src_width, src_height,
>       dst_width, dst_height,
>-      rotation,
>-      webrtc::ConvertVideoType(src_video_type));
>+      rotation);
> }
> 
> void RealtimeOutgoingVideoSourceCocoa::sampleBufferUpdated(MediaStreamTrackPrivate&, MediaSample& sample)
>@@ -147,41 +153,23 @@ void RealtimeOutgoingVideoSourceCocoa::sampleBufferUpdated(MediaStreamTrackPriva
>     auto pixelBuffer = static_cast<CVPixelBufferRef>(CMSampleBufferGetImageBuffer(sample.platformSample().sample.cmSampleBuffer));
>     auto pixelFormatType = CVPixelBufferGetPixelFormatType(pixelBuffer);
> 
>-    if (pixelFormatType == kCVPixelFormatType_420YpCbCr8Planar || pixelFormatType == kCVPixelFormatType_420YpCbCr8BiPlanarFullRange) {
>-        rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer = webrtc::pixelBufferToFrame(pixelBuffer);
>-        if (m_shouldApplyRotation && m_currentRotation != webrtc::kVideoRotation_0) {
>-            // FIXME: We should make AVVideoCaptureSource handle the rotation whenever possible.
>-            // This implementation is inefficient, we should rotate on the CMSampleBuffer directly instead of doing this double allocation.
>-            auto rotatedBuffer = buffer->ToI420();
>-            ASSERT(rotatedBuffer);
>-            buffer = webrtc::I420Buffer::Rotate(*rotatedBuffer, m_currentRotation);
>-        }
>-        sendFrame(WTFMove(buffer));
>-        return;
>+    rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer;
>+    RetainPtr<CVPixelBufferRef> convertedBuffer;
>+    if (pixelFormatType == kCVPixelFormatType_420YpCbCr8Planar || pixelFormatType == kCVPixelFormatType_420YpCbCr8BiPlanarFullRange)
>+        buffer = webrtc::pixelBufferToFrame(pixelBuffer);
>+    else {
>+        convertedBuffer = convertToYUV(pixelBuffer);
>+        buffer = webrtc::pixelBufferToFrame(convertedBuffer.get());
>     }
> 
>-    CVPixelBufferLockBaseAddress(pixelBuffer, 0);
>-    auto* source = reinterpret_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(pixelBuffer, 0));
>-
>-    ASSERT(m_width);
>-    ASSERT(m_height);
>-
>-    auto newBuffer = m_bufferPool.CreateBuffer(m_width, m_height);
>-    ASSERT(newBuffer);
>-    if (!newBuffer) {
>-        RELEASE_LOG(WebRTC, "RealtimeOutgoingVideoSourceCocoa::videoSampleAvailable unable to allocate buffer for conversion to YUV");
>-        return;
>-    }
>-    if (pixelFormatType == kCVPixelFormatType_32BGRA)
>-        ConvertToI420(webrtc::VideoType::kARGB, source, 0, 0, m_width, m_height, 0, libyuv::kRotate0, newBuffer);
>-    else {
>-        ASSERT(pixelFormatType == kCVPixelFormatType_32ARGB);
>-        ConvertToI420(webrtc::VideoType::kBGRA, source, 0, 0, m_width, m_height, 0, libyuv::kRotate0, newBuffer);
>+    if (m_shouldApplyRotation && m_currentRotation != webrtc::kVideoRotation_0) {
>+        // FIXME: We should make AVVideoCaptureSource handle the rotation whenever possible.
>+        // This implementation is inefficient, we should rotate on the CMSampleBuffer directly instead of doing this double allocation.
>+        auto rotatedBuffer = buffer->ToI420();
>+        ASSERT(rotatedBuffer);
>+        buffer = webrtc::I420Buffer::Rotate(*rotatedBuffer, m_currentRotation);
>     }
>-    CVPixelBufferUnlockBaseAddress(pixelBuffer, 0);
>-    if (m_shouldApplyRotation && m_currentRotation != webrtc::kVideoRotation_0)
>-        newBuffer = webrtc::I420Buffer::Rotate(*newBuffer, m_currentRotation);
>-    sendFrame(WTFMove(newBuffer));
>+    sendFrame(WTFMove(buffer));
> }
> 
> 
>diff --git a/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.h b/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.h
>index 77bef461660..961d512d7b4 100644
>--- a/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.h
>+++ b/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.h
>@@ -28,6 +28,7 @@
> #if USE(LIBWEBRTC)
> 
> #include "RealtimeOutgoingVideoSource.h"
>+#include "PixelBufferConformerCV.h"
> 
> namespace WebCore {
> 
>@@ -41,6 +42,10 @@ private:
>     // MediaStreamTrackPrivate::Observer API
>     void sampleBufferUpdated(MediaStreamTrackPrivate&, MediaSample&) final;
> 
>+    RetainPtr<CVPixelBufferRef> convertToYUV(CVPixelBufferRef);
>+
>+    std::unique_ptr<PixelBufferConformerCV> m_pixelBufferConformer;
>+
> #if !RELEASE_LOG_DISABLED
>     size_t m_numberOfFrames { 0 };
> #endif
>diff --git a/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.mm b/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.mm
>new file mode 100644
>index 00000000000..881f5711fa0
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.mm
>@@ -0,0 +1,50 @@
>+/*
>+ * Copyright (C) 2018 Apple Inc.
>+ *
>+ * Redistribution and use in source and binary forms, with or without
>+ * modification, are permitted, provided that the following conditions
>+ * are required to be 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. AND 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.
>+ */
>+
>+#import "config.h"
>+#import "RealtimeOutgoingVideoSourceCocoa.h"
>+
>+#if USE(LIBWEBRTC)
>+
>+#import "PixelBufferConformerCV.h"
>+#import <pal/cf/CoreMediaSoftLink.h>
>+#import "CoreVideoSoftLink.h"
>+
>+namespace WebCore {
>+
>+RetainPtr<CVPixelBufferRef> RealtimeOutgoingVideoSourceCocoa::convertToYUV(CVPixelBufferRef pixelBuffer)
>+{
>+    if (!pixelBuffer)
>+        return nullptr;
>+
>+    if (!m_pixelBufferConformer)
>+        m_pixelBufferConformer = std::make_unique<PixelBufferConformerCV>((__bridge CFDictionaryRef)@{ (__bridge NSString *)kCVPixelBufferPixelFormatTypeKey: @(kCVPixelFormatType_420YpCbCr8Planar) });
>+
>+    return m_pixelBufferConformer->convert(pixelBuffer);
>+}
>+
>+} // namespace WebCore
>+
>+#endif // USE(LIBWEBRTC)
>diff --git a/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp b/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp
>index b5d0de1c213..35a121f8d1b 100644
>--- a/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp
>+++ b/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp
>@@ -239,17 +239,17 @@ rtc::scoped_refptr<webrtc::DataChannelInterface> MockLibWebRTCPeerConnection::Cr
>     return new rtc::RefCountedObject<MockLibWebRTCDataChannel>(std::string(label), parameters.ordered, parameters.reliable, parameters.id);
> }
> 
>-rtc::scoped_refptr<webrtc::RtpSenderInterface> MockLibWebRTCPeerConnection::AddTrack(webrtc::MediaStreamTrackInterface* track, std::vector<webrtc::MediaStreamInterface*> streams)
>+webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpSenderInterface>> MockLibWebRTCPeerConnection::AddTrack(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track, const std::vector<std::string>& streamIds)
> {
>     LibWebRTCProvider::callOnWebRTCSignalingThread([observer = &m_observer] {
>         observer->OnRenegotiationNeeded();
>     });
> 
>-    if (streams.size())
>-        m_streamLabel = streams.front()->label();
>+    if (!streamIds.empty())
>+        m_streamLabel = streamIds.front();
> 
>-    m_senders.append(new rtc::RefCountedObject<MockRtpSender>(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>(track)));
>-    return m_senders.last().get();
>+    m_senders.append(new rtc::RefCountedObject<MockRtpSender>(WTFMove(track)));
>+    return rtc::scoped_refptr<webrtc::RtpSenderInterface>(m_senders.last().get());
> }
> 
> bool MockLibWebRTCPeerConnection::RemoveTrack(webrtc::RtpSenderInterface* sender)
>diff --git a/Source/WebCore/testing/MockLibWebRTCPeerConnection.h b/Source/WebCore/testing/MockLibWebRTCPeerConnection.h
>index 2b7e0dcd94f..9a80da48fee 100644
>--- a/Source/WebCore/testing/MockLibWebRTCPeerConnection.h
>+++ b/Source/WebCore/testing/MockLibWebRTCPeerConnection.h
>@@ -27,8 +27,15 @@
> #if USE(LIBWEBRTC)
> 
> #include "LibWebRTCMacros.h"
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/mediastreaminterface.h>
> #include <webrtc/api/peerconnectioninterface.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <wtf/text/WTFString.h>
> 
> namespace WebCore {
>@@ -49,11 +56,9 @@ protected:
> private:
>     rtc::scoped_refptr<webrtc::StreamCollectionInterface> local_streams() override { return nullptr; }
>     rtc::scoped_refptr<webrtc::StreamCollectionInterface> remote_streams() override { return nullptr; }
>-    rtc::scoped_refptr<webrtc::DtmfSenderInterface> CreateDtmfSender(webrtc::AudioTrackInterface*) override { return nullptr; }
>     const webrtc::SessionDescriptionInterface* local_description() const override { return nullptr; }
>     const webrtc::SessionDescriptionInterface* remote_description() const override { return nullptr; }
>     bool AddIceCandidate(const webrtc::IceCandidateInterface*) override { return true; }
>-    void RegisterUMAObserver(webrtc::UMAObserver*) override { }
>     SignalingState signaling_state() override { return kStable; }
>     IceConnectionState ice_connection_state() override { return kIceConnectionNew; }
>     IceGatheringState ice_gathering_state() override { return kIceGatheringNew; }
>@@ -67,7 +72,7 @@ protected:
>     void SetRemoteDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) final;
>     void CreateAnswer(webrtc::CreateSessionDescriptionObserver*, const webrtc::MediaConstraintsInterface*) final;
>     rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel(const std::string&, const webrtc::DataChannelInit*) final;
>-    rtc::scoped_refptr<webrtc::RtpSenderInterface> AddTrack(webrtc::MediaStreamTrackInterface*, std::vector<webrtc::MediaStreamInterface*> streams) final;
>+    webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpSenderInterface>> AddTrack(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>, const std::vector<std::string>& streams) final;
>     bool RemoveTrack(webrtc::RtpSenderInterface*) final;
>     webrtc::RTCError SetBitrate(const BitrateParameters&) final { return { }; }
> 
>@@ -221,8 +226,8 @@ public:
>     cricket::MediaType media_type() const { return cricket::MEDIA_TYPE_VIDEO; }
>     std::string id() const { return ""; }
>     std::vector<std::string> stream_ids() const { return { }; }
>-    webrtc::RtpParameters GetParameters() const { return { }; }
>-    bool SetParameters(const webrtc::RtpParameters&) { return false; }
>+    webrtc::RtpParameters GetParameters() { return { }; }
>+    webrtc::RTCError SetParameters(const webrtc::RtpParameters&) { return { }; }
>     rtc::scoped_refptr<webrtc::DtmfSenderInterface> GetDtmfSender() const { return nullptr; }
> 
> private:
>@@ -245,7 +250,6 @@ private:
> 
>     void SetOptions(const Options&) final { }
>     rtc::scoped_refptr<webrtc::AudioSourceInterface> CreateAudioSource(const cricket::AudioOptions&) final { return nullptr; }
>-    rtc::scoped_refptr<webrtc::AudioSourceInterface> CreateAudioSource(const webrtc::MediaConstraintsInterface*) final { return nullptr; }
> 
>     rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> CreateVideoSource(cricket::VideoCapturer*) final { return nullptr; }
>     rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> CreateVideoSource(cricket::VideoCapturer*, const webrtc::MediaConstraintsInterface*) final { return nullptr; }

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