Refresh libwebrtc up to 343f4144be
authoryouenn@apple.com <youenn@apple.com@268f45cc-cd09-0410-ab3c-d52691b4dbfc>
Fri, 12 Oct 2018 20:45:16 +0000 (20:45 +0000)
committeryouenn@apple.com <youenn@apple.com@268f45cc-cd09-0410-ab3c-d52691b4dbfc>
Fri, 12 Oct 2018 20:45:16 +0000 (20:45 +0000)
    https://bugs.webkit.org/show_bug.cgi?id=190361

    Reviewed by Chris Dumez.

    This refresh should not change behavior but will allow us to implement better
    mdns candidate support and simulcast activation surfacing through addTransceiver.
    Covered by existing tests.

    * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
    (WebCore::LibWebRTCMediaEndpoint::doCreateAnswer):
    * testing/MockLibWebRTCPeerConnection.cpp:
    (WebCore::MockLibWebRTCPeerConnection::CreateAnswer):
    * testing/MockLibWebRTCPeerConnection.h:
    (WebCore::MockRtpSender::GetParameters): Deleted.
    (WebCore::MockRtpSender::SetParameters): Deleted.
    (WebCore::MockRtpSender::GetDtmfSender const): Deleted.

ThirdParty/libwebrtc

    * Configurations/libwebrtc.iOS.exp:
    * Configurations/libwebrtc.iOSsim.exp:
    * Configurations/libwebrtc.mac.exp:
    * Configurations/libwebrtc.xcconfig:
    * Source/webrtc: Resynced.
    * WebKit/0001-Updating-webrtc.patch: Removed.
    * libwebrtc.xcodeproj/project.pbxproj:

git-svn-id: https://svn.webkit.org/repository/webkit/trunk@237075 268f45cc-cd09-0410-ab3c-d52691b4dbfc

2142 files changed:
Source/ThirdParty/libwebrtc/CMakeLists.txt
Source/ThirdParty/libwebrtc/ChangeLog
Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp
Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp
Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp
Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig
Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view.h
Source/ThirdParty/libwebrtc/Source/webrtc/.gitignore
Source/ThirdParty/libwebrtc/Source/webrtc/AUTHORS
Source/ThirdParty/libwebrtc/Source/webrtc/BUILD.gn
Source/ThirdParty/libwebrtc/Source/webrtc/DEPS
Source/ThirdParty/libwebrtc/Source/webrtc/ENG_REVIEW_OWNERS [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/OWNERS
Source/ThirdParty/libwebrtc/Source/webrtc/PRESUBMIT.py
Source/ThirdParty/libwebrtc/Source/webrtc/WATCHLISTS
Source/ThirdParty/libwebrtc/Source/webrtc/abseil-in-webrtc.md [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/BUILD.gn
Source/ThirdParty/libwebrtc/Source/webrtc/api/asyncresolverfactory.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_frame.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/BUILD.gn
Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/crypto/framedecryptorinterface.h [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/crypto/frameencryptorinterface.h [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/dtmfsenderinterface.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/media_transport_interface.h [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/ortcfactoryinterface.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionfactoryproxy.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionproxy.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/rtcerror.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/rtcerror.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/rtcerror_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_headers.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_headers.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpreceiverinterface.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpreceiverinterface.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpsenderinterface.cc [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpsenderinterface.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats_objects.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_video_quality_test_fixture.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_video_quality_test_fixture.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_decryptor.cc [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_decryptor.h [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_encryptor.cc [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_encryptor.h [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_peerconnectioninterface.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_rtpsender.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/neteq_simulator.cc [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/neteq_simulator.h [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/neteq_simulator_factory.cc [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/neteq_simulator_factory.h [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/simulated_network.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/test_dependency_factory.cc [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/test_dependency_factory.h [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video_quality_test_fixture.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/videocodec_test_fixture.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/videocodec_test_stats.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/test/videocodec_test_stats.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_control.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/video/BUILD.gn
Source/ThirdParty/libwebrtc/Source/webrtc/api/video/encoded_frame.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/video/encoded_frame.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocator_factory.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame_marking.h [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_observer.cc [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_observer.h
Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/BUILD.gn
Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/BUILD.gn
Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder_factory.cc [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder_factory.h
Source/ThirdParty/libwebrtc/Source/webrtc/audio/BUILD.gn
Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.cc
Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.h
Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.cc
Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.h
Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_state.cc
Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_state.h
Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_transport_impl.cc
Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel.cc [deleted file]
Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel.h [deleted file]
Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_proxy.cc [deleted file]
Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_proxy.h [deleted file]
Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive.cc [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive.h [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive_proxy.cc [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive_proxy.h [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send.cc [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send.h [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send_proxy.cc [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send_proxy.h [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/audio/mock_voe_channel_proxy.h
Source/ThirdParty/libwebrtc/Source/webrtc/audio/null_audio_poller.h
Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.cc
Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.h
Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.cc
Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.h
Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_stats_test.cc
Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/low_bandwidth_audio_test.cc
Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/low_bandwidth_audio_test.py
Source/ThirdParty/libwebrtc/Source/webrtc/audio/transport_feedback_packet_loss_tracker.cc
Source/ThirdParty/libwebrtc/Source/webrtc/audio/transport_feedback_packet_loss_tracker.h
Source/ThirdParty/libwebrtc/Source/webrtc/audio/transport_feedback_packet_loss_tracker_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/BUILD.gn
Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_receive_stream.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_estimator_tests.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/call.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/call.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/call_config.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/call_config.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/call_perf_tests.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/call_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/callfactory.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/degraded_call.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/degraded_call.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_network_pipe.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_network_pipe.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream_impl.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/packet_receiver.cc [deleted file]
Source/ThirdParty/libwebrtc/Source/webrtc/call/packet_receiver.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send_interface.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender_interface.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/rtx_receive_stream.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/simulated_network.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/simulated_packet_receiver.h [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/call/test/fake_network_pipe_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_rtp_transport_controller_send.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/video_receive_stream.cc
Source/ThirdParty/libwebrtc/Source/webrtc/call/video_receive_stream.h
Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.h
Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/BUILD.gn
Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_ring_buffer.cc [deleted file]
Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_ring_buffer.h [deleted file]
Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_ring_buffer_unittest.cc [deleted file]
Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/blocker.cc [deleted file]
Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/blocker.h [deleted file]
Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/blocker_unittest.cc [deleted file]
Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/lapped_transform.cc [deleted file]
Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/lapped_transform.h [deleted file]
Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/lapped_transform_unittest.cc [deleted file]
Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/include/push_resampler.h
Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_resampler.cc
Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_resampler_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/resampler_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/ring_buffer.h
Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/common_types.h
Source/ThirdParty/libwebrtc/Source/webrtc/common_video/BUILD.gn
Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.cc
Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser.cc
Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser.h
Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_bitrate_allocator.h [deleted file]
Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame.h
Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.cc
Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.h
Source/ThirdParty/libwebrtc/Source/webrtc/data/common_video/jpeg/webrtc_logo.jpg [deleted file]
Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/H263Foreman_CIF_Iframe.bin [deleted file]
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Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/rtpplay.exe [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/examples/BUILD.gn
Source/ThirdParty/libwebrtc/Source/webrtc/examples/DEPS
Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java
Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCProximitySensor.java
Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java
Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CaptureQualityController.java
Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/ConnectActivity.java
Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/RecordedAudioToFileController.java
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Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/video_quality_loopback_test.py
Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient+Internal.h
Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient.h
Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient.m
Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.m
Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDCaptureController.h
Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDCaptureController.m
Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.m
Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsModel+Private.h
Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsModel.h
Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsModel.m
Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.h
Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.m
Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.m
Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.m
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Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtp_frame_reference_finder.h
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Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/session_info.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/frame_dropper.cc
Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/frame_dropper.h
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Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/framerate_controller.h [new file with mode: 0644]
Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/framerate_controller_unittest.cc [new file with mode: 0644]
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Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/ivf_file_writer_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h [deleted file]
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Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator.h
Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_test_fixture_impl.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_utility.h
Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer.cc
Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.cc
Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.h
Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_packet_buffer_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.h
Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_crypto_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_histogram_unittest.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rampup_tests.cc
Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rtp_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionendtoend_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion_unittest.cc
Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsenderreceiver_unittest.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakepeerconnectionbase.h
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Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_rtpsenderinternal.h
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Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis.h
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Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/HardwareVideoDecoderFactory.java
Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/HardwareVideoEncoderFactory.java
Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaCodecVideoDecoder.java