Branch: refs/heads/main
Home: https://github.com/WebKit/WebKit
Commit: 0f9939d61b9156fd173d7433f36fdfa95e47c632
https://github.com/WebKit/WebKit/commit/0f9939d61b9156fd173d7433f36fdfa95e47c632
Author: Youenn Fablet <[email protected]>
Date: 2025-12-05 (Fri, 05 Dec 2025)
Changed paths:
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LayoutTests/fast/mediastream/RTCPeerConnection-addIceCandidate-expected.txt
M LayoutTests/fast/mediastream/RTCPeerConnection-inspect-answer-expected.txt
M LayoutTests/fast/mediastream/RTCPeerConnection-inspect-offer-expected.txt
M LayoutTests/fast/mediastream/RTCPeerConnection-statsSelector-expected.txt
M LayoutTests/fast/mediastream/RTCPeerConnection-statsSelector.html
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LayoutTests/imported/w3c/web-platform-tests/webrtc-stats/rtp-stats-creation-expected.txt
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LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpReceiver-getStats.https-expected.txt
M LayoutTests/webrtc/audio-muted-stats.html
M LayoutTests/webrtc/video-stats.html
M Source/ThirdParty/libwebrtc/Configurations/libwebrtc.debug.exp
M Source/ThirdParty/libwebrtc/Configurations/libwebrtc.exp
M Source/ThirdParty/libwebrtc/Configurations/libwebrtc.release.exp
M Source/ThirdParty/libwebrtc/Source/webrtc/.gn
M Source/ThirdParty/libwebrtc/Source/webrtc/.vpython3
M Source/ThirdParty/libwebrtc/Source/webrtc/AUTHORS
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M Source/ThirdParty/libwebrtc/Source/webrtc/DEPS
M Source/ThirdParty/libwebrtc/Source/webrtc/PRESUBMIT.py
M Source/ThirdParty/libwebrtc/Source/webrtc/README.chromium
M Source/ThirdParty/libwebrtc/Source/webrtc/api/BUILD.gn
M Source/ThirdParty/libwebrtc/Source/webrtc/api/DEPS
M Source/ThirdParty/libwebrtc/Source/webrtc/api/OWNERS
M Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/async_dns_resolver.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/BUILD.gn
A Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/DEPS
M Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/OWNERS
M Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_frame.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_frame.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_processing.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_processing_statistics.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_processing_statistics.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/builtin_audio_processing_builder.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/builtin_audio_processing_builder.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_factory.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder.cc
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M Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/api/call/bitrate_allocation.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.cc
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M Source/ThirdParty/libwebrtc/Source/webrtc/api/data_channel_interface.h
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M Source/ThirdParty/libwebrtc/Source/webrtc/api/dtls_transport_interface.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/dtmf_sender_interface.h
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M Source/ThirdParty/libwebrtc/Source/webrtc/api/field_trials_unittest.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/api/field_trials_view.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/frame_transformer_interface.h
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M Source/ThirdParty/libwebrtc/Source/webrtc/api/ice_transport_factory.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/api/ice_transport_factory.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/ice_transport_interface.h
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M Source/ThirdParty/libwebrtc/Source/webrtc/api/packet_socket_factory.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/peer_connection_interface.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/api/peer_connection_interface.h
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M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtc_event_log/BUILD.gn
M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtc_event_log/rtc_event.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtc_event_log/rtc_event.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_parameters.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_parameters.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_parameters_unittest.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_receiver_interface.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_sender_interface.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/sctp_transport_interface.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/sequence_checker.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/sequence_checker_unittest.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats_objects.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/task_queue/pending_task_safety_flag.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/test/compile_all_headers.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/api/test/metrics/stdout_metrics_exporter_test.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_datagram_connection.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_datagram_connection_observer.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_local_network_access_permission.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_local_network_access_permission.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_packet_socket_factory.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_peerconnectioninterface.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_session_description_interface.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_transformable_video_frame.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_decoder.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/test/network_emulation/leaky_bucket_network_queue_unittest.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/field_trial_based_config.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_control.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/sctp_transport_factory_interface.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/uma_metrics.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.h
M Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp_unittest.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/api/video/corruption_detection/frame_instrumentation_data.cc
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M Source/ThirdParty/libwebrtc/Source/webrtc/api/video/encoded_frame.h
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M Source/ThirdParty/libwebrtc/Source/webrtc/api/video/frame_buffer.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/video/timing/simulator/rendering_tracker_unittest.cc
A
Source/ThirdParty/libwebrtc/Source/webrtc/video/timing/simulator/rtc_event_log_driver.cc
A
Source/ThirdParty/libwebrtc/Source/webrtc/video/timing/simulator/rtc_event_log_driver.h
A
Source/ThirdParty/libwebrtc/Source/webrtc/video/timing/simulator/rtc_event_log_driver_unittest.cc
A
Source/ThirdParty/libwebrtc/Source/webrtc/video/timing/simulator/rtp_packet_simulator.cc
A
Source/ThirdParty/libwebrtc/Source/webrtc/video/timing/simulator/rtp_packet_simulator.h
A
Source/ThirdParty/libwebrtc/Source/webrtc/video/timing/simulator/rtp_packet_simulator_unittest.cc
A
Source/ThirdParty/libwebrtc/Source/webrtc/video/timing/simulator/test/BUILD.gn
A
Source/ThirdParty/libwebrtc/Source/webrtc/video/timing/simulator/test/encoded_frame_generators.cc
A
Source/ThirdParty/libwebrtc/Source/webrtc/video/timing/simulator/test/encoded_frame_generators.h
A
Source/ThirdParty/libwebrtc/Source/webrtc/video/timing/simulator/test/encoded_frame_generators_unittest.cc
A
Source/ThirdParty/libwebrtc/Source/webrtc/video/timing/simulator/test/matchers.h
A
Source/ThirdParty/libwebrtc/Source/webrtc/video/timing/simulator/test/parsed_rtc_event_log_builder.cc
A
Source/ThirdParty/libwebrtc/Source/webrtc/video/timing/simulator/test/parsed_rtc_event_log_builder.h
A
Source/ThirdParty/libwebrtc/Source/webrtc/video/timing/simulator/test/simulated_time_test_fixture.cc
A
Source/ThirdParty/libwebrtc/Source/webrtc/video/timing/simulator/test/simulated_time_test_fixture.h
M Source/ThirdParty/libwebrtc/Source/webrtc/video/video_analyzer.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/video/video_loopback.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream2.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream2.h
M
Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream2_unittest.cc
M
Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream_timeout_tracker_unittest.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl.cc
M
Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl_unittest.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_tests.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.cc
M Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.h
M
Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder_unittest.cc
M
Source/ThirdParty/libwebrtc/Source/webrtc/webkit_sdk/WebKit/WebKitDecoder.h
M
Source/ThirdParty/libwebrtc/Source/webrtc/webkit_sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm
M Source/ThirdParty/libwebrtc/Source/webrtc/webrtc.gni
M Source/ThirdParty/libwebrtc/Source/webrtc/whitespace.txt
M Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj
M Source/WebCore/Modules/mediastream/RTCIceCandidateFields.cpp
M
Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCIceTransportBackend.cpp
M Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
M Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
M
Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverTransformBackend.cpp
M
Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderTransformBackend.cpp
M Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp
M Source/WebCore/platform/libwebrtc/LibWebRTCVPXVideoEncoder.cpp
M Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp
M Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProviderCocoa.cpp
M Source/WebCore/platform/video-codecs/cocoa/WebRTCVideoDecoder.mm
M Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp
M Source/WebCore/testing/MockLibWebRTCPeerConnection.h
M Source/WebKit/Shared/RTCWebKitEncodedFrameInfo.h
M Source/WebKit/WebProcess/Network/webrtc/LibWebRTCProvider.cpp
M Source/WebKit/WebProcess/Network/webrtc/LibWebRTCSocketFactory.cpp
M Source/WebKit/WebProcess/Network/webrtc/LibWebRTCSocketFactory.h
Log Message:
-----------
Resync libwebrtc to M143
https://bugs.webkit.org/show_bug.cgi?id=303477
rdar://problem/165764376
Reviewed by Eric Carlson.
We update WebCore and WebKit according API changes.
We also have to update some tests as stats handling has changed in libwebrtc.
This allows to do some clean-up like passing unique_ptr instead of raw pointers
for async sockets.
We have to rebase some of the mock connection tests since we are now parsing
the provided SDPs via libwebrtc.
In particular inbound and outbound stats are only populated when packets are
flowing, while previously it would happen as soon as the peer connection is
created.
The updated or rebased tests exhibit the same behaviour (including the newly
failing WPT tests which will require changes).
* Source/ThirdParty/libwebrtc/Configurations/libwebrtc.exp:
* Source/ThirdParty/libwebrtc/Source/webrtc/webkit_sdk/WebKit/WebKitDecoder.h:
*
Source/ThirdParty/libwebrtc/Source/webrtc/webkit_sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm:
(-[RTCEncodedImage initWithNativeEncodedImage:]):
* Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj:
* Source/WebCore/Modules/mediastream/RTCIceCandidateFields.cpp:
(WebCore::parseIceCandidateSDP):
* Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCIceTransportBackend.cpp:
(WebCore::LibWebRTCIceTransportBackendObserver::start):
(WebCore::LibWebRTCIceTransportBackendObserver::stop):
* Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
(WebCore::prepareConfiguration):
(WebCore::LibWebRTCMediaEndpoint::OnIceCandidatesRemoved): Deleted.
* Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
*
Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpReceiverTransformBackend.cpp:
(WebCore::LibWebRTCRtpReceiverTransformBackend::setTransformableFrameCallback):
*
Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpSenderTransformBackend.cpp:
(WebCore::LibWebRTCRtpSenderTransformBackend::setTransformableFrameCallback):
* Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCUtils.cpp:
* Source/WebCore/platform/libwebrtc/LibWebRTCVPXVideoEncoder.cpp:
* Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp:
* Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProviderCocoa.cpp:
* Source/WebCore/platform/video-codecs/cocoa/WebRTCVideoDecoder.mm:
* Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp:
(WebCore::MockLibWebRTCPeerConnection::CreateOffer):
(WebCore::MockLibWebRTCPeerConnection::CreateAnswer):
* Source/WebCore/testing/MockLibWebRTCPeerConnection.h:
* Source/WebKit/Shared/RTCWebKitEncodedFrameInfo.h:
* Source/WebKit/WebProcess/Network/webrtc/LibWebRTCProvider.cpp:
(WebKit::RTCSocketFactory::CreateUdpSocket):
(WebKit::RTCSocketFactory::CreateClientTcpSocket):
* Source/WebKit/WebProcess/Network/webrtc/LibWebRTCSocketFactory.cpp:
(WebKit::LibWebRTCSocketFactory::createUdpSocket):
(WebKit::LibWebRTCSocketFactory::createClientTcpSocket):
* Source/WebKit/WebProcess/Network/webrtc/LibWebRTCSocketFactory.h:
Canonical link: https://commits.webkit.org/303981@main
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