[webkit-changes] [WebKit/WebKit] 709ec7: Update libwebrtc to M128

youennf Mon, 29 Jul 2024 00:04:49 -0700

  Branch: refs/heads/main
  Home:   https://github.com/WebKit/WebKit
  Commit: 709ec78d82966e476e61c08a8c33a8dcbcc34947
      
https://github.com/WebKit/WebKit/commit/709ec78d82966e476e61c08a8c33a8dcbcc34947
  Author: Youenn Fablet <you...@apple.com>
  Date:   2024-07-29 (Mon, 29 Jul 2024)


  Changed paths:
    M Source/ThirdParty/libwebrtc/Configurations/libwebrtc.exp
    M Source/ThirdParty/libwebrtc/Source/webrtc/.gn
    M Source/ThirdParty/libwebrtc/Source/webrtc/AUTHORS
    M Source/ThirdParty/libwebrtc/Source/webrtc/BUILD.gn
    M Source/ThirdParty/libwebrtc/Source/webrtc/DEPS
    M Source/ThirdParty/libwebrtc/Source/webrtc/PRESUBMIT.py
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/BUILD.gn
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/DEPS
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/adaptation/BUILD.gn
    R Source/ThirdParty/libwebrtc/Source/webrtc/api/adaptation/DEPS
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/adaptation/resource.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view_unittest.cc
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/BUILD.gn
    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_mixer.h
    A Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_view.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_factory.cc
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    M Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/BUILD.gn
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/BUILD.gn
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder.cc
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory_template.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.cc
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory_template.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.cc
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    M Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/BUILD.gn
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    M Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/BUILD.gn
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    M Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/BUILD.gn
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    M Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.cc
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.cc
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    M Source/ThirdParty/libwebrtc/Source/webrtc/api/dtls_transport_interface.cc
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/dtls_transport_interface.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/enable_media.cc
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    M Source/ThirdParty/libwebrtc/Source/webrtc/api/field_trials.cc
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/field_trials_registry.cc
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/field_trials_unittest.cc
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/frame_transformer_factory.cc
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/frame_transformer_factory.h
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    M Source/ThirdParty/libwebrtc/Source/webrtc/api/function_view.h
    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/legacy_stats_types.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/make_ref_counted.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/media_stream_interface.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/metronome/BUILD.gn
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/metronome/test/BUILD.gn
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    M Source/ThirdParty/libwebrtc/Source/webrtc/api/neteq/BUILD.gn
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/neteq/custom_neteq_factory.h
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Source/ThirdParty/libwebrtc/Source/webrtc/api/neteq/default_neteq_controller_factory.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/neteq/neteq.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/neteq/tick_timer_unittest.cc
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/network_state_predictor.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/numerics/BUILD.gn
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Source/ThirdParty/libwebrtc/Source/webrtc/api/numerics/samples_stats_counter.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/peer_connection_interface.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/ref_counted_base.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtc_error.h
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    M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtc_event_log_output.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtc_event_log_output_file.cc
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    M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_headers.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_packet_info.cc
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_packet_info_unittest.cc
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_packet_infos.h
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    M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_parameters.cc
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_receiver_interface.cc
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_transceiver_interface.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/scoped_refptr.h
    M Source/ThirdParty/libwebrtc/Source/webrtc/api/scoped_refptr_unittest.cc
    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
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Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_transformable_video_frame.h
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    M Source/ThirdParty/libwebrtc/Source/webrtc/video/encoder_rtcp_feedback.cc
    M Source/ThirdParty/libwebrtc/Source/webrtc/video/encoder_rtcp_feedback.h
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    M Source/ThirdParty/libwebrtc/Source/webrtc/video/full_stack_tests.cc
    M Source/ThirdParty/libwebrtc/Source/webrtc/video/pc_full_stack_tests.cc
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    M Source/ThirdParty/libwebrtc/Source/webrtc/video/screenshare_loopback.cc
    M Source/ThirdParty/libwebrtc/Source/webrtc/video/sv_loopback.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
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    M Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl.cc
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    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/webrtc.gni
    M Source/ThirdParty/libwebrtc/Source/webrtc/whitespace.txt
    M Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj
    M Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
    M Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCObservers.h
    M 
Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransformBackend.h
    M Source/WebCore/platform/mediastream/RealtimeOutgoingAudioSource.h
    M Source/WebCore/platform/mediastream/RealtimeOutgoingVideoSource.h
    M Source/WebCore/testing/MockLibWebRTCPeerConnection.h

  Log Message:
  -----------
  Update libwebrtc to M128
https://bugs.webkit.org/show_bug.cgi?id=277131
rdar://132558297

Reviewed by Eric Carlson.

Update libwebrtc to M128 branch and update client code according API changes.

* Source/ThirdParty/libwebrtc/Configurations/libwebrtc.exp:
* Source/ThirdParty/libwebrtc/Source/webrtc: Resynced.
* Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj:
* Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
(WebCore::LibWebRTCMediaEndpoint::Release const):
* Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCObservers.h:
* Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCRtpTransformBackend.h:
(WebCore::LibWebRTCRtpTransformBackend::Release const):
* Source/WebCore/platform/mediastream/RealtimeOutgoingAudioSource.h:
* Source/WebCore/platform/mediastream/RealtimeOutgoingVideoSource.h:
* Source/WebCore/testing/MockLibWebRTCPeerConnection.h:

Canonical link: https://commits.webkit.org/281483@main



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[webkit-changes] [WebKit/WebKit] 709ec7: Update libwebrtc to M128

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