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webrtc-version-updater 42151dfc73 Update WebRTC code version (2024-05-31T04:02:17). 
Bug: None
Change-Id: I86506f31ededc967d0aaa4b41fe8f2c43a066ac5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353061
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42418}
2024-05-31 05:24:02 +00:00
..
adaptation
Mass removal of absl_deps in all BUILD.gn files
2024-05-23 15:09:46 +00:00
test
Add missing header
2024-04-19 06:13:16 +00:00
audio_receive_stream.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
audio_receive_stream.h
RTCP: implement reduced size RTCP for audio
2024-05-16 18:24:10 +00:00
audio_send_stream.cc
…
audio_send_stream.h
RTCP: implement reduced size RTCP for audio
2024-05-16 18:24:10 +00:00
audio_sender.h
…
audio_state.cc
…
audio_state.h
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
bitrate_allocator_unittest.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
bitrate_allocator.cc
…
bitrate_allocator.h
…
bitrate_estimator_tests.cc
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
BUILD.gn
Mass removal of absl_deps in all BUILD.gn files
2024-05-23 15:09:46 +00:00
call_config.cc
Use Environment in RtpTransportyControllerSend
2023-12-20 14:47:51 +00:00
call_config.h
Reland "FrameCadenceAdapter: align video encoding to metronome"
2024-01-08 13:54:56 +00:00
call_perf_tests.cc
Ensure test capturer does not set ntp time
2024-05-29 10:26:54 +00:00
call_unittest.cc
Pass Environment instead of clock to Fake video encoders at construction
2024-04-12 07:42:48 +00:00
call.cc
Expose network to deliverrtp histogram metric
2024-05-29 08:50:22 +00:00
call.h
Pass Clock through Environment when constructing Call
2023-12-06 19:13:39 +00:00
create_call.cc
Pass Clock through Environment when constructing Call
2023-12-06 19:13:39 +00:00
create_call.h
Delete CallFactoryInterface as no longer needed
2023-12-05 15:44:43 +00:00
degraded_call.cc
Remove internal overrides using old SendRtp and SendRtcp interfaces.
2023-08-15 13:20:21 +00:00
degraded_call.h
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
DEPS
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
fake_network_pipe_unittest.cc
Reland "Replace usage of link_capacity_kbps with DataRate link_capacity"
2024-05-16 10:39:10 +00:00
fake_network_pipe.cc
Delete unused constructor of FakeNetworkPipe
2023-08-18 13:07:10 +00:00
fake_network_pipe.h
Delete unused constructor of FakeNetworkPipe
2023-08-18 13:07:10 +00:00
flexfec_receive_stream_impl.cc
stats: implement flexfec fecBytesReceived stats for FlexFEC
2023-06-21 13:04:31 +00:00
flexfec_receive_stream_impl.h
stats: implement flexfec fecBytesReceived stats for FlexFEC
2023-06-21 13:04:31 +00:00
flexfec_receive_stream_unittest.cc
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
flexfec_receive_stream.cc
…
flexfec_receive_stream.h
stats: implement flexfec fecBytesReceived stats for FlexFEC
2023-06-21 13:04:31 +00:00
OWNERS
Make perkj owner of call/
2024-05-07 15:13:29 +00:00
packet_receiver.h
Dont create RTX receive stream before media SSRC is known
2024-02-22 14:40:43 +00:00
rampup_tests.cc
Reland "Replace usage of link_capacity_kbps with DataRate link_capacity"
2024-05-16 10:39:10 +00:00
rampup_tests.h
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
receive_stream.h
RTCP: implement reduced size RTCP for audio
2024-05-16 18:24:10 +00:00
receive_time_calculator_unittest.cc
…
receive_time_calculator.cc
…
receive_time_calculator.h
…
rtp_bitrate_configurator_unittest.cc
…
rtp_bitrate_configurator.cc
…
rtp_bitrate_configurator.h
…
rtp_config.cc
…
rtp_config.h
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtp_demuxer_unittest.cc
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
rtp_demuxer.cc
Remove SSRCs from libSRTP when removing them from the rtp_demuxer
2023-11-08 10:24:10 +00:00
rtp_demuxer.h
Remove SSRCs from libSRTP when removing them from the rtp_demuxer
2023-11-08 10:24:10 +00:00
rtp_packet_sink_interface.h
…
rtp_payload_params_unittest.cc
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtp_payload_params.cc
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtp_payload_params.h
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtp_stream_receiver_controller_interface.h
Demote RtpStreamReceiverController AddSink/RemoveSink to private
2022-07-06 09:31:54 +00:00
rtp_stream_receiver_controller.cc
Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
2022-12-22 14:04:21 +00:00
rtp_stream_receiver_controller.h
Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
2022-12-22 14:04:21 +00:00
rtp_transport_config.h
Add PeerConnectionInterface::ReconfigureBandwidthEstimation
2024-02-07 14:10:02 +00:00
rtp_transport_controller_send_factory_interface.h
Use Environment in RtpTransportyControllerSend
2023-12-20 14:47:51 +00:00
rtp_transport_controller_send_factory.h
Use Environment in RtpTransportyControllerSend
2023-12-20 14:47:51 +00:00
rtp_transport_controller_send_interface.h
PacketRouter directly notify RtpTransportControllerSender when sending
2024-03-28 09:27:43 +00:00
rtp_transport_controller_send.cc
Add field trial to reset BWE if Network adapter changes instead of if IP address changes
2024-05-16 12:17:51 +00:00
rtp_transport_controller_send.h
Add field trial to reset BWE if Network adapter changes instead of if IP address changes
2024-05-16 12:17:51 +00:00
rtp_video_sender_interface.h
Refactor RtpVideoSender::SetActiveModules.
2024-01-26 10:34:46 +00:00
rtp_video_sender_unittest.cc
Change WebRTC-Video-SimulcastIndependentFrameIds to a kill-switch.
2024-05-23 09:42:23 +00:00
rtp_video_sender.cc
Change WebRTC-Video-SimulcastIndependentFrameIds to a kill-switch.
2024-05-23 09:42:23 +00:00
rtp_video_sender.h
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtx_receive_stream_unittest.cc
…
rtx_receive_stream.cc
Updated associated payload types without recreating receive streams.
2022-08-16 13:38:24 +00:00
rtx_receive_stream.h
Updated associated payload types without recreating receive streams.
2022-08-16 13:38:24 +00:00
simulated_network.h
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
simulated_packet_receiver.h
…
syncable.cc
…
syncable.h
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
version.cc
Update WebRTC code version (2024-05-31T04:02:17).
2024-05-31 05:24:02 +00:00
version.h
…
video_receive_stream.cc
Add missing comma in VideoReceiveStreamInterface::Stats::ToString
2023-10-17 10:42:06 +00:00
video_receive_stream.h
RTCP: implement reduced size RTCP for audio
2024-05-16 18:24:10 +00:00
video_send_stream.cc
Cleanup usasge of ReportBlockData::report_block accessor
2023-05-05 09:56:30 +00:00
video_send_stream.h
Remove VideoSendStream::StartPerRtpStream
2024-01-26 09:19:50 +00:00