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webrtc-version-updater 9d653660e9 Update WebRTC code version (2024-02-02T04:03:16). 
Bug: None
Change-Id: Iec8f8291065a0700697ec961cc1cad0c7ea7395a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337290
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@{#41658}
2024-02-02 06:19:28 +00:00
..
adaptation
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
test
Refactor RtpVideoSender::SetActiveModules.
2024-01-26 10:34:46 +00:00
audio_receive_stream.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
audio_receive_stream.h
Propagate time of the last received packet with Timestamp type
2023-06-02 14:29:19 +00:00
audio_send_stream.cc
…
audio_send_stream.h
Replace "rcvd" with "received" for readability
2023-04-24 15:30:07 +00:00
audio_sender.h
…
audio_state.cc
…
audio_state.h
…
bitrate_allocator_unittest.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
bitrate_allocator.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1
2022-03-09 13:23:21 +00:00
bitrate_allocator.h
…
bitrate_estimator_tests.cc
Update CallTests to create Call using Environment
2023-12-01 13:16:41 +00:00
BUILD.gn
Consolidate encoded transform mocks into api/test/
2024-01-26 12:46:34 +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
Update CallTests to create Call using Environment
2023-12-01 13:16:41 +00:00
call_unittest.cc
Allow to create webrtc::Call with Environment
2023-11-28 10:26:56 +00:00
call.cc
Combine video_send_stream_impl.cc and video_send_stream.cc
2024-01-18 13:37:42 +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
Remove internal overrides using old SendRtp and SendRtcp interfaces.
2023-08-15 13:20:21 +00:00
DEPS
…
fake_network_pipe_unittest.cc
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
2023-01-25 18:18:29 +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
Update OWNERS for call/
2022-06-03 12:01:46 +00:00
packet_receiver.h
Allow injecting packets of type Any to Call::DeliverRtpPacket
2023-03-29 06:36:17 +00:00
rampup_tests.cc
Update CallTests to create Call using Environment
2023-12-01 13:16:41 +00:00
rampup_tests.h
Stop overriding extensions in rampup tests
2023-01-25 13:18:49 +00:00
receive_stream.h
Remove rtp header extension from config of Call audio and video receivers
2023-01-31 11:58:43 +00:00
receive_time_calculator_unittest.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2
2022-03-09 22:17:52 +00:00
receive_time_calculator.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
receive_time_calculator.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
rtp_bitrate_configurator_unittest.cc
…
rtp_bitrate_configurator.cc
…
rtp_bitrate_configurator.h
…
rtp_config.cc
…
rtp_config.h
Introduce support for video packet batching.
2023-05-08 16:24:03 +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
When simulating chains from VP9 codec specific info support first_active_layer > 0
2023-08-03 13:19:00 +00:00
rtp_payload_params.cc
Add codec name H265 to support H265 in WebRTC
2023-09-20 09:25:32 +00:00
rtp_payload_params.h
For VP9 assume max number of spatial layers to simulate generic descriptor
2022-06-08 11:36:54 +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
Introduce RtpTransportConfig:allow_bandwidht_estimation_probe_without_media
2024-01-29 07:41:32 +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
Refactor RtpVideoSender::SetActiveModules.
2024-01-26 10:34:46 +00:00
rtp_transport_controller_send.cc
Introduce RtpTransportConfig:allow_bandwidht_estimation_probe_without_media
2024-01-29 07:41:32 +00:00
rtp_transport_controller_send.h
Introduce RtpTransportConfig:allow_bandwidht_estimation_probe_without_media
2024-01-29 07:41:32 +00:00
rtp_video_sender_interface.h
Refactor RtpVideoSender::SetActiveModules.
2024-01-26 10:34:46 +00:00
rtp_video_sender_unittest.cc
Consolidate encoded transform mocks into api/test/
2024-01-26 12:46:34 +00:00
rtp_video_sender.cc
Refactor RtpVideoSender::SetActiveModules.
2024-01-26 10:34:46 +00:00
rtp_video_sender.h
Refactor RtpVideoSender::SetActiveModules.
2024-01-26 10:34:46 +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_unittest.cc
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
2022-11-06 13:14:26 +00:00
simulated_network.cc
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
2022-11-06 13:14:26 +00:00
simulated_network.h
Export webrtc::SimulatedNetwork for Chrome component builds
2023-11-27 16:03:23 +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 (2024-02-02T04:03:16).
2024-02-02 06:19:28 +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
Remove default "unknown" encoderImplementation/decoderImplementation
2023-06-22 11:49:58 +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