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webrtc-version-updater cedc3c7200 Update WebRTC code version (2021-01-31T04:03:17). 
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: If4128e29576cbce59ef62852cdd6ea4e9d2d1aff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204821
Reviewed-by: 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/master@{#33115}
2021-01-31 06:10:01 +00:00
..
adaptation
Implement fake PixelLimitResource for TestBed.
2020-12-04 10:35:53 +00:00
test
Adds ability to delay pacer start until media is added.
2020-09-14 21:42:55 +00:00
audio_receive_stream.cc
…
audio_receive_stream.h
Reduce redundant flags for audio stream playout state.
2021-01-18 09:09:09 +00:00
audio_send_stream.cc
Include configured mid and rid when logging AudioSendStream config
2020-12-22 14:19:33 +00:00
audio_send_stream.h
negotiate RED codec for audio
2020-06-25 06:24:18 +00:00
audio_sender.h
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
2020-01-13 18:31:30 +00:00
audio_state.cc
…
audio_state.h
Async audio processing API
2020-10-02 12:33:34 +00:00
bitrate_allocator_unittest.cc
In call/ replace mock macros with unified MOCK_METHOD macro
2020-05-15 13:36:00 +00:00
bitrate_allocator.cc
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
bitrate_allocator.h
Reland "Introduce RTC_NO_UNIQUE_ADDRESS."
2020-11-23 11:29:36 +00:00
bitrate_estimator_tests.cc
Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
2020-08-06 11:50:08 +00:00
BUILD.gn
Fix call_tests target dependencies
2021-01-29 13:39:09 +00:00
call_config.cc
…
call_config.h
Remove deprecated constant.
2020-04-27 10:32:45 +00:00
call_factory.cc
Ensure CreateTimeControllerBasedCallFactory use simulated time in Call::SharedModuleThread
2020-06-30 15:38:35 +00:00
call_factory.h
Reland "Introduce RTC_NO_UNIQUE_ADDRESS."
2020-11-23 11:29:36 +00:00
call_perf_tests.cc
Fix data race on destruction in Audio Video sync tests
2020-12-30 17:11:31 +00:00
call_unittest.cc
Move deprecated code to their own build targets.
2021-01-18 13:09:47 +00:00
call.cc
Add DeliverPacketAsync method to PacketReceiver.
2021-01-19 11:53:50 +00:00
call.h
Reland: Wires up WebrtcKeyValueBasedConfig in media engines.
2020-09-22 16:08:22 +00:00
degraded_call.cc
Reland: Wires up WebrtcKeyValueBasedConfig in media engines.
2020-09-22 16:08:22 +00:00
degraded_call.h
Reland: Wires up WebrtcKeyValueBasedConfig in media engines.
2020-09-22 16:08:22 +00:00
DEPS
Async audio processing API
2020-10-02 12:33:34 +00:00
fake_network_pipe_unittest.cc
In call/ replace mock macros with unified MOCK_METHOD macro
2020-05-15 13:36:00 +00:00
fake_network_pipe.cc
Migrate call/ to webrtc::Mutex.
2020-07-06 15:48:30 +00:00
fake_network_pipe.h
Migrate call/ to webrtc::Mutex.
2020-07-06 15:48:30 +00:00
flexfec_receive_stream_impl.cc
Revert "Add task queue to RtpRtcpInterface::Configuration."
2021-01-12 17:47:32 +00:00
flexfec_receive_stream_impl.h
Remove dependency from RtpRtcp on the Module interface.
2020-06-04 08:11:21 +00:00
flexfec_receive_stream_unittest.cc
…
flexfec_receive_stream.cc
…
flexfec_receive_stream.h
…
OWNERS
Make sprang@ owner in call
2020-10-19 10:30:03 +00:00
packet_receiver.h
Add DeliverPacketAsync method to PacketReceiver.
2021-01-19 11:53:50 +00:00
rampup_tests.cc
Default enable sending transport sequence numbers on audio packets.
2020-11-24 09:19:54 +00:00
rampup_tests.h
…
receive_time_calculator_unittest.cc
…
receive_time_calculator.cc
Use newer version of TimeDelta and TimeStamp factories in webrtc
2020-02-10 12:21:17 +00:00
receive_time_calculator.h
…
rtp_bitrate_configurator_unittest.cc
…
rtp_bitrate_configurator.cc
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +00:00
rtp_bitrate_configurator.h
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +00:00
rtp_config.cc
Reland "Improve outbound-rtp statistics for simulcast"
2020-05-05 20:22:19 +00:00
rtp_config.h
Reland "Improve outbound-rtp statistics for simulcast"
2020-05-05 20:22:19 +00:00
rtp_demuxer_unittest.cc
Delete callbacks from RtpDemuxer on ssrc binding
2020-07-17 15:41:39 +00:00
rtp_demuxer.cc
Remove lock from RtpStreamReceiverController.
2021-01-18 09:10:14 +00:00
rtp_demuxer.h
Remove lock from RtpStreamReceiverController.
2021-01-18 09:10:14 +00:00
rtp_packet_sink_interface.h
…
rtp_payload_params_unittest.cc
in Av1 encoder wrapper communicate end_of_picture flag similar to VP9
2020-11-11 14:00:52 +00:00
rtp_payload_params.cc
Ignore frame type when calculating generic frame dependencies.
2020-11-23 10:52:06 +00:00
rtp_payload_params.h
Ignore frame type when calculating generic frame dependencies.
2020-11-23 10:52:06 +00:00
rtp_stream_receiver_controller_interface.h
…
rtp_stream_receiver_controller.cc
Remove lock from RtpStreamReceiverController.
2021-01-18 09:10:14 +00:00
rtp_stream_receiver_controller.h
Remove lock from RtpStreamReceiverController.
2021-01-18 09:10:14 +00:00
rtp_transport_controller_send_interface.h
Delete unneeded dependencies on the Module abstraction
2020-12-21 09:09:57 +00:00
rtp_transport_controller_send.cc
Reland "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
2020-11-02 11:05:56 +00:00
rtp_transport_controller_send.h
Adds ability to delay pacer start until media is added.
2020-09-14 21:42:55 +00:00
rtp_video_sender_interface.h
Hookup VideoSendStreamImpl::OnVideoLayersAllocationUpdate to RtpVideoSender.
2020-10-19 11:37:23 +00:00
rtp_video_sender_unittest.cc
Avoid treating VP8 key frame in simulcast as delta frame
2021-01-28 12:11:50 +00:00
rtp_video_sender.cc
Avoid treating VP8 key frame in simulcast as delta frame
2021-01-28 12:11:50 +00:00
rtp_video_sender.h
Use frame rate in video overhead calculation.
2020-12-01 09:42:40 +00:00
rtx_receive_stream_unittest.cc
…
rtx_receive_stream.cc
…
rtx_receive_stream.h
IWYU: uint32_t is defined in cstdint
2020-05-07 17:04:15 +00:00
simulated_network_unittest.cc
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
simulated_network.cc
Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
2020-11-09 10:47:55 +00:00
simulated_network.h
Migrate call/ to webrtc::Mutex.
2020-07-06 15:48:30 +00:00
simulated_packet_receiver.h
…
syncable.cc
…
syncable.h
Make AV sync robust to failures to set a desired minimum delay
2020-09-09 15:44:47 +00:00
version.cc
Update WebRTC code version (2021-01-31T04:03:17).
2021-01-31 06:10:01 +00:00
version.h
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
video_receive_stream.cc
Add commas between codec parameters in VideoReceiveStream logging.
2020-03-09 02:45:34 +00:00
video_receive_stream.h
Move FrameCounts and FrameCountObserver to common_video/frame_counts.h
2020-08-27 09:53:18 +00:00
video_send_stream.cc
Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
2020-11-09 10:47:55 +00:00
video_send_stream.h
Move FrameCounts and FrameCountObserver to common_video/frame_counts.h
2020-08-27 09:53:18 +00:00