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webrtc-version-updater a1806e97ed Update WebRTC code version (2020-12-17T07:06:37). 
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: I8298b3bf1b3a3a0b769178fdeafc00c06513b5ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198060
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@{#32851}
2020-12-17 09:11:41 +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
Fix standard GetStats to not modify NetEq state.
2020-09-14 09:51:21 +00:00
audio_send_stream.cc
Log audio network adaptor and DSCP in AudioSendStream.
2020-08-13 14:05:46 +00:00
audio_send_stream.h
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audio_sender.h
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audio_state.cc
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audio_state.h
Async audio processing API
2020-10-02 12:33:34 +00:00
bitrate_allocator_unittest.cc
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bitrate_allocator.cc
…
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
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
call_config.cc
…
call_config.h
…
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
Disable flaky test.
2020-11-18 18:22:58 +00:00
call_unittest.cc
Remove nesting of Naggy/Strict/NiceMock
2020-12-07 08:19:50 +00:00
call.cc
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +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
…
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
…
flexfec_receive_stream_impl.h
…
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
…
rampup_tests.cc
Default enable sending transport sequence numbers on audio packets.
2020-11-24 09:19:54 +00:00
rampup_tests.h
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receive_time_calculator_unittest.cc
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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
…
rtp_demuxer_unittest.cc
Delete callbacks from RtpDemuxer on ssrc binding
2020-07-17 15:41:39 +00:00
rtp_demuxer.cc
Delete callbacks from RtpDemuxer on ssrc binding
2020-07-17 15:41:39 +00:00
rtp_demuxer.h
Delete callbacks from RtpDemuxer on ssrc binding
2020-07-17 15:41:39 +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
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rtp_stream_receiver_controller.cc
…
rtp_stream_receiver_controller.h
Rename CriticalSection to RecursiveCriticalSection.
2020-07-17 09:19:50 +00:00
rtp_transport_controller_send_interface.h
Adds ability to delay pacer start until media is added.
2020-09-14 21:42:55 +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
Don't instantiate any CallStats object in RtpVideoSender tests
2020-12-16 14:01:02 +00:00
rtp_video_sender.cc
Use frame rate in video overhead calculation.
2020-12-01 09:42:40 +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
…
simulated_network_unittest.cc
…
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 (2020-12-17T07:06:37).
2020-12-17 09:11:41 +00:00
version.h
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
video_receive_stream.cc
…
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