admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/call
History
webrtc-version-updater a7ccacc87f Update WebRTC code version (2021-01-15T04:04:08). 
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: Ib519879af9426e56c0808fbfc36d064e8788b3db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201961
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@{#32988}
2021-01-15 05:31:49 +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
Include configured mid and rid when logging AudioSendStream config
2020-12-22 14:19:33 +00:00
audio_send_stream.h
…
audio_sender.h
…
audio_state.cc
…
audio_state.h
Async audio processing API
2020-10-02 12:33:34 +00:00
bitrate_allocator_unittest.cc
…
bitrate_allocator.cc
…
bitrate_allocator.h
Reland "Introduce RTC_NO_UNIQUE_ADDRESS."
2020-11-23 11:29:36 +00:00
bitrate_estimator_tests.cc
…
BUILD.gn
Revert "Refactor rtc_base build targets."
2021-01-14 21:27:38 +00:00
call_config.cc
…
call_config.h
…
call_factory.cc
…
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
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
…
fake_network_pipe.h
…
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
…
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
…
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
…
rtp_demuxer_unittest.cc
…
rtp_demuxer.cc
…
rtp_demuxer.h
…
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
…
rtp_stream_receiver_controller.h
…
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
Don't instantiate any CallStats object in RtpVideoSender tests
2020-12-16 14:01:02 +00:00
rtp_video_sender.cc
Revert "Add task queue to RtpRtcpInterface::Configuration."
2021-01-12 17:47:32 +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
…
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 (2021-01-15T04:04:08).
2021-01-15 05:31:49 +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