admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/call
History
webrtc-version-updater 70f445e592 Update WebRTC code version (2021-12-09T04:05:04). 
Bug: None
Change-Id: I5a6ca60f80741bc052a9c58b1054dfd2fbd2f982
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240443
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@{#35506}
2021-12-09 05:13:28 +00:00
..
adaptation
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
2021-11-15 21:44:59 +00:00
test
Remove unnecessary copy of suspended_ssrcs.
2021-11-29 10:54:30 +00:00
audio_receive_stream.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
audio_receive_stream.h
Add ability to set rtp header extensions without recreating streams.
2021-09-08 13:39:36 +00:00
audio_send_stream.cc
Reland "Wire up non-sender RTT for audio, and implement related standardized stats."
2021-09-06 14:26:55 +00:00
audio_send_stream.h
Reland "Wire up non-sender RTT for audio, and implement related standardized stats."
2021-09-06 14:26:55 +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
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
audio_state.h
Async audio processing API
2020-10-02 12:33:34 +00:00
bitrate_allocator_unittest.cc
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
bitrate_allocator.cc
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
bitrate_allocator.h
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +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
Change the type of RTCVideoSourceStats.framesPerSecond
2021-11-16 11:21:41 +00:00
call_config.cc
Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies.
2021-06-01 06:57:31 +00:00
call_config.h
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
call_factory.cc
Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies.
2021-06-01 06:57:31 +00:00
call_factory.h
Use SequenceChecker from public API
2021-02-10 15:04:55 +00:00
call_perf_tests.cc
CallPerfTest: Add encode framerate test with lower input fps.
2021-09-02 12:58:02 +00:00
call_unittest.cc
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
call.cc
Reland "Call: Deduplicate SentPacket notifications"
2021-12-07 16:15:47 +00:00
call.h
Update the sync_group id without recreating audio receive streams.
2021-06-16 19:34:18 +00:00
degraded_call.cc
Update the sync_group id without recreating audio receive streams.
2021-06-16 19:34:18 +00:00
degraded_call.h
Update the sync_group id without recreating audio receive streams.
2021-06-16 19:34:18 +00:00
DEPS
SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated.
2021-08-30 10:20:55 +00:00
fake_network_pipe_unittest.cc
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
fake_network_pipe.cc
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
fake_network_pipe.h
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
flexfec_receive_stream_impl.cc
Add ability to set rtp header extensions without recreating streams.
2021-09-08 13:39:36 +00:00
flexfec_receive_stream_impl.h
Add ability to set rtp header extensions without recreating streams.
2021-09-08 13:39:36 +00:00
flexfec_receive_stream_unittest.cc
Deduplicate set of the rtp header extension uri constants
2021-09-14 13:38:44 +00:00
flexfec_receive_stream.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
flexfec_receive_stream.h
Add rtp_config() accessor to ReceiveStream.
2021-06-14 17:57:57 +00:00
OWNERS
Fix WebRTC version updater bot.
2021-11-22 08:44:15 +00:00
packet_receiver.h
Remove DeliverPacketAsync.
2021-05-29 07:37:33 +00:00
rampup_tests.cc
Fix -Wunused-but-set-variable.
2021-10-28 12:53:49 +00:00
rampup_tests.h
Update call Rampup tests not to rely on DEPRECATED_SingleThreadedTaskQueueForTesting
2019-10-21 12:33:27 +00:00
receive_stream.h
Add ability to set rtp header extensions without recreating streams.
2021-09-08 13:39:36 +00:00
receive_time_calculator_unittest.cc
Format almost everything.
2019-07-08 13:45:15 +00:00
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
Format almost everything.
2019-07-08 13:45:15 +00:00
rtp_bitrate_configurator_unittest.cc
Revert "In RtpBitrateConfigurator ignore new parameters when set to default values."
2020-01-10 16:39:51 +00:00
rtp_bitrate_configurator.cc
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +00:00
rtp_bitrate_configurator.h
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
rtp_config.cc
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
2021-11-15 21:44:59 +00:00
rtp_config.h
Use backticks not vertical bars to denote variables in comments
2021-08-10 10:40:03 +00:00
rtp_demuxer_unittest.cc
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
rtp_demuxer.cc
Use flat_map/flat_set in RtpDemuxer
2021-07-07 08:22:55 +00:00
rtp_demuxer.h
Use flat_map/flat_set in RtpDemuxer
2021-07-07 08:22:55 +00:00
rtp_packet_sink_interface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_payload_params_unittest.cc
Reland "Reland "Enable WebRTC-Vp9DependencyDescriptor by default""
2021-09-15 13:48:58 +00:00
rtp_payload_params.cc
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
2021-11-15 21:44:59 +00:00
rtp_payload_params.h
Cleanup WebRTC-Vp9DependencyDescriptor field trial
2021-10-19 14:29:29 +00:00
rtp_stream_receiver_controller_interface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_stream_receiver_controller.cc
Remove lock from RtpStreamReceiverController.
2021-01-18 09:10:14 +00:00
rtp_stream_receiver_controller.h
Use SequenceChecker from public API
2021-02-10 15:04:55 +00:00
rtp_transport_config.h
Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies.
2021-06-01 06:57:31 +00:00
rtp_transport_controller_send_factory_interface.h
Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies.
2021-06-01 06:57:31 +00:00
rtp_transport_controller_send_factory.h
Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies.
2021-06-01 06:57:31 +00:00
rtp_transport_controller_send_interface.h
Remove unnecessary copy of suspended_ssrcs.
2021-11-29 10:54:30 +00:00
rtp_transport_controller_send.cc
RtpTransportControllerSend: Restrict pacer outstanding data update
2021-12-07 11:58:07 +00:00
rtp_transport_controller_send.h
Remove unnecessary copy of suspended_ssrcs.
2021-11-29 10:54:30 +00:00
rtp_video_sender_interface.h
ModuleRtcRtcpImpl2: remove Module inheritance.
2021-06-22 14:51:04 +00:00
rtp_video_sender_unittest.cc
Reland "Reland "Enable WebRTC-Vp9DependencyDescriptor by default""
2021-09-15 13:48:58 +00:00
rtp_video_sender.cc
Remove unnecessary copy of suspended_ssrcs.
2021-11-29 10:54:30 +00:00
rtp_video_sender.h
Remove unnecessary copy of suspended_ssrcs.
2021-11-29 10:54:30 +00:00
rtx_receive_stream_unittest.cc
Store RtpPacketReceived::arrival_time as Timestamp.
2021-05-05 16:22:33 +00:00
rtx_receive_stream.cc
Store RtpPacketReceived::arrival_time as Timestamp.
2021-05-05 16:22:33 +00:00
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
Use backticks not vertical bars to denote variables in comments
2021-08-10 10:40:03 +00:00
simulated_network.h
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
simulated_packet_receiver.h
Calculate next process time in simulated network.
2019-02-08 19:33:17 +00:00
syncable.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
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 (2021-12-09T04:05:04).
2021-12-09 05:13:28 +00:00
version.h
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
video_receive_stream.cc
Prepare WebRtcVideoReceiveStream for configuration changes.
2021-07-01 11:23:51 +00:00
video_receive_stream.h
Remove unused VideoReceiveStream.
2021-12-07 15:21:21 +00:00
video_send_stream.cc
Change the type of RTCVideoSourceStats.framesPerSecond
2021-11-16 11:21:41 +00:00
video_send_stream.h
Change the type of RTCVideoSourceStats.framesPerSecond
2021-11-16 11:21:41 +00:00