webrtc/call
webrtc-version-updater d695ace023 Update WebRTC code version (2021-07-23T04:03:53).
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: Ida9dc0b22d0121bac154936f8857d29e2101bc8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226820
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@{#34535}
2021-07-23 05:49:17 +00:00
..
adaptation Ensure that fps adaptation count can go back to zero when framerate is unrestricted. 2021-06-10 16:00:39 +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 Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
audio_receive_stream.h Adding packetsDiscarded to RTCReceivedRtpStreamStats. 2021-07-13 20:34:45 +00:00
audio_send_stream.cc Delete unneeded references to string_encode.h 2021-07-01 09:35:23 +00:00
audio_send_stream.h Implement nack_count metric for outbound audio rtp streams. 2021-07-09 13:29:10 +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.cc Replace DataSize and DataRate factories with newer versions 2020-02-18 16:09:50 +00:00
bitrate_allocator.h Use SequenceChecker from public API 2021-02-10 15:04:55 +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_estimator_tests.cc Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory." 2020-08-06 11:50:08 +00:00
BUILD.gn Delete remaining usage of RtpHeaderParser test helper. 2021-07-22 10:15:07 +00:00
call.cc NackModule2: coalesce repeating tasks. 2021-07-22 12:11:13 +00:00
call.h Update the sync_group id without recreating audio receive streams. 2021-06-16 19:34:18 +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 Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies. 2021-06-01 06:57:31 +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 Delete remaining usage of RtpHeaderParser test helper. 2021-07-22 10:15:07 +00:00
call_unittest.cc Factor out common receive stream methods to a common interface. 2021-06-14 16:54:07 +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 Async audio processing API 2020-10-02 12:33:34 +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
fake_network_pipe_unittest.cc In call/ replace mock macros with unified MOCK_METHOD macro 2020-05-15 13:36:00 +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
flexfec_receive_stream_impl.cc ModuleRtcRtcpImpl2: remove Module inheritance. 2021-06-22 14:51:04 +00:00
flexfec_receive_stream_impl.h ModuleRtcRtcpImpl2: remove Module inheritance. 2021-06-22 14:51:04 +00:00
flexfec_receive_stream_unittest.cc ModuleRtcRtcpImpl2: remove Module inheritance. 2021-06-22 14:51:04 +00:00
OWNERS Make sprang@ owner in call 2020-10-19 10:30:03 +00:00
packet_receiver.h Remove DeliverPacketAsync. 2021-05-29 07:37:33 +00:00
rampup_tests.cc Factor out common receive stream methods to a common interface. 2021-06-14 16:54:07 +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 rtp_config() accessor to ReceiveStream. 2021-06-14 17:57:57 +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
receive_time_calculator_unittest.cc Format almost everything. 2019-07-08 13:45:15 +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 Allow setting a bandwidth cap for relayed connections. 2020-03-26 20:41:46 +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_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.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_demuxer_unittest.cc Delete callbacks from RtpDemuxer on ssrc binding 2020-07-17 15:41:39 +00:00
rtp_packet_sink_interface.h
rtp_payload_params.cc Provide FrameDependecyStructure for VP9 when encoder doesn't fill it 2021-04-29 12:28:46 +00:00
rtp_payload_params.h Provide FrameDependecyStructure for VP9 when encoder doesn't fill it 2021-04-29 12:28:46 +00:00
rtp_payload_params_unittest.cc Calculate VP9 generic info from vp9 specific info 2021-04-26 17:49:59 +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_stream_receiver_controller_interface.h
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.cc Remove synchronization from VideoSendStream construction. 2021-06-03 19:13:45 +00:00
rtp_transport_controller_send.h Remove synchronization from VideoSendStream construction. 2021-06-03 19:13:45 +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_factory_interface.h Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies. 2021-06-01 06:57:31 +00:00
rtp_transport_controller_send_interface.h Delete RtcpStatisticsCallback in favor of ReportBlockDataObserver 2021-05-16 15:09:29 +00:00
rtp_video_sender.cc ModuleRtcRtcpImpl2: remove Module inheritance. 2021-06-22 14:51:04 +00:00
rtp_video_sender.h ModuleRtcRtcpImpl2: remove Module inheritance. 2021-06-22 14:51:04 +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 "Correctly handle retransmissions/padding in early loss detection." 2021-06-16 08:14:27 +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
rtx_receive_stream_unittest.cc Store RtpPacketReceived::arrival_time as Timestamp. 2021-05-05 16:22:33 +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 Use SequenceChecker from public API 2021-02-10 15:04:55 +00:00
simulated_network_unittest.cc Replace DataSize and DataRate factories with newer versions 2020-02-18 16:09:50 +00:00
simulated_packet_receiver.h Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
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 (2021-07-23T04:03:53). 2021-07-23 05:49:17 +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 Prepare WebRtcVideoReceiveStream for configuration changes. 2021-07-01 11:23:51 +00:00
video_send_stream.cc Remove redundant VideoSendStream::rtcp_stats field 2021-05-18 13:37:51 +00:00
video_send_stream.h Remove redundant VideoSendStream::rtcp_stats field 2021-05-18 13:37:51 +00:00