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webrtc-version-updater e07f8c578b Update WebRTC code version (2021-07-07T04:06:33). 
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: I4782699f036e7b3eea5337a7c607034abc72ebae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225241
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@{#34427}
2021-07-07 05:27:08 +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
Implement nack_count metric for inbound audio rtp streams.
2021-07-01 10:38:44 +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
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
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
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
Use SequenceChecker from public API
2021-02-10 15:04:55 +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
Factor out common receive stream methods to a common interface.
2021-06-14 16:54:07 +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
Reland "Remove Invoke from VideoChannel::FillBitrateInfo."
2021-05-03 15:16:34 +00:00
call_unittest.cc
Factor out common receive stream methods to a common interface.
2021-06-14 16:54:07 +00:00
call.cc
Unify helpers IsRtpPacket and IsRtcpPacket
2021-07-06 10:39:00 +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
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
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
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
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_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
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
Fix unsignalled ssrc race in WebRtcVideoChannel.
2021-04-16 09:33:42 +00:00
rtp_demuxer.h
Partial revert: "Use unordered map in RtpDemuxer"
2021-05-25 15:16:00 +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
Calculate VP9 generic info from vp9 specific info
2021-04-26 17:49:59 +00:00
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_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
Delete RtcpStatisticsCallback in favor of ReportBlockDataObserver
2021-05-16 15:09:29 +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_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
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
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
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_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 (2021-07-07T04:06:33).
2021-07-07 05:27:08 +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