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webrtc-version-updater f7ed83f68e Update WebRTC code version (2023-04-06T04:04:31). 
Bug: None
Change-Id: Icf0395a73598a1b7f9dcfac58c72e74bf986e476
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300421
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@{#39775}
2023-04-06 05:47:24 +00:00
..
adaptation
Apply resolution-bitrate limits collected from field trial (cl/294600) for AV1.
2023-03-16 19:04:32 +00:00
test
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +00:00
audio_receive_stream.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
audio_receive_stream.h
stats: use uint64_t for RTCSentRtpStreamStats.packetsSent
2023-03-16 06:46:19 +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
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +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
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
bitrate_allocator.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1
2022-03-09 13:23:21 +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
Remove rtp header extension from config of Call audio and video receivers
2023-01-31 11:58:43 +00:00
BUILD.gn
Allow injecting packets of type Any to Call::DeliverRtpPacket
2023-03-29 06:36:17 +00:00
call_config.cc
Add parameter to control the pacer's burst outside of field trials.
2022-11-15 08:46:30 +00:00
call_config.h
Add parameter to control the pacer's burst outside of field trials.
2022-11-15 08:46:30 +00:00
call_factory.cc
Remove CoDel from webrtc::SimulatedNetwork.
2022-09-08 06:51:05 +00:00
call_factory.h
Delete Call dependency on ProcessThread as unused
2022-06-21 08:59:38 +00:00
call_perf_tests.cc
Ensure CallTest derived tests per default set min/max audio bitrate.
2023-01-26 17:36:01 +00:00
call_unittest.cc
Allow injecting packets of type Any to Call::DeliverRtpPacket
2023-03-29 06:36:17 +00:00
call.cc
Allow injecting packets of type Any to Call::DeliverRtpPacket
2023-03-29 06:36:17 +00:00
call.h
Delete Call dependency on ProcessThread as unused
2022-06-21 08:59:38 +00:00
degraded_call.cc
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
2023-01-25 18:18:29 +00:00
degraded_call.h
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
2023-01-25 18:18:29 +00:00
DEPS
SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated.
2021-08-30 10:20:55 +00:00
fake_network_pipe_unittest.cc
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
2023-01-25 18:18:29 +00:00
fake_network_pipe.cc
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
2023-01-25 18:18:29 +00:00
fake_network_pipe.h
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
2023-01-25 18:18:29 +00:00
flexfec_receive_stream_impl.cc
Remove rtp header extension from config of Call audio and video receivers
2023-01-31 11:58:43 +00:00
flexfec_receive_stream_impl.h
Remove rtp header extension from config of Call audio and video receivers
2023-01-31 11:58:43 +00:00
flexfec_receive_stream_unittest.cc
Remove rtp header extension from config of Call audio and video receivers
2023-01-31 11:58:43 +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 SetPayloadType to FlexfecReceiveStream.
2022-07-28 21:24:50 +00:00
OWNERS
Update OWNERS for call/
2022-06-03 12:01:46 +00:00
packet_receiver.h
Allow injecting packets of type Any to Call::DeliverRtpPacket
2023-03-29 06:36:17 +00:00
rampup_tests.cc
Stop overriding extensions in rampup tests
2023-01-25 13:18:49 +00:00
rampup_tests.h
Stop overriding extensions in rampup tests
2023-01-25 13:18:49 +00:00
receive_stream.h
Remove rtp header extension from config of Call audio and video receivers
2023-01-31 11:58:43 +00:00
receive_time_calculator_unittest.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2
2022-03-09 22:17:52 +00:00
receive_time_calculator.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
receive_time_calculator.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +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
Remove RTC_DISALLOW_COPY_AND_ASSIGN more.
2022-01-20 11:00:18 +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
Update old TODO comments
2022-07-05 09:59:33 +00:00
rtp_demuxer_unittest.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_demuxer.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_demuxer.h
Adopt absl::string_view in call/
2022-05-17 12:00:45 +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
Introduce EncodedImage.SimulcastIndex().
2023-02-15 15:02:57 +00:00
rtp_payload_params.cc
Reland "Make SimulcastIndex() and SpatialIndex() distinct (remove fallback)."
2023-02-21 18:30:35 +00:00
rtp_payload_params.h
For VP9 assume max number of spatial layers to simulate generic descriptor
2022-06-08 11:36:54 +00:00
rtp_stream_receiver_controller_interface.h
Demote RtpStreamReceiverController AddSink/RemoveSink to private
2022-07-06 09:31:54 +00:00
rtp_stream_receiver_controller.cc
Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
2022-12-22 14:04:21 +00:00
rtp_stream_receiver_controller.h
Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
2022-12-22 14:04:21 +00:00
rtp_transport_config.h
Add parameter to control the pacer's burst outside of field trials.
2022-11-15 08:46:30 +00:00
rtp_transport_controller_send_factory_interface.h
Delete Call dependency on ProcessThread as unused
2022-06-21 08:59:38 +00:00
rtp_transport_controller_send_factory.h
Refactor some config plumbing in call/.
2022-11-16 09:18:40 +00:00
rtp_transport_controller_send_interface.h
Replace TaskQueue with MaybeWorkerThread in RtpTransportControllerInterface
2022-10-10 11:56:52 +00:00
rtp_transport_controller_send.cc
Stop Posting tasks when we don't need to.
2023-03-06 15:13:39 +00:00
rtp_transport_controller_send.h
Stop Posting tasks when we don't need to.
2023-03-06 15:13:39 +00:00
rtp_video_sender_interface.h
Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream"
2022-12-02 12:03:25 +00:00
rtp_video_sender_unittest.cc
Introduce EncodedImage.SimulcastIndex().
2023-02-15 15:02:57 +00:00
rtp_video_sender.cc
Add a field trial string to make enable_retransmit_all_layers configurable.
2023-03-03 15:20:41 +00:00
rtp_video_sender.h
Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream"
2022-12-02 12:03:25 +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
Updated associated payload types without recreating receive streams.
2022-08-16 13:38:24 +00:00
rtx_receive_stream.h
Updated associated payload types without recreating receive streams.
2022-08-16 13:38:24 +00:00
simulated_network_unittest.cc
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
2022-11-06 13:14:26 +00:00
simulated_network.cc
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
2022-11-06 13:14:26 +00:00
simulated_network.h
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork."
2022-11-06 13:14:26 +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
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
version.cc
Update WebRTC code version (2023-04-06T04:04:31).
2023-04-06 05:47:24 +00:00
version.h
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
video_receive_stream.cc
Remove rtp header extension from config of Call audio and video receivers
2023-01-31 11:58:43 +00:00
video_receive_stream.h
Allow RTX ssrc to be updated on receive streams
2023-02-01 12:54:46 +00:00
video_send_stream.cc
Change the type of RTCVideoSourceStats.framesPerSecond
2021-11-16 11:21:41 +00:00
video_send_stream.h
Add scalability mode to RTCOutboundRtpStreamStats stats
2022-12-08 11:46:06 +00:00