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webrtc-version-updater c479e46618 Update WebRTC code version (2023-08-08T04:02:52). 
Bug: None
Change-Id: Id31d7cfb54c95986ae1cd156b80d4033b4e579b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315200
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@{#40522}
2023-08-08 05:26:14 +00:00
..
adaptation
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
test
Pass rtcp message to RtpTransportController through newer interface
2023-05-17 17:19:23 +00:00
audio_receive_stream.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
audio_receive_stream.h
Propagate time of the last received packet with Timestamp type
2023-06-02 14:29:19 +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
Replace "rcvd" with "received" for readability
2023-04-24 15:30:07 +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
Reland: FrameGeneratorCapturer: don't generate video before Start is called
2023-06-29 14:47:05 +00:00
BUILD.gn
stats: implement flexfec fecBytesReceived stats for FlexFEC
2023-06-21 13:04:31 +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
Introduce support for video packet batching.
2023-05-08 16:24:03 +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
Reland "Remove dependency of video_replay on TestADM."
2023-04-25 09:39:22 +00:00
call_unittest.cc
Use Timestamp type in RtpState struct
2023-04-27 11:24:38 +00:00
call.cc
Pass and process capture time through SendPacketObserver with Timestamp type
2023-06-22 17:16:41 +00:00
call.h
Delete Call dependency on ProcessThread as unused
2022-06-21 08:59:38 +00:00
degraded_call.cc
Migrate PostTask+Wait to BlockingCall to avoid deadlock in DegradedCall.
2023-04-19 09:40:33 +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
Use ArrayView versions of SendRtp and SendRtcp
2023-08-07 08:28:48 +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
stats: implement flexfec fecBytesReceived stats for FlexFEC
2023-06-21 13:04:31 +00:00
flexfec_receive_stream_impl.h
stats: implement flexfec fecBytesReceived stats for FlexFEC
2023-06-21 13:04:31 +00:00
flexfec_receive_stream_unittest.cc
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +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
stats: implement flexfec fecBytesReceived stats for FlexFEC
2023-06-21 13:04:31 +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
Reland "Remove dependency of video_replay on TestADM."
2023-04-25 09:39:22 +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
Introduce support for video packet batching.
2023-05-08 16:24:03 +00:00
rtp_demuxer_unittest.cc
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +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
…
rtp_payload_params_unittest.cc
When simulating chains from VP9 codec specific info support first_active_layer > 0
2023-08-03 13:19:00 +00:00
rtp_payload_params.cc
When simulating chains from VP9 codec specific info support first_active_layer > 0
2023-08-03 13:19:00 +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
Pass rtcp message to RtpTransportController through newer interface
2023-05-17 17:19:23 +00:00
rtp_transport_controller_send.cc
Reduce precision of RTT in RtrpTransportControllerSend
2023-05-19 10:38:37 +00:00
rtp_transport_controller_send.h
Cleanup RtcpReceiver from passing TransportFeedback via older interface
2023-05-22 10:14:04 +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
Add SetRetransmissionMode() to FecController, this will be used to control RTX settings in FecController.
2023-06-01 07:51:56 +00:00
rtp_video_sender.cc
Recreate the stream when switching from standard to legacy API.
2023-07-26 13:48:41 +00:00
rtp_video_sender.h
Add SetRetransmissionMode() to FecController, this will be used to control RTX settings in FecController.
2023-06-01 07:51:56 +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
…
syncable.h
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
version.cc
Update WebRTC code version (2023-08-08T04:02:52).
2023-08-08 05:26:14 +00:00
version.h
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
video_receive_stream.cc
Implement video versions of RTCInboundRtpStreamStats.jitterBuffer{Target,Minimum}Delay
2023-05-26 13:34:09 +00:00
video_receive_stream.h
Remove default "unknown" encoderImplementation/decoderImplementation
2023-06-22 11:49:58 +00:00
video_send_stream.cc
Cleanup usasge of ReportBlockData::report_block accessor
2023-05-05 09:56:30 +00:00
video_send_stream.h
Remove default "unknown" encoderImplementation/decoderImplementation
2023-06-22 11:49:58 +00:00