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webrtc-version-updater 818163b939 Update WebRTC code version (2022-08-08T04:02:58). 
Bug: None
Change-Id: If61b7921fad21309cb298ecd2e4266e20593c51c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270861
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#37706}
2022-08-08 06:42:45 +00:00
..
adaptation
Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable
2022-07-20 08:15:08 +00:00
test
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audio_receive_stream.cc
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audio_receive_stream.h
Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay
2022-07-20 09:14:03 +00:00
audio_send_stream.cc
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audio_send_stream.h
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audio_sender.h
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audio_state.cc
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audio_state.h
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bitrate_allocator_unittest.cc
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bitrate_allocator.cc
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bitrate_allocator.h
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bitrate_estimator_tests.cc
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BUILD.gn
Migrate call/ to absl::AnyInvocable based TaskQueueBase interface
2022-07-07 10:32:26 +00:00
call_config.cc
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call_config.h
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call_factory.cc
Remove legacy WebRTCFakeNetwork field trials.
2022-07-27 07:33:15 +00:00
call_factory.h
Delete Call dependency on ProcessThread as unused
2022-06-21 08:59:38 +00:00
call_perf_tests.cc
Migrate call/ to absl::AnyInvocable based TaskQueueBase interface
2022-07-07 10:32:26 +00:00
call_unittest.cc
Delete Call dependency on ProcessThread as unused
2022-06-21 08:59:38 +00:00
call.cc
Allow recursive check for RTC_DCHECK_RUN_ON macro
2022-07-26 09:27:23 +00:00
call.h
Delete Call dependency on ProcessThread as unused
2022-06-21 08:59:38 +00:00
degraded_call.cc
Migrate call/ to absl::AnyInvocable based TaskQueueBase interface
2022-07-07 10:32:26 +00:00
degraded_call.h
Move to_queued_task.h and pending_task_safety_flag.h into public API
2022-06-17 09:20:39 +00:00
DEPS
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fake_network_pipe_unittest.cc
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fake_network_pipe.cc
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fake_network_pipe.h
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flexfec_receive_stream_impl.cc
Add SetPayloadType to FlexfecReceiveStream.
2022-07-28 21:24:50 +00:00
flexfec_receive_stream_impl.h
Add SetPayloadType to FlexfecReceiveStream.
2022-07-28 21:24:50 +00:00
flexfec_receive_stream_unittest.cc
test: fix flexfec test
2022-07-06 10:37:19 +00:00
flexfec_receive_stream.cc
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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
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rampup_tests.cc
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rampup_tests.h
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receive_stream.h
Add SetTransportCc to ReceiveStreamInterface.
2022-05-30 14:07:04 +00:00
receive_time_calculator_unittest.cc
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receive_time_calculator.cc
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receive_time_calculator.h
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rtp_bitrate_configurator_unittest.cc
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rtp_bitrate_configurator.cc
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rtp_bitrate_configurator.h
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rtp_config.cc
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rtp_config.h
Update old TODO comments
2022-07-05 09:59:33 +00:00
rtp_demuxer_unittest.cc
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rtp_demuxer.cc
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rtp_demuxer.h
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rtp_packet_sink_interface.h
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rtp_payload_params_unittest.cc
For VP9 assume max number of spatial layers to simulate generic descriptor
2022-06-08 11:36:54 +00:00
rtp_payload_params.cc
For VP9 assume max number of spatial layers to simulate generic descriptor
2022-06-08 11:36:54 +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
Demote RtpStreamReceiverController AddSink/RemoveSink to private
2022-07-06 09:31:54 +00:00
rtp_stream_receiver_controller.h
Demote RtpStreamReceiverController AddSink/RemoveSink to private
2022-07-06 09:31:54 +00:00
rtp_transport_config.h
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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
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rtp_transport_controller_send_interface.h
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rtp_transport_controller_send.cc
Remove unused field trial kill-switch WebRTC-LazyPacerStart.
2022-07-07 11:21:55 +00:00
rtp_transport_controller_send.h
Update/delete old TODOs
2022-07-06 07:49:04 +00:00
rtp_video_sender_interface.h
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rtp_video_sender_unittest.cc
clean up misc TimeDelta use
2022-08-02 13:52:36 +00:00
rtp_video_sender.cc
For VP9 assume max number of spatial layers to simulate generic descriptor
2022-06-08 11:36:54 +00:00
rtp_video_sender.h
For VP9 assume max number of spatial layers to simulate generic descriptor
2022-06-08 11:36:54 +00:00
rtx_receive_stream_unittest.cc
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rtx_receive_stream.cc
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rtx_receive_stream.h
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simulated_network_unittest.cc
clean up misc TimeDelta use
2022-08-02 13:52:36 +00:00
simulated_network.cc
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simulated_network.h
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simulated_packet_receiver.h
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syncable.cc
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syncable.h
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version.cc
Update WebRTC code version (2022-08-08T04:02:58).
2022-08-08 06:42:45 +00:00
version.h
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video_receive_stream.cc
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video_receive_stream.h
Make nack history configurable.
2022-08-05 22:58:43 +00:00
video_send_stream.cc
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video_send_stream.h
RtpSenderInterface::SetEncoderSelector
2022-06-08 11:06:52 +00:00