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webrtc_m130/audio
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Danil Chapovalov 3e39254b67 Pass rtcp message to RtpTransportController through newer interface 
NetworkLinkRtcpObserver is similar to RtcpBandwidthObserver but pass
time variables using unit types instead of raw integers.

Bug: webrtc:13757
Change-Id: Iaa0bbe0b108620b3a24013c40e7d9004032e904d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305022
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40087}
2023-05-17 17:19:23 +00:00
..
test
Reland "Remove dependency of video_replay on TestADM."
2023-04-25 09:39:22 +00:00
utility
Remove dependency on rtc_base_approved from most targets
2022-04-25 12:15:30 +00:00
voip
Add RtpRtcpInterface::LastRtt function to replace RtpRtcpInterface::RTT
2023-05-09 14:54:50 +00:00
audio_level.cc
Migrate audio/ to use webrtc::Mutex
2020-07-06 14:21:38 +00:00
audio_level.h
Migrate audio/ to use webrtc::Mutex
2020-07-06 14:21:38 +00:00
audio_receive_stream_unittest.cc
[SourceTracker] Move state to the worker thread, remove mutex.
2023-04-25 08:18:42 +00:00
audio_receive_stream.cc
[SourceTracker] Move state to the worker thread, remove mutex.
2023-04-25 08:18:42 +00:00
audio_receive_stream.h
Use SequenceChecker(SequenceChecker::kDetached) in a few places.
2023-03-24 07:44:18 +00:00
audio_send_stream_tests.cc
Reland "Remove dependency of video_replay on TestADM."
2023-04-25 09:39:22 +00:00
audio_send_stream_unittest.cc
Pass rtcp message to RtpTransportController through newer interface
2023-05-17 17:19:23 +00:00
audio_send_stream.cc
Simplify handling rtcp messages in audio send channel
2023-05-17 06:32:12 +00:00
audio_send_stream.h
Simplify handling rtcp messages in audio send channel
2023-05-17 06:32:12 +00:00
audio_state_unittest.cc
Implement support for Chrome task origin tracing. #3.5/4
2023-03-01 11:11:37 +00:00
audio_state.cc
Use SequenceChecker(SequenceChecker::kDetached) in a few places.
2023-03-24 07:44:18 +00:00
audio_state.h
Use SequenceChecker(SequenceChecker::kDetached) in a few places.
2023-03-24 07:44:18 +00:00
audio_transport_impl.cc
Make capture timestamp optional in ADM.
2023-01-23 17:29:06 +00:00
audio_transport_impl.h
Make capture timestamp optional in ADM.
2023-01-23 17:29:06 +00:00
BUILD.gn
Reland "Remove dependency of video_replay on TestADM."
2023-04-25 09:39:22 +00:00
channel_receive_frame_transformer_delegate_unittest.cc
Move rtc::make_ref_counted to api/
2022-06-15 09:47:38 +00:00
channel_receive_frame_transformer_delegate.cc
Add GetContributionSources to TransformableIncomingAudioFrame
2022-10-11 12:52:21 +00:00
channel_receive_frame_transformer_delegate.h
Use backticks not vertical bars to denote variables in comments for /audio
2021-07-27 15:36:40 +00:00
channel_receive_unittest.cc
Create unit test for the population of capture_start_ntp_time
2023-02-06 14:00:39 +00:00
channel_receive.cc
Add RtpRtcpInterface::LastRtt function to replace RtpRtcpInterface::RTT
2023-05-09 14:54:50 +00:00
channel_receive.h
Replace "rcvd" with "received" for readability
2023-04-24 15:30:07 +00:00
channel_send_frame_transformer_delegate_unittest.cc
Move rtc::make_ref_counted to api/
2022-06-15 09:47:38 +00:00
channel_send_frame_transformer_delegate.cc
Add a clone method to the audio frame transformer API.
2023-03-06 08:22:25 +00:00
channel_send_frame_transformer_delegate.h
Add a clone method to the audio frame transformer API.
2023-03-06 08:22:25 +00:00
channel_send_unittest.cc
Simplify handling rtcp messages in audio send channel
2023-05-17 06:32:12 +00:00
channel_send.cc
Pass rtcp message to RtpTransportController through newer interface
2023-05-17 17:19:23 +00:00
channel_send.h
Simplify handling rtcp messages in audio send channel
2023-05-17 06:32:12 +00:00
conversion.h
Make header files self contained.
2022-10-08 08:38:36 +00:00
DEPS
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +00:00
mock_voe_channel_proxy.h
Simplify handling rtcp messages in audio send channel
2023-05-17 06:32:12 +00:00
OWNERS
Add alessiob@webrtc.org in audio/OWNERS
2022-09-09 07:33:11 +00:00
remix_resample_unittest.cc
Clarify and extend test support for certain sample rates in audio processing
2022-08-03 14:26:36 +00:00
remix_resample.cc
Reland "Rename FATAL() into RTC_FATAL()."
2020-11-18 20:49:08 +00:00
remix_resample.h
Use backticks not vertical bars to denote variables in comments for /audio
2021-07-27 15:36:40 +00:00