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webrtc_m130/audio
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Jakob Ivarsson 9986ff7f4a Use LastRtt from RTCP module in ChannelSend. 
It is a bit more flexible than the current implementation.

Also cleanup ChannelSend::GetRTT since it is not called from the receive stream anymore.

Bug: none
Change-Id: I4403c8b1840012f2287d189be934fd1069de85fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374160
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43728}
2025-01-14 01:42:25 -08:00
..
test
Replace gunit.h macros with WaitUntil in audio/
2025-01-13 02:24:11 -08:00
utility
Cleanup expired field trial WebRTC-VoIPChannelRemixingAdjustmentKillSwitch
2024-07-23 13:23:26 +00:00
voip
Format /audio folder
2025-01-07 23:43:38 -08:00
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
Cleanup AssociateSendStream for audio.
2025-01-10 05:12:40 -08:00
audio_receive_stream.cc
Cleanup AssociateSendStream for audio.
2025-01-10 05:12:40 -08:00
audio_receive_stream.h
Cleanup AssociateSendStream for audio.
2025-01-10 05:12:40 -08:00
audio_send_stream_tests.cc
Format /audio folder
2025-01-07 23:43:38 -08:00
audio_send_stream_unittest.cc
Format /audio folder
2025-01-07 23:43:38 -08:00
audio_send_stream.cc
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
audio_send_stream.h
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
audio_state_unittest.cc
Comment unused variables in implemented functions 8\n
2024-10-28 12:05:18 +00:00
audio_state.cc
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +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
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
audio_transport_impl.h
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
BUILD.gn
Use LastRtt from RTCP module in ChannelSend.
2025-01-14 01:42:25 -08:00
channel_receive_frame_transformer_delegate_unittest.cc
Pass receive_time through frame transformer
2024-08-02 07:01:33 +00:00
channel_receive_frame_transformer_delegate.cc
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
channel_receive_frame_transformer_delegate.h
Pass receive_time through frame transformer
2024-08-02 07:01:33 +00:00
channel_receive_unittest.cc
Avoid depending on codec info for audio jitter stat.
2024-11-21 18:47:29 +00:00
channel_receive.cc
Cleanup AssociateSendStream for audio.
2025-01-10 05:12:40 -08:00
channel_receive.h
Cleanup AssociateSendStream for audio.
2025-01-10 05:12:40 -08:00
channel_send_frame_transformer_delegate_unittest.cc
Comment unused variables in implemented functions 8\n
2024-10-28 12:05:18 +00:00
channel_send_frame_transformer_delegate.cc
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
channel_send_frame_transformer_delegate.h
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
channel_send_unittest.cc
Replace gunit.h macros with WaitUntil in audio/
2025-01-13 02:24:11 -08:00
channel_send.cc
Use LastRtt from RTCP module in ChannelSend.
2025-01-14 01:42:25 -08:00
channel_send.h
Use LastRtt from RTCP module in ChannelSend.
2025-01-14 01:42:25 -08:00
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
Use LastRtt from RTCP module in ChannelSend.
2025-01-14 01:42:25 -08:00
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
Remove PushResampler<T>::InitializeIfNeeded
2024-07-04 10:33:21 +00:00
remix_resample.h
Update RemixAndResample to use audio views
2024-07-03 09:52:24 +00:00