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webrtc/audio
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Lionel Koenig ec38238af7 Ensure the AudioCodingModule is reset when sending is stopped. 
Bug: webrtc:42226041
Change-Id: Ife3548bda3042a7447b7c50f48f023a2bc0bc443
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362103
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43017}
2024-09-12 22:47:11 +00:00
..
test Move call/simulated_network to test/network 2024-04-29 09:55:06 +00:00
utility Cleanup expired field trial WebRTC-VoIPChannelRemixingAdjustmentKillSwitch 2024-07-23 13:23:26 +00:00
voip Migrate absl::optional to std::optional 2024-09-02 12:16:47 +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.cc Reland "Return audio stats regarless if we have a codec." 2024-09-06 08:25:36 +00:00
audio_receive_stream.h Move thread handling from source tracker. 2024-09-05 08:45:11 +00:00
audio_receive_stream_unittest.cc Move thread handling from source tracker. 2024-09-05 08:45:11 +00:00
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_send_stream_tests.cc Expose AudioLevel as an absl::optional struct in api/rtp_headers.h 2024-03-22 10:07:47 +00:00
audio_send_stream_unittest.cc Migrate absl::optional to std::optional 2024-09-02 12:16:47 +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_state_unittest.cc Implement support for Chrome task origin tracing. #3.5/4 2023-03-01 11:11:37 +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 Remove use of AcmReceiver in ChannelReceive 2024-09-06 12:47:36 +00:00
channel_receive.cc Remove use of AcmReceiver in ChannelReceive 2024-09-06 12:47:36 +00:00
channel_receive.h Move thread handling from source tracker. 2024-09-05 08:45:11 +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_frame_transformer_delegate_unittest.cc Pass receive_time through frame transformer 2024-08-02 07:01:33 +00:00
channel_receive_unittest.cc Migrate absl::optional to std::optional 2024-09-02 12:16:47 +00:00
channel_send.cc Ensure the AudioCodingModule is reset when sending is stopped. 2024-09-12 22:47:11 +00:00
channel_send.h Migrate absl::optional to std::optional 2024-09-02 12:16:47 +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_frame_transformer_delegate_unittest.cc Migrate absl::optional to std::optional 2024-09-02 12:16:47 +00:00
channel_send_unittest.cc Ensure the AudioCodingModule is reset when sending is stopped. 2024-09-12 22:47:11 +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 Move thread handling from source tracker. 2024-09-05 08:45:11 +00:00
OWNERS Add alessiob@webrtc.org in audio/OWNERS 2022-09-09 07:33:11 +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
remix_resample_unittest.cc Clarify and extend test support for certain sample rates in audio processing 2022-08-03 14:26:36 +00:00