mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-12 21:30:45 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
webrtc/audio
History
Jakob Ivarsson 1e5f88c5be Make muted param in GetAudio optional. 
It is not necessary for the caller to use it and the mute info can be
found on AudioFrame.muted().

Bug: None
Change-Id: I458f1f2e8489c1d8f8a9078b21f889b2540bdab9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349940
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42236}
2024-05-06 18:07:34 +00:00
..
test Move call/simulated_network to test/network 2024-04-29 09:55:06 +00:00
utility Remove AudioFrameOperations::Add, ApplyHalfGain and Scale. 2024-05-02 19:39:20 +00:00
voip Move webrtc::AudioDeviceModule include to api/ folder 2024-04-22 08:56:31 +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 Log audio stream start/stop. 2023-12-12 10:43:47 +00:00
audio_receive_stream.h Use SequenceChecker(SequenceChecker::kDetached) in a few places. 2023-03-24 07:44:18 +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_send_stream.cc Move webrtc::AudioProcessing include to api/ folder 2024-04-20 07:02:50 +00:00
audio_send_stream.h Add option for the audio encoder to allocate a bitrate range. 2024-02-07 09:47:16 +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 Move webrtc::AudioProcessing include to api/ folder 2024-04-20 07:02:50 +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 Start using ArrayView in AudioFrame, update PushResampler 2024-04-30 15:33:08 +00:00
audio_transport_impl.h Move webrtc::AudioDeviceModule include to api/ folder 2024-04-22 08:56:31 +00:00
BUILD.gn Move call/simulated_network to test/network 2024-04-29 09:55:06 +00:00
channel_receive.cc Make muted param in GetAudio optional. 2024-05-06 18:07:34 +00:00
channel_receive.h Propagate time of the last received packet with Timestamp type 2023-06-02 14:29:19 +00:00
channel_receive_frame_transformer_delegate.cc Calculate the audio level of audio packets before encoded transforms 2024-04-29 15:14:25 +00:00
channel_receive_frame_transformer_delegate.h Cleanup usage of the rtc::TaskQueue in audio/ 2024-01-18 12:24:14 +00:00
channel_receive_frame_transformer_delegate_unittest.cc Calculate the audio level of audio packets before encoded transforms 2024-04-29 15:14:25 +00:00
channel_receive_unittest.cc Move webrtc::AudioDeviceModule include to api/ folder 2024-04-22 08:56:31 +00:00
channel_send.cc Calculate the audio level of audio packets before encoded transforms 2024-04-29 15:14:25 +00:00
channel_send.h Expose audio mimeType for insertable streams 2023-11-03 09:49:12 +00:00
channel_send_frame_transformer_delegate.cc Calculate the audio level of audio packets before encoded transforms 2024-04-29 15:14:25 +00:00
channel_send_frame_transformer_delegate.h Calculate the audio level of audio packets before encoded transforms 2024-04-29 15:14:25 +00:00
channel_send_frame_transformer_delegate_unittest.cc Calculate the audio level of audio packets before encoded transforms 2024-04-29 15:14:25 +00:00
channel_send_unittest.cc Calculate the audio level of audio packets before encoded transforms 2024-04-29 15:14:25 +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 Expose audio mimeType for insertable streams 2023-11-03 09:49:12 +00:00
OWNERS Add alessiob@webrtc.org in audio/OWNERS 2022-09-09 07:33:11 +00:00
remix_resample.cc Update AudioFrameOperations to require ArrayView 2024-04-30 21:26:56 +00:00
remix_resample.h Start using ArrayView in AudioFrame, update PushResampler 2024-04-30 15:33:08 +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