..
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