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test
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Move call/simulated_network to test/network
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2024-04-29 09:55:06 +00:00 |
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utility
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Remove AudioFrameOperations::Add, ApplyHalfGain and Scale.
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2024-05-02 19:39:20 +00:00 |
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voip
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Move webrtc::AudioDeviceModule include to api/ folder
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2024-04-22 08:56:31 +00:00 |
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audio_level.cc
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Update to WebRTC 4389 (e7d9f74)
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2021-04-16 13:26:31 -07:00 |
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audio_level.h
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Update to WebRTC 4389 (e7d9f74)
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2021-04-16 13:26:31 -07:00 |
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audio_receive_stream.cc
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Merge branch m122
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2024-02-14 22:44:28 -08:00 |
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audio_receive_stream.h
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Add Rust_setIncomingAudioMuted
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2023-09-27 12:16:54 -04:00 |
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audio_receive_stream_unittest.cc
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[SourceTracker] Move state to the worker thread, remove mutex.
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2023-04-25 08:18:42 +00:00 |
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audio_send_stream.cc
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Merge remote-tracking branch 'google/branch-heads/6478'
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2024-06-21 16:31:45 -07:00 |
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audio_send_stream.h
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Merge m123/6312
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2024-06-12 22:25:35 -07:00 |
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audio_send_stream_tests.cc
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Expose AudioLevel as an absl::optional struct in api/rtp_headers.h
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2024-03-22 10:07:47 +00:00 |
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audio_send_stream_unittest.cc
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Move webrtc::AudioProcessing include to api/ folder
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2024-04-20 07:02:50 +00:00 |
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audio_state.cc
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Merge remote-tracking branch 'google/branch-heads/6478'
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2024-06-21 16:31:45 -07:00 |
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audio_state.h
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Disable audio and media flow by default
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2023-09-12 18:03:28 -07:00 |
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audio_state_unittest.cc
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Implement support for Chrome task origin tracing. #3.5/4
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2023-03-01 11:11:37 +00:00 |
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audio_transport_impl.cc
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Start using ArrayView in AudioFrame, update PushResampler
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2024-04-30 15:33:08 +00:00 |
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audio_transport_impl.h
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Move webrtc::AudioDeviceModule include to api/ folder
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2024-04-22 08:56:31 +00:00 |
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BUILD.gn
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Move call/simulated_network to test/network
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2024-04-29 09:55:06 +00:00 |
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channel_receive.cc
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m126 merge fixes
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2024-06-25 14:25:19 -07:00 |
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channel_receive.h
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Add Rust_setIncomingAudioMuted
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2023-09-27 12:16:54 -04:00 |
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channel_receive_frame_transformer_delegate.cc
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Calculate the audio level of audio packets before encoded transforms
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2024-04-29 15:14:25 +00:00 |
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channel_receive_frame_transformer_delegate.h
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Cleanup usage of the rtc::TaskQueue in audio/
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2024-01-18 12:24:14 +00:00 |
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channel_receive_frame_transformer_delegate_unittest.cc
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Calculate the audio level of audio packets before encoded transforms
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2024-04-29 15:14:25 +00:00 |
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channel_receive_unittest.cc
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Merge remote-tracking branch 'google/branch-heads/6478'
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2024-06-21 16:31:45 -07:00 |
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channel_send.cc
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Calculate the audio level of audio packets before encoded transforms
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2024-04-29 15:14:25 +00:00 |
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channel_send.h
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Expose audio mimeType for insertable streams
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2023-11-03 09:49:12 +00:00 |
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channel_send_frame_transformer_delegate.cc
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Calculate the audio level of audio packets before encoded transforms
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2024-04-29 15:14:25 +00:00 |
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channel_send_frame_transformer_delegate.h
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Calculate the audio level of audio packets before encoded transforms
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2024-04-29 15:14:25 +00:00 |
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channel_send_frame_transformer_delegate_unittest.cc
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Calculate the audio level of audio packets before encoded transforms
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2024-04-29 15:14:25 +00:00 |
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channel_send_unittest.cc
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Calculate the audio level of audio packets before encoded transforms
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2024-04-29 15:14:25 +00:00 |
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conversion.h
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Make header files self contained.
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2022-10-08 08:38:36 +00:00 |
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DEPS
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pc: Add asynchronous RtpSender::SetParameters() call
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2022-11-15 15:31:40 +00:00 |
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mock_voe_channel_proxy.h
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Test fixes
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2024-06-11 10:30:46 -04:00 |
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OWNERS
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Add alessiob@webrtc.org in audio/OWNERS
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2022-09-09 07:33:11 +00:00 |
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remix_resample.cc
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Update AudioFrameOperations to require ArrayView
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2024-04-30 21:26:56 +00:00 |
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remix_resample.h
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Start using ArrayView in AudioFrame, update PushResampler
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2024-04-30 15:33:08 +00:00 |
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remix_resample_unittest.cc
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Clarify and extend test support for certain sample rates in audio processing
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2022-08-03 14:26:36 +00:00 |
