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webrtc_m130/audio
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Evan Shrubsole d9dd939d66 Move safe_minmax.h to webrtc namespace 
Bug: webrtc:42232595
Change-Id: Ia3d96dfe1b1c25b6cc21bbd99d24ded7461924cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378061
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43942}
2025-02-20 05:12:52 -08:00
..
test
Replace gunit.h macros with WaitUntil in audio/
2025-01-13 02:24:11 -08:00
utility
Move string_builder.h to webrtc namespace
2025-02-19 06:30:53 -08:00
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
Move string_builder.h to webrtc namespace
2025-02-19 06:30:53 -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
Add support for the Absolute Capture Timestamp extension to TransformableAudioFrameInterface
2025-02-14 01:27:19 -08:00
channel_receive_frame_transformer_delegate.cc
Add support for the Absolute Capture Timestamp extension to TransformableAudioFrameInterface
2025-02-14 01:27:19 -08:00
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
Move safe_minmax.h to webrtc namespace
2025-02-20 05:12:52 -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
Add support for the Absolute Capture Timestamp extension to TransformableAudioFrameInterface
2025-02-14 01:27:19 -08:00
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
Move string_builder.h to webrtc namespace
2025-02-19 06:30:53 -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