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webrtc/audio
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Björn Terelius 77ffbd3099 Include-what-you-use api/rtc_event_log/ 
Bug: webrtc:42226242
Change-Id: I8802beb672e398c598728fc3bb5173bcdad16efc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354624
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42490}
2024-06-16 13:53:56 +00:00
..
test Move call/simulated_network to test/network 2024-04-29 09:55:06 +00:00
utility Add audio view classes 2024-05-24 18:08:37 +00:00
voip Create Environment for VoipCore 2024-06-11 10:49:19 +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 RTCP: implement reduced size RTCP for audio 2024-05-16 18:24:10 +00:00
audio_receive_stream.h RTCP: implement reduced size RTCP for audio 2024-05-16 18:24:10 +00:00
audio_receive_stream_unittest.cc RTCP: implement reduced size RTCP for audio 2024-05-16 18:24:10 +00:00
audio_send_stream.cc Pass Environment into audio ChannelSend 2024-05-29 07:05:05 +00:00
audio_send_stream.h Pass Environment into audio ChannelSend 2024-05-29 07:05:05 +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 Pass Environment into audio ChannelSend 2024-05-29 07:05:05 +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 Add audio view classes 2024-05-24 18:08:37 +00:00
audio_transport_impl.h Move webrtc::AudioDeviceModule include to api/ folder 2024-04-22 08:56:31 +00:00
BUILD.gn Reland "Run IWYU on some files I intend to work on" 2024-06-05 08:59:49 +00:00
channel_receive.cc Propagate arrival time inside NetEq 2024-05-30 14:21:42 +00:00
channel_receive.h RTCP: implement reduced size RTCP for audio 2024-05-16 18:24:10 +00:00
channel_receive_frame_transformer_delegate.cc Reland "Run IWYU on some files I intend to work on" 2024-06-05 08:59:49 +00:00
channel_receive_frame_transformer_delegate.h Reland "Run IWYU on some files I intend to work on" 2024-06-05 08:59:49 +00:00
channel_receive_frame_transformer_delegate_unittest.cc Reland "Run IWYU on some files I intend to work on" 2024-06-05 08:59:49 +00:00
channel_receive_unittest.cc Split out time_util to separate target ntp_time_util 2024-05-28 13:31:00 +00:00
channel_send.cc Include-what-you-use api/rtc_event_log/ 2024-06-16 13:53:56 +00:00
channel_send.h Pass Environment into audio ChannelSend 2024-05-29 07:05:05 +00:00
channel_send_frame_transformer_delegate.cc Add passkey to TransformableFrameInterface to prevent external impls 2024-05-16 13:12:51 +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 Reland "Run IWYU on some files I intend to work on" 2024-06-05 08:59:49 +00:00
channel_send_unittest.cc Pass Environment into audio ChannelSend 2024-05-29 07:05:05 +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 RTCP: implement reduced size RTCP for audio 2024-05-16 18:24:10 +00:00
OWNERS Add alessiob@webrtc.org in audio/OWNERS 2022-09-09 07:33:11 +00:00
remix_resample.cc Use std::array<> consistently for reusable audio buffers. 2024-05-29 09:20:36 +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