admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/audio
History
Harald Alvestrand 93c9aa1914 Apply include-cleaner to call/ 
with downstream fixes.

Bug: webrtc:42226242
Change-Id: I88d7b5ffc1f86c01ea13948c27b4210d032f4190
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361360
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42921}
2024-09-03 07:51:03 +00:00
..
test
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
utility
Cleanup expired field trial WebRTC-VoIPChannelRemixingAdjustmentKillSwitch
2024-07-23 13:23:26 +00:00
voip
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +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_unittest.cc
Pass Clock and RtcEventLog as Environment into AudioReceiveStream
2024-08-02 11:58:23 +00:00
audio_receive_stream.cc
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
audio_receive_stream.h
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +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
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
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
Implement support for Chrome task origin tracing. #3.5/4
2023-03-01 11:11:37 +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
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
channel_receive_frame_transformer_delegate_unittest.cc
Pass receive_time through frame transformer
2024-08-02 07:01:33 +00:00
channel_receive_frame_transformer_delegate.cc
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
channel_receive_frame_transformer_delegate.h
Pass receive_time through frame transformer
2024-08-02 07:01:33 +00:00
channel_receive_unittest.cc
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
channel_receive.cc
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
channel_receive.h
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
channel_send_frame_transformer_delegate_unittest.cc
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
channel_send_frame_transformer_delegate.cc
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
channel_send_frame_transformer_delegate.h
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
channel_send_unittest.cc
Apply include-cleaner to call/
2024-09-03 07:51:03 +00:00
channel_send.cc
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
channel_send.h
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +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
Migrate absl::optional to std::optional
2024-09-02 12:16:47 +00:00
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