admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/audio
History
Tomas Lundqvist 64e8e64e80 Revert "Reland "Return audio stats regarless if we have a codec."" 
This reverts commit 4334cdfc5c0619a5f06125ea1f039cb123ccf21e.

Reason for revert: Breaks downstream project.

Original change's description:
> Reland "Return audio stats regarless if we have a codec."
>
> This is a reland of commit 7fff587a096c6ef40f5601f47ef50b221b3a4abf
>
> Original change's description:
> > Return audio stats regarless if we have a codec.
> >
> > Bug: b/331602608
> > Change-Id: I2d12a3ed83645fe1e7cbd8950fd86d5ba2d7c94d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361743
> > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> > Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#42964}
>
> Bug: b/331602608
> Change-Id: I95c89e7059005bc8dd8569ef41bfe9e863b4082f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361762
> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42969}

Bug: b/331602608
Change-Id: Ifbe332a8749d024f603b75b6e787551dd6762dd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363001
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43059}
2024-09-20 10:38:19 +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
Remove all use of AcmReceiver from WebRTC
2024-09-16 08:49:25 +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
Move thread handling from source tracker.
2024-09-05 08:45:11 +00:00
audio_receive_stream.cc
Revert "Reland "Return audio stats regarless if we have a codec.""
2024-09-20 10:38:19 +00:00
audio_receive_stream.h
Move thread handling from source tracker.
2024-09-05 08:45:11 +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
Remove use of AcmReceiver in ChannelReceive
2024-09-06 12:47:36 +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
Remove use of AcmReceiver in ChannelReceive
2024-09-06 12:47:36 +00:00
channel_receive.h
Move thread handling from source tracker.
2024-09-05 08:45:11 +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
Ensure the AudioCodingModule is reset when sending is stopped.
2024-09-12 22:47:11 +00:00
channel_send.cc
Ensure the AudioCodingModule is reset when sending is stopped.
2024-09-12 22:47:11 +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
Move thread handling from source tracker.
2024-09-05 08:45:11 +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