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webrtc/audio
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Artem Titov 8a9f3a8f53 Reland "Remove dependency of video_replay on TestADM." 
This reverts commit f9e3bdd2ce.

Reason for revert: reland with fix

Original change's description:
> Revert "Remove dependency of video_replay on TestADM."
>
> This reverts commit 01716663a9.
>
> Reason for revert:  breaking CallPerfTest
> https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview 
>
> Original change's description:
> > Remove dependency of video_replay on TestADM.
> >
> > This should remove requirement to build TestADM in chromium build.
> >
> > Bug: b/272350185, webrtc:15081
> > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Artem Titov <titovartem@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39934}
>
> Bug: b/272350185, webrtc:15081
> Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Owners-Override: Jeremy Leconte <jleconte@google.com>
> Commit-Queue: Jeremy Leconte <jleconte@google.com>
> Cr-Commit-Position: refs/heads/main@{#39939}

Bug: b/272350185, webrtc:15081
Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Auto-Submit: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:39:22 +00:00
..
test Reland "Remove dependency of video_replay on TestADM." 2023-04-25 09:39:22 +00:00
utility Remove dependency on rtc_base_approved from most targets 2022-04-25 12:15:30 +00:00
voip Cleanup ReportBlockData class: use Timestamp and TimeDelta 2023-04-13 08:51:12 +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 [SourceTracker] Move state to the worker thread, remove mutex. 2023-04-25 08:18:42 +00:00
audio_receive_stream.h Use SequenceChecker(SequenceChecker::kDetached) in a few places. 2023-03-24 07:44:18 +00:00
audio_receive_stream_unittest.cc [SourceTracker] Move state to the worker thread, remove mutex. 2023-04-25 08:18:42 +00:00
audio_send_stream.cc Replace "rcvd" with "received" for readability 2023-04-24 15:30:07 +00:00
audio_send_stream.h Reland "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream" 2023-04-12 13:39:05 +00:00
audio_send_stream_tests.cc Reland "Remove dependency of video_replay on TestADM." 2023-04-25 09:39:22 +00:00
audio_send_stream_unittest.cc Reland "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream" 2023-04-12 13:39:05 +00:00
audio_state.cc Use SequenceChecker(SequenceChecker::kDetached) in a few places. 2023-03-24 07:44:18 +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 Make capture timestamp optional in ADM. 2023-01-23 17:29:06 +00:00
audio_transport_impl.h Make capture timestamp optional in ADM. 2023-01-23 17:29:06 +00:00
BUILD.gn Reland "Remove dependency of video_replay on TestADM." 2023-04-25 09:39:22 +00:00
channel_receive.cc Replace "rcvd" with "received" for readability 2023-04-24 15:30:07 +00:00
channel_receive.h Replace "rcvd" with "received" for readability 2023-04-24 15:30:07 +00:00
channel_receive_frame_transformer_delegate.cc Add GetContributionSources to TransformableIncomingAudioFrame 2022-10-11 12:52:21 +00:00
channel_receive_frame_transformer_delegate.h Use backticks not vertical bars to denote variables in comments for /audio 2021-07-27 15:36:40 +00:00
channel_receive_frame_transformer_delegate_unittest.cc Move rtc::make_ref_counted to api/ 2022-06-15 09:47:38 +00:00
channel_receive_unittest.cc Create unit test for the population of capture_start_ntp_time 2023-02-06 14:00:39 +00:00
channel_send.cc Replace "rcvd" with "received" for readability 2023-04-24 15:30:07 +00:00
channel_send.h Replace "rcvd" with "received" for readability 2023-04-24 15:30:07 +00:00
channel_send_frame_transformer_delegate.cc Add a clone method to the audio frame transformer API. 2023-03-06 08:22:25 +00:00
channel_send_frame_transformer_delegate.h Add a clone method to the audio frame transformer API. 2023-03-06 08:22:25 +00:00
channel_send_frame_transformer_delegate_unittest.cc Move rtc::make_ref_counted to api/ 2022-06-15 09:47:38 +00:00
channel_send_unittest.cc Update RTP timestamp based on capture timestamp when audio send stream is resumed. 2023-01-27 15:46:32 +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 Implement RTCOutboundRtpStreamStats.targetBitrate for audio. 2021-11-12 09:24:34 +00:00
OWNERS Add alessiob@webrtc.org in audio/OWNERS 2022-09-09 07:33:11 +00:00
remix_resample.cc Reland "Rename FATAL() into RTC_FATAL()." 2020-11-18 20:49:08 +00:00
remix_resample.h Use backticks not vertical bars to denote variables in comments for /audio 2021-07-27 15:36:40 +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