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webrtc_m130/audio
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Danil Chapovalov 853a407273 Revert "Migrate RemoteNtpTimeEstimator to more precise time representations" 
This reverts commit a154a15c978a0eae133d957dcad3581fd5f98c7b.

Reason for revert: breaks downstream tests

Original change's description:
> Migrate RemoteNtpTimeEstimator to more precise time representations
>
> Bug: webrtc:13757
> Change-Id: I880ab3cc6e4f72da587ae42ddca051332907c07f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261311
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36817}

Bug: webrtc:13757
Change-Id: Id21edb1378e6e944b24955396250ddc33fa70663
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261722
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36819}
2022-05-09 15:48:29 +00:00
..
test
Remove typing detection
2022-03-23 10:23:54 +00:00
utility
Remove dependency on rtc_base_approved from most targets
2022-04-25 12:15:30 +00:00
voip
Revert "Migrate RemoteNtpTimeEstimator to more precise time representations"
2022-05-09 15:48:29 +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
Remove config() getter from AudioReceiveStream().
2022-05-09 11:21:44 +00:00
audio_receive_stream.cc
Remove config() getter from AudioReceiveStream().
2022-05-09 11:21:44 +00:00
audio_receive_stream.h
Separate reading remote_ssrc from using the rtp_config() getter.
2022-05-09 14:55:00 +00:00
audio_send_stream_tests.cc
Default enable sending transport sequence numbers on audio packets.
2020-11-24 09:19:54 +00:00
audio_send_stream_unittest.cc
Remove typing detection
2022-03-23 10:23:54 +00:00
audio_send_stream.cc
Update audio/, media/, and video/ to not use implicit conversion
2022-04-21 09:00:14 +00:00
audio_send_stream.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
audio_state_unittest.cc
Replace new rtc::RefCountedObject with rtc::make_ref_counted in a few files
2021-04-23 12:04:39 +00:00
audio_state.cc
Update audio/, media/, and video/ to not use implicit conversion
2022-04-21 09:00:14 +00:00
audio_state.h
Remove typing detection
2022-03-23 10:23:54 +00:00
audio_transport_impl.cc
Reland "Reland "Remove unused APM voice activity detection sub-module""
2022-02-16 08:41:30 +00:00
audio_transport_impl.h
Remove typing detection
2022-03-23 10:23:54 +00:00
BUILD.gn
Remove dependency on rtc_base_approved from most targets
2022-04-25 12:15:30 +00:00
channel_receive_frame_transformer_delegate_unittest.cc
Replace new rtc::RefCountedObject with rtc::make_ref_counted in a few files
2021-04-23 12:04:39 +00:00
channel_receive_frame_transformer_delegate.cc
Update audio code to not use implicit T* --> scoped_refptr<T> conversion
2022-01-13 15:49:49 +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.cc
Revert "Migrate RemoteNtpTimeEstimator to more precise time representations"
2022-05-09 15:48:29 +00:00
channel_receive.h
Reland "Wire up non-sender RTT for audio, and implement related standardized stats."
2021-09-06 14:26:55 +00:00
channel_send_frame_transformer_delegate_unittest.cc
Replace new rtc::RefCountedObject with rtc::make_ref_counted in a few files
2021-04-23 12:04:39 +00:00
channel_send_frame_transformer_delegate.cc
Update audio code to not use implicit T* --> scoped_refptr<T> conversion
2022-01-13 15:49:49 +00:00
channel_send_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_send.cc
Delete rtc_base/format_macros.h
2022-05-09 12:03:21 +00:00
channel_send.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
conversion.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
DEPS
Cleanup of bwe_defines.h
2020-11-26 12:26:02 +00:00
mock_voe_channel_proxy.h
Implement RTCOutboundRtpStreamStats.targetBitrate for audio.
2021-11-12 09:24:34 +00:00
null_audio_poller.cc
Use backticks not vertical bars to denote variables in comments for /audio
2021-07-27 15:36:40 +00:00
null_audio_poller.h
Use SequenceChecker from public API
2021-02-10 15:04:55 +00:00
OWNERS
Add jakobi@webrtc.org to audio/OWNERS
2021-08-19 15:30:32 +00:00
remix_resample_unittest.cc
Delete rtc_base/format_macros.h
2022-05-09 12:03:21 +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