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webrtc_m130/modules/audio_device
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Patrik Höglund e2924d555d Revert "Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC." 
This reverts commit f217903a67995496a1d67674d77d5f237772b01b.

Reason for revert: Breaks downstream tests

Original change's description:
> Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC.
> 
> Also ensures that audio parameters are accessed atomically.
> 
> Bug: b/113648245
> Change-Id: Ic812bfe2b2c4cfb3b00d9d411bb4986dfeda1028
> Reviewed-on: https://webrtc-review.googlesource.com/97331
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24550}

TBR=henrika@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org

Change-Id: I620406f25762cf76db0470b3b29b50bc146935c7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/113648245
Reviewed-on: https://webrtc-review.googlesource.com/97941
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24569}
2018-09-05 08:52:51 +00:00
..
android
Revert "Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC."
2018-09-05 08:52:51 +00:00
dummy
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
include
Revert "Reland "Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.""
2018-09-04 13:04:11 +00:00
ios
Obj-C SDK Cleanup
2018-08-30 10:42:41 +00:00
linux
Stop using typeof() GNU extension
2018-08-20 08:15:13 +00:00
mac
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
win
Removing -Wno-reorder from audio_device.
2018-07-12 08:37:45 +00:00
audio_device_buffer.cc
Revert "Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC."
2018-09-05 08:52:51 +00:00
audio_device_buffer.h
Revert "Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC."
2018-09-05 08:52:51 +00:00
audio_device_config.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_device_data_observer.cc
AudioTransport: Remove PushCaptureData() method
2018-04-04 08:04:09 +00:00
audio_device_generic.cc
Removes Set/GetLoudspeakerStatus APIs from the ADM.
2017-11-16 19:44:24 +00:00
audio_device_generic.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_device_impl.cc
Adds support for new Windows ADM with limited API support.
2018-06-08 14:44:38 +00:00
audio_device_impl.h
Removes usage of AGC APIs in the ADM.
2017-12-13 16:32:21 +00:00
audio_device_name.cc
Ports CoreAudioUtil from Chrome to WebRTC.
2018-05-18 10:25:26 +00:00
audio_device_name.h
Ports CoreAudioUtil from Chrome to WebRTC.
2018-05-18 10:25:26 +00:00
audio_device_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
BUILD.gn
Remove check_includes=false from audio_device_impl.
2018-09-04 10:09:42 +00:00
DEPS
Obj-C SDK Cleanup
2018-08-30 10:42:41 +00:00
fine_audio_buffer_unittest.cc
Adds stereo support to FineAudioBuffer for mobile platforms.
2018-04-24 11:58:54 +00:00
fine_audio_buffer.cc
Adds support for new Windows ADM with limited API support.
2018-06-08 14:44:38 +00:00
fine_audio_buffer.h
Adds stereo support to FineAudioBuffer for mobile platforms.
2018-04-24 11:58:54 +00:00
mock_audio_device_buffer.h
Remove unused members from AudioDeviceBuffer
2018-01-16 10:20:32 +00:00
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