webrtc_m130

History

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

java/src/org/webrtc/voiceengine

Temporary suppress bytebuffer warnings.

2018-04-20 11:45:28 +00:00

aaudio_player.cc

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

aaudio_player.h

Add support of AAudio in native WebRTC on Android O and above

2018-03-16 10:20:27 +00:00

aaudio_recorder.cc

Use absl::make_unique and absl::WrapUnique directly