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Revert "Adds detection of audio glitches for playout on iOS. "

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This reverts commit 33e4e65706c56f6df65bb4ceb07464f5ec4269ea.

Reason for revert: breaks https://build.chromium.org/p/client.webrtc/builders/iOS%20API%20Framework%20Builder

Original change's description:
> Adds detection of audio glitches for playout on iOS. 
> 
> Bug: b/38018041
> Change-Id: If6b53d3909a52333543c8aade500fd4c26b47255
> Reviewed-on: https://chromium-review.googlesource.com/522563
> Commit-Queue: Henrik Andreasson <henrika@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18570}

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

Change-Id: I3dd354d83a1f0ac1b5cab643147ae9c1672f342b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/38018041
Reviewed-on: https://chromium-review.googlesource.com/533533
Reviewed-by: Henrik Andreasson <henrika@webrtc.org>
Commit-Queue: Henrik Andreasson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18571}
This commit is contained in:
Henrik Andreasson
2017-06-13 15:20:52 +00:00
committed by Commit Bot
parent 33e4e65706
commit 6e286cba7e
7 changed files with 10 additions and 102 deletions
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4
webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSessionConfiguration.m
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@ -41,8 +41,8 @@ const double kRTCAudioSessionLowComplexitySampleRate = 16000.0;
// ~10.6667ms or 512 audio frames per buffer. The FineAudioBuffer instance will
// take care of any buffering required to convert between native buffers and
// buffers used by WebRTC. It is beneficial for the performance if the native
// size is as an even multiple of 10ms as possible since it results in "clean"
// callback sequence without bursts of callbacks back to back.
// size is as close to 10ms as possible since it results in "clean" callback
// sequence without bursts of callbacks back to back.
const double kRTCAudioSessionHighPerformanceIOBufferDuration = 0.01;
// Use a larger buffer size on devices with only one core (e.g. iPhone 4).