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Adds detection of audio glitches for playout on iOS.

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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}
This commit is contained in:
henrika
2017-06-13 16:11:08 +02:00
committed by Commit Bot
parent dea075c7a6
commit 33e4e65706
7 changed files with 102 additions and 10 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 close to 10ms as possible since it results in "clean" callback
// sequence without bursts of callbacks back to back.
// size is as an even multiple of 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).