Re: [webrtc-stats] Stat for audio acceleration/expand rate?

And another question about inserting and removing sample, when does 
this happen or under what conditions does it happen?

For example,
+ to bring audio and video in to sync?
+ when the jitterbuffer is going to underflow or overflow? I am 
assuming the overflow can only happen in the case of a fixed size 
jitter buffer or at the moment in time it cannot be expanded?
+ or when a low or high watermark in the jitter buffer is met, i.e., 
the application realises that it needs to slow down or speed up the 
processing, otherwise this will cause perceptible audio quality 
issues.

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Received on Wednesday, 15 February 2017 11:37:33 UTC