Re: [webrtc-stats] Stat for audio playout delay

I agree with @alvestrand that the delay measurements need exact 
details on when the clock starts and when the clock stops. There are 
several components in this calculation: the more difficult one is 
soundcard delay, although the algorithmic and jitter buffer can be 
measured very accurately. 

I would start with something that defines the sum of the playout 
algorithm+jitter buffer. Both the current (which could be the last 
played sample or the delay if there were a current sample to play) and
 a long-term total (based on each frame).

-- 
GitHub Notification of comment by vr000m
Please view or discuss this issue at 
https://github.com/w3c/webrtc-stats/issues/151#issuecomment-278199957 
using your GitHub account