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

From: Varun Singh via GitHub <sysbot+gh@w3.org>
Date: Wed, 08 Feb 2017 01:23:27 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-278199957-1486517006-sysbot+gh@w3.org>
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).

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Received on Wednesday, 8 February 2017 01:23:33 UTC

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