- From: Varun Singh via GitHub <sysbot+gh@w3.org>
- Date: Wed, 08 Feb 2017 01:23:27 +0000
- To: public-webrtc-logs@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). -- 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
Received on Wednesday, 8 February 2017 01:23:33 UTC