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

From: hlundin via GitHub <sysbot+gh@w3.org>
Date: Fri, 17 Feb 2017 11:31:48 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-280626654-1487331106-sysbot+gh@w3.org>
The terms "jitter buffer delay" and "decode buffer delay" can be 
interpreted differently depending on implementation details. For 
instance, given the implementation in NetEq (webrtc.org), I would not 
define decode buffer delay like @vr000m above, and the two are very 
tightly coupled.

I would suggest to have one jitter buffer delay, defined as the delay 
between the first packet belonging to an audio/video frame entering 
the jitter buffer and the complete frame being sent for rendering.

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Received on Friday, 17 February 2017 11:31:54 UTC

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