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[webrtc-provisional-stats] Add RTCInboundRtpStreamStats.jitterBufferTargetDelay (#19)

From: henbos via GitHub <sysbot+gh@w3.org>
Date: Tue, 03 Mar 2020 13:28:23 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issues.opened-574668101-1583242102-sysbot+gh@w3.org>
henbos has just created a new issue for https://github.com/w3c/webrtc-provisional-stats:

== Add RTCInboundRtpStreamStats.jitterBufferTargetDelay ==
In the webrtc.org's implementation of the audio jitter buffer, the jitter buffer has an actual delay ([jitterBufferDelay](https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferdelay)) and a target delay.

An implementation does not need to have a target, but for testing it is useful to be able to compare the actual value with the target value. This tells you something about how well samples get accelerated or decelerated.

Because this is a bit implementation-specific, and tells you more about the implementation's ability to deliver what it wants to deliver rather than direct measurements of achieved quality, this is added to the provisional stats spec for further evaluation.

Please view or discuss this issue at https://github.com/w3c/webrtc-provisional-stats/issues/19 using your GitHub account
Received on Tuesday, 3 March 2020 13:28:24 UTC

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