[webrtc-stats] insertedSamplesForDeceleration and removedSamplesForAcceleration of inbound stats confusing (#809)

hamishwillee has just created a new issue for https://github.com/w3c/webrtc-stats:

== insertedSamplesForDeceleration and removedSamplesForAcceleration of inbound stats confusing ==
I'm working on the docs for MDN in https://github.com/mdn/content/pull/40495, and the specification isn't very clear (to me):

> When playout is slowed down, this counter is increased by the difference between the number of samples received and the number of samples played out. If playout is slowed down by inserting samples, this will be the number of inserted samples.

When is playout slowed down and when is it sped up - i.e. what is the context here.
This might imply that we're choosing to slow the audio that people hear, which seems unlikely. 
What I think it means is that the playout of the _jitter buffer_ is slowed down because we're not getting enough samples. To make the audio appear like it isn't being affected we're inserting synthesized samples. Is that right?

But what are recipients supposed to do with this information? Look at the rate of this changing and use that as a measure of link quality or something? 

Ditto for `removedSamplesForAcceleration` - what could cause more samples than we can play out? What are users supposed to do with the information? 

There was some attempt to ask these questions in https://github.com/w3c/webrtc-stats/issues/146#issuecomment-279699345 and  https://github.com/w3c/webrtc-stats/issues/151#issuecomment-280629504)., but I don't think there were answers.



Please view or discuss this issue at https://github.com/w3c/webrtc-stats/issues/809 using your GitHub account


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Received on Monday, 28 July 2025 06:28:12 UTC