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Re: [webrtc-pc] Add adaptivePtime to RTCConfiguration (#2309)

From: Elad Alon via GitHub <sysbot+gh@w3.org>
Date: Thu, 26 Sep 2019 10:39:59 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-535445753-1569494398-sysbot+gh@w3.org>
> Is there any sort of event or statistics that will inform what audio frame length change or will allow to detect what audio frame length is currently being used?
No. We can discuss adding that separately, I think?

> How would minimum and maximum supported audio frame lengths will be set or negotiated?
This is up for discussion. I suggest as a first approach to leave it up to the implementation, as it mostly is at the moment. Another possibility is to change adaptivePtime  to a dictionary that holds the allowed range.

> What is the benefit of this approach vs generating renegotiation needed event and changing ptime using signaling exchange?
Signaling and renegotiation take time and bandwidth. When the available bandwidth drops, both are at a premium. We allow an implementation to vary the target bitrate for video and audio dynamically, without renegotiation. The same rationale goes for varying the frame length, and the overhead it consumes, I believe.


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Received on Thursday, 26 September 2019 10:40:01 UTC

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