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[webrtc-pc] Add adaptivePtime to RTCConfiguration (#2310)

From: Elad Alon via GitHub <sysbot+gh@w3.org>
Date: Thu, 26 Sep 2019 10:30:56 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issues.opened-498800691-1569493855-sysbot+gh@w3.org>
eladalon1983 has just created a new issue for https://github.com/w3c/webrtc-pc:

== Add adaptivePtime to RTCConfiguration ==
Currently, there is no way to tell the browser that it should attempt to optimize the trade-off between audio latency and bandwidth consumption, by varying the length of audio frames dynamically during a call. It would be good to add this.

See [this document](https://docs.google.com/document/d/12sVXRog-Dl9c-hbMInz9obmODk_yft06z7GMgfpdjPw/edit?usp=sharing) for details.

Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/2310 using your GitHub account
Received on Thursday, 26 September 2019 10:30:58 UTC

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