WebRTC Audio Codec and Processing Requirements
draft-ietf-rtcweb-audio-07
The information below is for an old version of the document | |||||
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Document | Type | Expired Internet-Draft (rtcweb WG) | |||
Authors | Jean-Marc Valin , Cary Bran | ||||
Last updated | 2015-04-27 (latest revision 2014-10-24) | ||||
Replaces | draft-cbran-rtcweb-codec | ||||
Stream | Internent Engineering Task Force (IETF) | ||||
Formats |
Expired & archived
pdf
htmlized (tools)
htmlized
bibtex
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Reviews | |||||
Additional Resources |
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Stream | WG state | WG Document | |||
Document shepherd | Cullen Jennings | ||||
Shepherd write-up | Show (last changed 2015-04-08) | ||||
IESG | IESG state | Expired | |||
Consensus Boilerplate | Unknown | ||||
Telechat date | |||||
Responsible AD | (None) | ||||
Send notices to | (None) |
This Internet-Draft is no longer active. A copy of
the expired Internet-Draft can be found at
https://www.ietf.org/archive/id/draft-ietf-rtcweb-audio-07.txt
https://www.ietf.org/archive/id/draft-ietf-rtcweb-audio-07.txt
Abstract
This document outlines the audio codec and processing requirements for WebRTC client application and endpoint devices.
Authors
Jean-Marc Valin
(jmvalin@jmvalin.ca)
Cary Bran
(cary.bran@plantronics.com)
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)