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Re: Bug in section 12.3

From: Stefan Håkansson LK <stefan.lk.hakansson@ericsson.com>
Date: Wed, 18 Dec 2013 08:40:30 +0000
To: Nils Ohlmeier <nohlmeier@mozilla.com>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Message-ID: <1447FA0C20ED5147A1AA0EF02890A64B1C42B73D@ESESSMB209.ericsson.se>
On 2013-12-17 22:05, Nils Ohlmeier wrote:
> Hello,
>
> I believe I found a bug in the WebRTC draft
> http://dev.w3.org/2011/webrtc/editor/webrtc.html
>
> In section 12.3 it says:
>
> ' A track in a |MediaStream|, received with an ||RTCPeerConnection|
> <http://dev.w3.org/2011/webrtc/editor/webrtc.html#idl-def-RTCPeerConnection>|,
> /MUST/ have its |readyState| attribute [GETUSERMEDIA
> <http://dev.w3.org/2011/webrtc/editor/webrtc.html#bib-GETUSERMEDIA>] set
> to |muted| until media data arrives.'
>
> But the enum for MediaStreamTrackState in
> http://www.w3.org/TR/mediacapture-streams/ does not have a state 'muted'.

Looking at 
http://dev.w3.org/2011/webrtc/editor/getusermedia.html#mediastreamtrack, 
I think what is meant is that the track should have the property "muted" 
being true until data arrives.

>
> I assume the sentence in the WebRTC draft should be changed to say "...
> set to to new until media data arrives."

I think it should be set to "live" actually - there is a source (the 
PeerConnection) attached.


Stefan

> Best regards
>    Nils Ohlmeier
>    WebRTC QE
>    Mozilla
Received on Wednesday, 18 December 2013 08:41:09 UTC

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