- 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>
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