- From: Martin Thomson <martin.thomson@gmail.com>
- Date: Thu, 5 Sep 2013 10:31:41 -0700
- To: "public-webrtc@w3.org" <public-webrtc@w3.org>
There was a question about how to do simulcast on the call. Here's how it might be possible to do simulcast without additional API surface. 1. Acquire original stream containing one video track. 2. Clone the track and rescale it. 3. Assemble a new stream containing the original and the rescaled track. 4. Send the stream. 5. At the receiver, play the video stream. That's the user part, now for the under-the-covers stuff: I know we discussed the rendering of multiple video tracks in the past, but it's not possible to read the following documents and reach any sensible conclusions: http://dev.w3.org/2011/webrtc/editor/getusermedia.html http://www.w3.org/TR/html5/embedded-content-0.html#concept-media-load-resource What needs to happen in this case is to ensure that the two video tracks are folded together with the higher "quality" version being displayed and the lower "quality" version being used to fill in any gaps that might appear in the higher "quality" one. That depends on the <video> element being able to identify the tracks as being equivalent, and possibly being able to identify which is the higher quality. This is where something like the srcname proposal could be useful (http://tools.ietf.org/html/draft-westerlund-avtext-rtcp-sdes-srcname-02). The only missing piece is exposing metadata on tracks such that this behaviour is discoverable. Adding an attribute on tracks (srcname perhaps, arbaon), could provide a hook for triggering the folding behaviour I'm talking about.
Received on Thursday, 5 September 2013 17:32:09 UTC