Re: [rtcweb] draft-ietf-rtcweb-rtp-usage: CSRC in the API?

I think this is very useful for a bunch of things in conferences. I'd like to see an API so the CSRC list for outgoing RTP can be set on a per track basis, and the API allows the JS to get the CSRC of most recently received RTP packets for a given PC-Track and also be nice to have a callback any time the received CSRC changes. 

On Feb 22, 2013, at 1:19 AM, Magnus Westerlund <magnus.westerlund@ericsson.com> wrote:

> WebRTC WG,
> 
> As editor of the RTP usage specification in RTCWEB WG, we have had a
> noted issue in our draft specification
> (https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/). In
> Section 12.5. of version 05 (Contributing Sources) we had the following:
> 
> (tbd: does the API need to provide the ability to add a CSRC list to
>   an outgoing packet? this is only useful if the sender is mixing
>   content)
> 
> This is clearly an API question. We intended to remove it. However, I
> like to hand it over to you in W3C to consider on the impact on the API
> this has.
> 
> From my personal view point this has two aspects:
> 
> Exposing when a received MediaStreamTrack is actually the mix (or
> switch) of other MediaStreamTracks, a mix performed by a WebRTC endpoint
> or RTP middlebox (Mixer). Applications that like to know who are
> currently seen or audible needs this information mapping. We also have
> specified an optional to support RTP header extension (RFC6465) that
> provide energy levels for each contributing source. If that is used,
> that information would be something an application would like to render
> somehow.
> 
> The other aspect is when an WebRTC endpoint mixes media to produce a new
> MediaStreamTrack, for example with the Web Audio API, then one need to
> consider if and how the CSRC list is populated.
> 
> Cheers
> 
> Magnus Westerlund
> 
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