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Re: active speaker information in mixed streams

From: Tim Panton new <thp@westhawk.co.uk>
Date: Wed, 12 Feb 2014 09:42:36 +0000
Cc: Iņaki Baz Castillo <ibc@aliax.net>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Message-Id: <B4EF61E7-8C74-4265-BAED-6CBA858E018F@westhawk.co.uk>
To: Harald Alvestrand <hta@google.com>

On 11 Feb 2014, at 23:07, Harald Alvestrand <hta@google.com> wrote:

> 
> 
> 
> On Tue, Feb 11, 2014 at 11:55 PM, Iņaki Baz Castillo <ibc@aliax.net> wrote:
> 2014-02-07 14:54 GMT+01:00 Harald Alvestrand <harald@alvestrand.no>:
> > 1) It is only needed when talking to devices that use CSRC mapping. This
> > is a property set that belongs to the peerconnection abstraction
> > (probably on the receiving doohickey), not to the more generic
> > MediaStram connection.
> 
> If that "event" (or whatever) belongs to PeerConnection then the
> reference to the MediaStream and/or MediaStreamTrack should also be
> provided by the callback, so we end in the same case. The problem here
> (IMHO) is having all the stuff on a single object (PeerConnection). Of
> course I don't want to expose WebRTC related stuff into
> MediaStream(Track), but adding everything on top of PC is not the
> proper way to go (IMHO).
> 
> 
> Yes, that's why I want it to be on the RTPReceiver object.
> We already have one of those per MediaStreamTrack.
>  

Isn't this the sort of thing we should be delegating to the web-audio API ?
It is fully capable of doing this.

I'm kicking back against _any_ changes/enhancements that will delay the 1.0 spec.
The principle feedback I'm getting from web devs about why they won't use webRTC is (in order)

1) standard isn't stable yet
2) half the browsers don't support it
3) it is just a thing for voip apps

Anything we do that delays the standard by adding VoIP specific special cases just exacerbates 
these problems.


Tim.
Received on Wednesday, 12 February 2014 09:43:08 UTC

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