- From: Tim Panton new <thp@westhawk.co.uk>
- Date: Thu, 13 Feb 2014 14:59:23 +0000
- To: Iņaki Baz Castillo <ibc@aliax.net>
- Cc: "public-webrtc@w3.org" <public-webrtc@w3.org>
On 13 Feb 2014, at 14:43, Iņaki Baz Castillo <ibc@aliax.net> wrote: > 2014-02-13 15:38 GMT+01:00 Emil Ivov <emcho@jitsi.org>: >>> May I understand how the WebAudio API could be aware of RTP fields? >> >> I don't believe anyone is suggesting that CSRCs be surfaced through >> the WebAudio API. My understanding is that the current discussion is >> about whether or not such details could be bubbled through the WebRTC >> API (1.0). > > Right, the subject of this thread is "active speaker information in > mixed streams" so clearly we need the CSRC values inspection in client > side. That can only be achieved by the WebRTC API (and the WebAudio > API is totally useless for this subject). > You SIP guys are so funny, you insist that the benefit of SIP is that it decouples signalling from media, then you go adding more and more "media-meta-data" to the media channel, till it looks like a signalling channel. Sigh. The 'correct' solution to this is to re-engineer your mixer so it sends active speaker info over the data channel, not to further delay the standard by adding more VoIP specific legacy features. WebAudio cropped up, because I originally assumed we were discussing in an un-mixed peer (i.e. the prime p2p usecase of WebRTC) , the where peer browser could use webAudio to generate the relevant info and send it down the datachannel. T. > > > -- > Iņaki Baz Castillo > <ibc@aliax.net>
Received on Thursday, 13 February 2014 15:00:00 UTC