- From: Ralph Giles <giles@thaumas.net>
- Date: Mon, 11 Jul 2011 16:47:13 -0700
- To: Ian Hickson <ian@hixie.ch>
- Cc: Anant Narayanan <anant@mozilla.com>, public-webrtc@w3.org
On 11 July 2011 15:09, Ian Hickson <ian@hixie.ch> wrote: > One of the differences is that your proposal allows the author to set > things like the quality of the audio. It's not clear to me what the use > case is for that. Can you elaborate on that? Perhaps one example is the sort of thing described by MediaStreamTrackHints in the proposal. The Opus audio codec from the IETF standardization effort can switch between separate "voip" and "audio" coding modes. The script setting up the connection may have context information about which of these are more appropriate for its application. These are qualitative encoding choices, which significant overlap on any quality/bitrate scale. Information like that is always going to be codec-specific, so an concrete way of passing ad hoc parameters to the user-agent would find use there. -r
Received on Tuesday, 12 July 2011 10:00:16 UTC