webrtc reqs related to the Audio WG

Dear Audio WG (cc webrtc),

in the latest version of the use-cases - and - requirements document 
(<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/?include_text=1>) 
for webrtc the requirements on audio processing have been changed:

    ----------------------------------------------------------------

    A13             The Web API MUST provide means for the web
                    application to apply spatialization effects to
                    audio streams.
    ----------------------------------------------------------------
    A14             The Web API MUST provide means for the web
                    application to detect the level in audio
                    streams.
    ----------------------------------------------------------------
    A15             The Web API MUST provide means for the web
                    application to adjust the level in audio
                    streams.
    ----------------------------------------------------------------

    A16             The Web API MUST provide means for the web
                    application to mix audio streams.
    ----------------------------------------------------------------

The term "audio stream" was selected at an early stage; I would say it 
corresponds a "Track" in the MediaStream object that is currently in the 
API draft (<http://dev.w3.org/2011/webrtc/editor/webrtc-20111004.html>).

Anyway, feedback on these requirements is welcome (I'm not sure I'm 
using good wording).

A14 and A15 are in the use-cases motivated by the need to equalize 
levels between audio streams (Tracks) coming from different participants 
in a multiparty session.
But I can see other uses of A14: display the level in a meter locally to 
calibrate mic settings before a session, detect silence, detect noise 
generating party in a multiparty session etc.

As said,
feedback would be appreciated.

Stefan

Received on Thursday, 6 October 2011 07:26:46 UTC