Re: [codec] WG Review: Internet Wideband Audio Codec (codec)

"Roni Even" <ron.even.tlv@gmail.com> Wed, 13 January 2010 20:54 UTC

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From: Roni Even <ron.even.tlv@gmail.com>
To: dpetrie@sipez.com, 'Alexander Chemeris' <Alexander.Chemeris@sipez.com>, 'Christian Hoene' <hoene@uni-tuebingen.de>
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Date: Wed, 13 Jan 2010 22:53:05 +0200
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Cc: codec@ietf.org
Subject: Re: [codec] WG Review: Internet Wideband Audio Codec (codec)
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Daniel,
These are not a codec specific information and there is work on it in AVT.
See
http://tools.ietf.org/id/draft-lennox-avt-rtp-audio-level-exthdr-01.txt and
there is also
http://tools.ietf.org/id/draft-ivov-avt-slic-02.txt which provides
information to the receiver.

Roni Even

> -----Original Message-----
> From: codec-bounces@ietf.org [mailto:codec-bounces@ietf.org] On Behalf
> Of Daniel Petrie
> Sent: Wednesday, January 13, 2010 7:54 PM
> To: 'Alexander Chemeris'; Christian Hoene
> Cc: codec@ietf.org
> Subject: Re: [codec] WG Review: Internet Wideband Audio Codec (codec)
> 
> While we are discussing conferencing, I would like to add my wish list
> of requirements:
> 
> I would want the following information included in the RTP packet so
> that I can do speaker selection on the streams without having to have
> the expense of decoding them and run the algorithms to calculate these
> values.  This is useful in all 3 of Christian's conferencing scenarios.
> 
> 1) Voice activity indication (minimally binary, ideally magnitude)
> 2) gain control either already applied or a multiplier to apply to
> achieve a consistent gain.
> 3) Energy level
> 
> --- On Wed, 1/13/10, Christian Hoene <hoene@uni-tuebingen.de> wrote:
> 
> > From: Christian Hoene <hoene@uni-tuebingen.de>
> > Subject: Re: [codec] WG Review: Internet Wideband Audio Codec (codec)
> > To: "'Alexander Chemeris'" <Alexander.Chemeris@sipez.com>
> > Cc: codec@ietf.org
> > Date: Wednesday, January 13, 2010, 5:55 AM
> > > This codec should be also useful
> > in
> > > conference
> > > scenarios, where you have to decode it at conference
> > server and encode
> > > mixed signal again. And if I haven't missed something,
> > this use-case
> > > was
> > > explicitly listed in codec requirements.
> >
> > The conferencing scenario is not yet well described in the
> > current codec requirements document. Different kinds of
> > systems are possible:
> >
> > 1) A central conferencing server that serves all clients.
> > - a) which does a efficient single changing mixing - still
> > a challenge if you have multiple operational modes in a
> > codec.
> > - b) which does a more intelligent mixing using (at least)
> > two channels to support spatial hearing - this increases the
> > quality of conference call significant. Thus, stereo modes
> > must be supported if the codec shall support state of the
> > art conferencing technologies.
> > 2) A mesh-like N-to-N transmission conference call. Then
> > you might benefit from layered coding because you have at a
> > client you have to encode only once. But then again, how to
> > support multiple coding modes?
> > 3) One client acts as conference server (similar to 1)
> >
> > In total, the support of conferencing calls will require:
> > - Layered coding
> > - Efficient mixing of streams
> > - Support of stereo
> > - Good (self-)transcoding
> > Is this all achievable?
> >
> > Christian
> >
> >
> > _______________________________________________
> > codec mailing list
> > codec@ietf.org
> > https://www.ietf.org/mailman/listinfo/codec
> >
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