Hello all,

Since I've been named I guess a quick comment might be in order. Of course, 
Dave is correct about the potential for artifacts of bss algorithms and that 
means that a single first-order b-format stream is sufficient for all uses, 
even with upsampling and whatnot. More channels are required to fully harness 
large loudspeaker arrays, and the main options on the table are then to add 
direct loudspeaker channels, vbap-panned audio objects or higher order 
ambisonics. Another option exists, using the mentioned bss-related methods that 
I have some interest in, so I'll throw it out there:

If the audio material is distributed as several 1st order b-format streams, 
each having only a small number of sound sources, then any bss-related 
artifacts can be limited below any chosen level, limited only by the number of 
available channels. This assumes that each stream is decoded separately with a 
parametric decoder. You would for example put dialogue on a separate b-format 
stream. Center-positioned speech would then decode to the center speaker only, 
which is a basic requirement. Since audio tracks are typically downmixed from a 
multitude of tracks anyway, the need of downmixing to a handful of b-format 
streams shouldn't affect the workflow of sound designers much. Once suitable 
streams for the highest-quality targeted systems are created, it would be 
trivially simple to downmix them to a suitable number of b-format streams for 
lower-quality media intended for smaller loudspeaker arrays.

Cheers,

Svein

> From: Dave Malham <dave.mal...@york.ac.uk>
> Subject: Re: [Sursound] A proposal for an Ambisonics based 3D audio codec, 
> MPEG/ITU style...
> Date: 20. januar 2013 17:53:31 GMT+01:00
> To: Surround Sound discussion group <sursound@music.vt.edu>
> Reply-To: Surround Sound discussion group <sursound@music.vt.edu>
> 
> 
> Hi Jo,
>     To some extent, yes, but only if you are working with material
> which can tolerate the artefacts that blind source separation
> algorithms will, sooner or later - and usually sooner - generate. Such
> algorithms, including those in Harpex, are getting much better (I can
> remember hearing some really terrible earlier attempts) but, so far
> anyway, it is always possible to find some material that will screw
> them up.  There are two factors which will reduce this as time goes on
> - the continuing improvements in the technology and the fact that much
> of the audience these days have perceptions that are so screwed up by
> listening to compressed audio that maybe the artefacts maybe won't be
> heard by them anyway :-)
> 
>  All the best
>   Dave
> 
> 
> On 20 January 2013 11:58, Joseph Anderson <j.l.ander...@phonecoop.coop> wrote:
>> Hi Dave,
>> 
>> I'll throw in...
>> 
>> 
>> On 20 Jan 2013, at 9:02 am, Dave Malham <dave.mal...@york.ac.uk> wrote:
>> 
>>> The biggest limitation to
>>> the whole thing is the availability of higher order microphones,
>> 
>> Actually, as we've heard, upsampling from a 1st order mic can give very good 
>> results. Svein Berge's Harpex (http://www.harpex.net) is available as a 
>> plugin... which means the production path open.
>> 
>> 
>>> though Eigenmikes are becoming more widely available and one can hope
>>> that Peter Craven's new ideas
>>> (http://www.findaphd.com/search/ProjectDetails.aspx?PJID=40193) pan
>>> out and give us at least a good second order microphone.
>> 
>> 
>> 
>> It'll be great to see 'native' HOA mics on the market....
>> 
>> 
>> 
>> My best,
>> Jo
>> 
>> ~~
>> Joseph Anderson
>> 
>> Artist:                                 http://joseph-anderson.org
>> Ambisonic Toolkit:      http://ambisonictoolkit.net
>> 
>> 
>> 
>> 
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> 
> 
> 
> -- 
> As of 1st October 2012, I have retired from the University, so this
> disclaimer is redundant....
> 
> 
> These are my own views and may or may not be shared by my employer
> 
> Dave Malham
> Ex-Music Research Centre
> Department of Music
> The University of York
> Heslington
> York YO10 5DD
> UK
> 
> 'Ambisonics - Component Imaging for Audio'
> 
> 
> 
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