[vorbis-dev] Re: Speex: Open-source, patent-free speech coding

[Jean-Marc Valin]

> Speex shouldn't bother dealing with non-standard speech sampling-rates.
> Encoding tools like mine would downsample the signal before delivering
> it to Speex, and decoding tools like Tobias' DirectShowFilter should
> take care of the upsampling and summing the different tracks (common + speech).
> There's a brilliant, open-source, HQ sample-rate convertor, called 
> SSRC. it's under LPGL, and i even made a dll release of this fine tool.

Well, it all depends on whether 8 kHz bandwidth (16 kHz sampling) is OK
for you...

> sure thing! the speech track suppose to have lots of silent moments, so 
> DTX (AD/CFI) would help to drop the bitrate.
> problem is - can CELP handle multiple spokesmen (ie, when two ppl are
> talking at the same time), and will sound quality differ when compressing English track compared to encoding Russian track ?

The only potential problem is the one where two people are talking at
the same time. In this case, the solution could be to just boost the
bit-rate for a couple frames.

> the first step is :
> - decide how we extract the 'common' track

I'll leave that one to you. I have no idea about the properties of the
different tracks.

> - define Speex integration in ogg

Could anyone tell me how to do that?

> Jean-Marc, 
> you have a lot of knowledge regarding speech models. can you point out
> some useful sites/tools which i should check in order to implement the
> first stage of 'extracting the common track' ?

Sorry, I have no idea. I guess it first depends on whether that "common"
part is *exactly* the same in all tracks (ie english track - common
track = only english and nothing else).

        Jean-Marc


-- 
Jean-Marc Valin, M.Sc.A.
LABORIUS (http://www.gel.usherb.ca/laborius)
Université de Sherbrooke, Québec, Canada

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