[Speex-dev] SPEEX on iPhone ?

[Keith Kyzivat]

On Fri, Nov 14, 2008 at 3:57 AM, Vincent Burel
<vincent.burel at vb-audio.com>wrote:

> > Speech compression algorithms always are tunned to particular freq,
> > else they would take tons of time. That's because they use knowledge
> > that speech pitch (and other params) lies in well specified regions.
> > Thus if you feed algorithm with wrong freq, you'll fool it and it might
> > even not detect speech at all.
>
> Do you mean that speex is working at 44.1 Khz like at 32Khz without
> samplerate conversion ?
>
> It seems you suggest that working with 44.1 Khz signal instead of 32Khz
> (wich is the native samplingrate for the ultra -wide-band mode as far as i
> understand) ask for more CPU load. Since YOU have already worked with
> SPEEX,
> do you know how much CPU load it takes ? 10% more ? 50% more ?


Standard sample rate used with speex is 8kHz.
Speex wideband uses 16kHz
Speex ultra-wideband uses 32kHz

Many speech applications use use just the standard 8kHz speex mode.

What Alex is saying is, you'll (probably) want to record from sound card at
44.1kHz or 48kHz (depending on native rate of sound card-- which isn't
always easy to find out), then use a software resampler like speexdsp
provides (but any implementation will do) to resample the data down to your
target sample rate for speex (8kHz for standard speex, 16kHz for wb, 32kHz
for uwb).  Then push it through speex routines, send it where it needs to
be, decode with speex routines, then resample it back up to 44.1/48kHz, and
play it out at the destination end.

This of course, presumes that your audio source and sink are provided by the
myriad of sound cards that have been designed for Windows, and are supported
by other OSes.
Basically, sound cards these days are designed to work with only one or two
sample rates, and their drivers do the resampling to get it to
target/destination sample rate.  Quite a few years back (so some google
research suggests), Microsoft disseminated some faulty resampling reference
code that just got integrated with a lot of sound card drivers, and exists
to this day (so they say -- take this with a grain of salt though).  One
thing I can be sure of (I have seen), is that if you resample the data to
the sound card's native sample rate (which, I think it can be safe that most
all sound cards use 48kHz or 44.1kHz), latency (definitely) and
jitter(maybe?) goes down pretty significantly.

-- 
Keith Kyzivat

SIPez LLC.
SIP VoIP, IM and Presence Consulting
http://www.SIPez.com
tel: +1 (617) 273-4000
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