[Speex-dev] SPEEX on iPhone ?

[Ashhar Farhan]

you could use a simple filter like the one below. i just copied it
from my code. figure things out for yourself.
it down samples 16KHz to 8KHz.

- farhan


	p->coef[0] =	   2087;
	p->coef[1] =	   5979;
	p->coef[2] =     -7282;
	p->coef[3] =     -6747;
	p->coef[4] =	  28284;
	p->coef[5] =	  52763;
	p->coef[6] =	  28284;
	p->coef[7] =     -6747;
	p->coef[8] =     -7282;
	p->coef[9] =	   5979;
	p->coef[10] =	   2087;
	p->coef[11] =    -4816;



/* it block converts 320 samples at a time */
static void lowpass(struct Call *pc, short *pcm, int nsamples)
{
	int i, j;
	int	sum;
	short	out[6400];

	for (i = 0; i < nsamples; i++){

		pc->delay[0] = (int)pcm[i]; //update most recent sample
		sum = 0;

		for (j=0; j<11; j++){
			sum += (pc->coef[j]*-pc->delay[j]); //multiply sample by filter coef.s	
		}

		out[i]=(short)(sum/100000); //let sample at destination = filtered sample

		for (j=11; j>0; j--) //shift sample
				pc->delay[j] = pc->delay[j-1];
	}

	for (i = 0; i < nsamples; i++)
		pcm[i] = out[i];
}



On Fri, Nov 14, 2008 at 8:04 PM, Keith Kyzivat
<kkyzivat at tripleplayint.com> wrote:
>
>
> 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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