[vorbis] Low bitrate encoding

[Olaf van der Spek]

----- Original Message -----
From: "jaromil" <jaromil at dyne.org>
To: <vorbis at xiph.org>
Sent: Tuesday, January 09, 2001 1:14 PM
Subject: Re: [vorbis] Low bitrate encoding

> On Tue, Jan 09, 2001 at 11:49:09AM +0100, Olaf van der Spek wrote:
>
> > > i am currently working on down/upsampling routines for the next major
> > > release of muse (2:1/1:2, is that meant here?), as far as i get
> something
> > > useful i'll notice - next 2 weeks. found really useful to peek into
> the
> > > sourcecode of SDL (www.libsdl.org) src/audio/SDL_audiocvt.
> >
> > Is downsampling (from 44100 hz to 22500 hz) something else then just
> taking
> > the average of two samples from the read buffer and putting that in the
> > write buffer?
> >
>
> that's not exact... more than averaging it's a stretching operation.
> assuming 16bit samples i would do it like that:

This means you just discard half of the samples.

>
> /* downsampling >>1 */
>
> for(i=src_length>>2;i;--i) {
> dst[0] = src[0];
> dst[1] = src[1];
> src += 4;
> dst += 2;
> }

What about this?
for(i=src_length>>1;i;--i) {
dst[0] = src[0] + src[1];
src += 2;
dst++;
}

>
> /* upsampling <<1 */
>
> for(i=src_lenght>>1;i;--i) {
> src -= 1;
> dst -= 2;
> dst[0] = src[0];
> dst[1] = src[0];
> }

Would it be better to interpolate the second value?

>
> so at the end is really simple
> it's not the first time i see this algorithm,
> i trust it more since i saw sam lantinga using it.
>
> i guess it can be done in asm with mmx lubrification...
>
> jrml
>
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