Re: Quality resampling code for libasound

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At Wed, 21 Mar 2007 01:07:37 +1100,
Jean-Marc Valin wrote:
> 
> > Thanks.
> > It would be helpful if you provide a diff rather a complete tarball at
> > the next time.
> 
> OK, I'll do that next time.
> 
> >> diff -r 0bcf6f07c12c pph/rate_speexrate.c
> >> --- a/pph/rate_speexrate.c	Thu Mar 08 12:54:57 2007 +0100
> >> +++ b/pph/rate_speexrate.c	Tue Mar 20 14:08:30 2007 +0100
> >> @@ -29,6 +29,7 @@
> >>  */
> >>  
> >>  #include <stdio.h>
> >> +#include <samplerate.h>
> >>  #include <alsa/asoundlib.h>
> > 
> > This requires additional check in configure.
> > Why is this header needed now?
> 
> Oops, that's a mistake. There's no need for that include. Actually, it
> was also in the previous version I sent, so it's strange you're only
> noticing it now.

Heh, that's an advantage of reviewing a patch ;)

> >> @@ -42,7 +43,7 @@ struct rate_src {
> >>  
> >>  static snd_pcm_uframes_t input_frames(void *obj, snd_pcm_uframes_t frames)
> >>  {
> >> -   int num, den;
> >> +   spx_uint32_t num, den;
> > 
> > I don't see big merits to use its own types here...
> 
> Well, I use that type because that's what is used in the API. And the
> reason I use it in the API is because this code is meant to work even on
> platforms with 16-bit integers (e.g. TI C5X DSPs) and no C99 compiler.
> If that's causing problem, I guess you could still use unsigned int with
> no problem...

Don't worry, alsa-lib would be very unlikely ported to such an
architecture.


Takashi

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