Thanks for your answer!
Maybe I'm doing something wrong?
I run jack using :
jackd -d alsa -r 44100
then
pilo@pilo:~/schematic$ cat /proc/asound/card0/pcm0p/sub0/hw_params
access: MMAP_INTERLEAVED
format: S32_LE
subformat: STD
channels: 10
rate: 44100 (44100/1)
(...)
So when I run ardour, it uses 44100Hz for project sample rate. But, most of the time, alsa seems to assume that the output freq is 44.1Khz, but "requested" freq is 48Khz, and so resample the sound (which is then faster and higher in pitch as it would be).
Sometimes it works (for example when jack is the first one to use the sound card). The same trouble when I use a software which use oss emulation....
I really don't know what to do, I would prefer that alsa does no resampling at all!
Last time a friend send me drum tracks @44.1Khz, I play bass on those, and when I sent them back to him he told me "ouaw! you are tuned very low!", and I was not, that's how I discover there was a trouble with SR.
I try to play with .asounrc, by forcing Sample rate, but this doesn't work. And I wonder what would happen if I use the spdif in and out with this frequency trouble....
Anyway, thank you!
Pilo.
Hey,
I'm using that card (with jack and ardour even), but I'm not an
alsa-expert. Just guessing, but do you have jack set to run at 44100?
Jack seems to override whatever other ways you've set the sample rate.
Bigger problems with that card and alsa seem to be that you need to
set .asoundrc for it to work with certain apps, and even then it seems
to have problems if you're running pulse to alsa to it, or even
without it sometimes crashes. Not with any jack app, of course, just
other apps, I mean.
Walker
On Sun, 21 Feb 2010 12:02:31 +0100
Pierre-Laurent CHAMBERT <pierrelaurent.chambert@xxxxxxxxx> wrote:
> Hi everyone!
>
> I'm using an audiophile 2496 with alsa to do some pro audio working,
> mainly with ardour and jack.
> Unfortunately, I have a big problem with alsa : (automatic)
> resampling!
>
> Most of the time I can't work on 44100Hz project, even if I run jackd
> with a 44.1Khz freq, the card internal clock is at 44.1Khz (accord
> to /proc/asound info), but the sound is resampled just like if the
> card run @ 48Khz... and so the sound is faster and higher (impossible
> to work with). I don't understand how the resampling works in alsa,
> is there anyway to turn it off?
> When working at 48Khz it seems to always be fine...
> It is a real issue, and I can't really find useful info about it.
>
> By the way, in alsa 1.0.22 driver, ice1712 module (used for
> Audiophile 2496) is missing! (the source code is here, but no module
> is not compiled, whatever options is turned on during compiling).
>
> Thank you very much!
>
> Pilo
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