[linux-audio-dev] alsa, oss , efficiency?

David dplist at free.fr
Wed Nov 1 12:10:50 UTC 2006

On Wed, 01 Nov 2006 08:30:17 +0100
lemmel <lem0mel at gmail.com> wrote:

>      Hi everyone.
> for a project, we need to be able to play sound (at first look wav
> file), and we made several tests ; with a created stereo sound, we
> try to use alsa but the results doesn't fullfill our needs :
> sound played at the time, T, we want, and finished at the date, D=T
> +sound duration.
> (thi is a software with strict time constraints)
> The sound was always troncated (even with finished software such as
> xmms, amarok), and even randomly truncated, (sound created with
> audacity, and exported as WAV 16/32 bit etc).
> When we use OSS, all seems to be perfect.
> But, it seems that OSS is nowadays "deprecated", and consequently we
> shouldn't use OSS. What we can do ? Are our alsa results due to
> misconfigurations  ?

I am not sure I understand your problems or your needs, but my advice is
to use jack. It provides you with a very simple API that gives you
sample accurate synchronicity and hides all the hard parts like
directly dealing with the soundcard driver.

If your client code is clean enough, then you can even evaluate with
precision the latency of the software set made of your client, the
jackd server and the soundcard driver backend. If you need to know the
latency of your hardware, I invite you to google for jdelay, an
measurement app. released by M. Fons Adriensen.

Go read http://www.jackaudio.org/, you'll be delighted.

BTW, thanks a lot to all the developpers of jack !


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