On Mo, 2007-01-08 at 17:47 +0200, Sergei Steshenko wrote:> On Mon, 08 Jan 2007 10:43:38 +0100> Sebastian Schäfer <schaefer@xxxxxxx> wrote: > > Adjusting the buffer size really did the trick!> > It's done by echoing something in /proc/asound/card0/pcm0p/sub0/prealloc> > The maximum value is given in prealloc_max.> > Now I just have to try and adjust the plugin for use with 48 kHz.> > > > (Wish for the next release: A config line where the sample rate to be> > used can be given and bands too close will automatically get thrown out> > or whatever. At least if that is possible, of course.)> > > Sebastian, where do I find the value of prealloc_max ?> Only in ALSA source ? Or in kernel source ?> > Thanks for the info, now I myself know how do it ! :-)> prealloc_max is also a file in /proc/asound/card0/pcm0p/sub0 > > Regarding> > "> A config line where the sample rate to be> used can be given and bands too close will automatically get thrown out> or whatever. At least if that is possible, of course.)> "> > - this is very difficult at best - the problem is, as I wrote, that> when the Perl code used to actually the generate the "C" code is> run, sample rate is not yet known.> > And in the definition of LADSPA each time sample rate changes the> plugin should be reinstantiated, that's because there can> be things (DSP coefficients, delay line lengths, etc.) which are> functions of sample rate.> > Again, joining frequencies at runtime doesn't seem to be an option,> because this means joining control ports, i.e changing their number,> which is not possible, or blocking them, which adds to code complexity.> > As a compromise I can add to the Perl code something like> expected_sample_rate entity, expected_fft_length, so the code will try to adjust bands/ports according to the two entities.> > However, still, if, say, the bands are as close as possible at> 48kHz, and suddenly the plugin is used at 96kHz, it will fail. That's exactly what I meant :-) BTW: As I tried to get it working with 48 kHz and commented out the bandbetween 20 and 40 Hz, I figured out that without that one, the equalizerworks even with 96 kHz (I tried different sample rates using mplayer-srate [Hz] file.avi), but unfortunately mplayer crashed when using22050 Hz. I did not further investigate this, as I normally do not playfiles with such a sample rate.> > Regards,> Sergei..> Regards,Sebastian -------------------------------------------------------------------------Take Surveys. Earn Cash. Influence the Future of ITJoin SourceForge.net's Techsay panel and you'll get the chance to share youropinions on IT & business topics through brief surveys - and earn cashhttp://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV_______________________________________________Alsa-user mailing listAlsa-user@xxxxxxxxxxxxxxxxxxxxxxxxxx://lists.sourceforge.net/lists/listinfo/alsa-user
