On Mo, 2007-01-08 at 17:47 +0200, Sergei Steshenko wrote: > On Mon, 08 Jan 2007 10:43:38 +0100 > Sebastian Schäfer <[EMAIL PROTECTED]> 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 band between 20 and 40 Hz, I figured out that without that one, the equalizer works even with 96 kHz (I tried different sample rates using mplayer -srate [Hz] file.avi), but unfortunately mplayer crashed when using 22050 Hz. I did not further investigate this, as I normally do not play files with such a sample rate. > > Regards, > Sergei.. > Regards, Sebastian ------------------------------------------------------------------------- Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV _______________________________________________ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
