Audiophile 2496: resampling problem with ALSA [solved]

cyberpatrol · Last edited by cyberpatrol on Sat Jul 07, 2007 7:00 am; edited 1 time in total

Hi,

I've got an M-Audio Audiophile 2496 (ice1712) since a few weeks. When I installed this card everything was ok. I played a bit with envy24control, alsamixer, /etc/asound.conf, set the volumes, didn't lock the sample rate etc.

The first time every sound file was played with its own sample rate without resampling. If I played a file which was recorded at 44.1 kHz with e.g. Audacious or MOC the Actual Rate in envy24control showed 44100 and if I played a file which was recorded at 48 kHz the Actual Rate in envy24control showed 48000.

Everything incl. dmix worked perfectly without an ~/.asoundrc or /etc/asound.conf which doesn't seem to be necessary anymore as the ALSA documentation or the ebuild - I can't remember which one - tells.

Since a while, after playing a bit more with the envy24control, alsamixer and the file /etc/asound.conf where I always reset the settings to the working ones and after one or two `emerge -uDN world`, I'm having a big problem.

Now everything is played and recorded at 48 kHz, at least the Actual Rate in envy24control always switches to 48000, doesn't matter with which sample rate the file is recorded and to which rate I set the Master Clock. If I lock the rate to the file's sample rate the sound is played slower or faster and pitched depending on the file's and the locked sample rate. Btw., locking the sample rate wasn't an option anyway because I want a file to be played automatically in it's original sample rate without changing the sample rate for every file manually.

The Rate State option Reset in envy24control doesn't help either. Neither did help uninstalling every ALSA package, deleting the ALSA configfiles and remerging ALSA.

If I choose OSS as the output plugin in e.g. Audacious and MOC and therefore use the ALSA OSS plugin the files are played again in their original sample rate and the Actual Rate in envy24control changes to the particular sample rate as it is expected. But I have the impression that the sound quality with ALSA OSS plugin is worse than with the native ALSA.

And Audacious and MOC always show the original sample rates of the files. So I think that this is not a problem with the audio players but with ALSA.

I hope someone can help me and tell me how to fix this problem. Or should I file a bug report about this?

I'm actually using this software:

sys-kernel/gentoo-sources-2.6.21-r3
media-sound/alsa-headers-1.0.14
media-libs/alsa-lib-1.0.14a-r1
media-sound/alsa-utils-1.0.14
media-sound/alsa-tools-1.0.14
media-sound/audacious-1.3.2
media-sound/audacious-plugins-1.3.5
media-sound/moc-2.4.2

I'm of course using the in-kernel driver ice1712.

gimpel

cyberpatrol

gimpel

cyberpatrol

Akkara · Posted: Sat Jul 07, 2007 8:18 am Post subject:

Thanks for asking this question!

I have the same audio card and I had noticed a change a while back as well but wasn't sure how to describe it.

One day when I was testing with tones around 15KHz or so, and they had all sorts of weird artifacts where I didn't recall having heard them before. The artifacts weren't there when I tried burning the tones to a CD and playing that on a regular CD player.

I suspected that it might have been related to sampling rate when I noticed that envy24mixer was showing 48KHz when it used to always show whatever sampling rate the song was recorded with. Even my 96KHz stuff was being played at 48K. Perhaps the artifacts I was hearing was related to the resampling. They are especially prominent at 19KHz.

(Aside: For something like dmix, it'd be good to have an option (maybe even the default option) that sets the sampling rate to whatever sound is about to be played. If a second sound then comes up needing mixing, that second sound is resampled, if needed, to the already playing sound's sampling rate. Since most of the time only one sound at a time is playing, this means that most of the time no resampling will take place.)