We started out with a simple comparing of samplerate and samplerate_order. Out of the 5 he sent me, I got 5 correct. Was pretty simple considering that samplerate_order is pretty bad. We just did this one to make sure my ears were working.
Comparing samplerate and samplerate_best managed to embarrass me though. Out of 10, I got 1 right, 2 wrong, and 7 unsure. And honestly, I think the one I got right was a guess.
Can anyone give link to good test sample for checking resampling quality? I know about dial.wav but maybe better test sample exists?
And, I can tell right away that samplerate sounds better on udial than samplerate_best. samplerate_best has some added freqs on the last four tones. Same with samplerate_medium. What I find odd is that the default device is supposed to use dmix, or so that's what the ALSA docs have been saying. But clearly by the sounds of it, it's just be an alias to hw. Maybe the ALSA setup under Gentoo is different, not sure...
default definately goes through dmix on my machine (Ubuntu 6.06), but it is possible that Gentoo is different. To me, udial sounds best on samplerate_medium - but it is a very artificial sample so i'm not too worried about optimising that one.
Well, looking at the confs in /usr/share/alsa/cards, EMU10K1.conf does not use dmix for default. Almost all the other card's conf uses dmix, but the emu10k1 doesn't. I even checked the ALSA CVS, no dmix for the emu10k1. So, because of this, ALSA just uses plughw instead for default. So I guess only certain cards have been updated to use dmix. Even the conf for my on-board sound, VIA8237, doesn't use dmix as default.
Sorry guys for this Off-topic, but does have anyone an idea (or has tested) the same on windows? Does it have the same low-quality mixer or converter? If yes, what are the possibilities to make it better?If this is dependent on the hardware you use, I am mostly interested in the HDA built-in audio chips in notebooks (for example SoundMAX AD1986A)
As conclusion: ALSA hardware driver (from kernel) doesn't resample anything, it is work for ALSA library. And resampling works only with correct config for given soundcard.
The resampling problem is very usual with low cost soundcards also in Windows. My solution in Windows has been foobar2000 with resampler. There are few quite good resamplers for foobar2000 and the default PPHS resampler is quite good. You can test your audio hardware resampling with it. Use the problem sample mentioned earlier in this thread and test the resampler 48 kHz on and off. Check which setting sounds best with your hardware.
I am also posting a bug report against the equivalent Debian package.
See kmixer resampling measurements at http://www.kikeg.arrakis.es/measurements/
Edit: as to the need of outboard DACs for noise free audio, see the analog measurements. They are performed with and unexpensive, non-balanced internal card. Balanced internal soundcards usually offer even better noise performance. One of the best soundcards I know of (if not the best), the Lynx Two, is internal.