Being lazy here, so quick question:
I'm looking for a simple way to resample in ALSA from a capture/play interface that
supports only 48kHz, to 44.1kHz which is desired.
Both playback and capture resampling is needed, and it seems I neeed dsnoop and dmix.
However, I am bewildered by the somewhat byzantine configuration of .asoundrc.
There's one card on this machine, a "SB LIVE!" (which means "48kHz
only!"), running the emu10k driver.
There's no JACK up in this at all, it's just an old streaming PC running BUTT.
There is also no PulseAudio because I avoid PulseAudio.
If I were using ices2 I suppose I could use its built-in resampling, but that's not an
option in this case.
-ken