Fernando explained what is going on, I think (at least his bugfix works
on my machine). aplay uses the device "plughw:0", whereas snd/sndplay
default to "hw:0". plughw then provides low-fi sampling rate interpolation
if necessary, whereas hw just throws up its hands. To see aplay in the
same mode
aplay --device=hw:0 oboe.snd
and to get snd to use alsa's interpolator:
SNDLIB_ALSA_DEVICE=plughw:0 snd oboe.snd
(I think -- this looks like bash, and won't work in tcsh -- I did this:)
setenv SNDLIB_ALSA_DEVICE "plughw:0"
snd oboe.snd
I think I'll add scheme/ruby variables for alsa-device, alsa-playback-device,
and
alsa-capture-device so that this can be set in ~/.snd or wherever, and add
Fernando's two notes to grfsnd.html in a new section ("Snd and Alsa"), and
it seems to me that Snd should provide hi-fi interpolation in the fallback
case -- I didn't realize this was such a complex business. Any other
suggestions
are always welcome.
Another minor annoyance -- with plughw:0, snd now complains:
":;alsa_audio_open: plughw:0: could not set rate to exactly 22050, set to 22100
instead"
which I'll either replace with my own src, or add a way to turn the warning off.
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