Thanks Atte. I am not getting much help in the ALSA forums... will continue
to mainly post here.
My best guess is that the problem still has to do with alsa trying to
initialize the sample format (bitrate) for the R16 for 32-bit integer as
opposed to 24-packed-in-32 or preferably straight 24-bit integer. Even
when I specifically set the sample rate in the quirk, it still goes to 32
bit.
With the R16, I get this in the JACK log:
ALSA: final selected sample format for playback: 32bit integer
little-endian
When I use other 24-bit only devices (like the Roland UA4FX I use for
playback), I get this:
ALSA: final selected sample format for playback: 24bit integer
little-endian
It does work for capture with 32 bit integer, though. Could that be
different since the device isn't being forced to process data at 32 bits,
rather the driver is receiving it in that format (i.e. 24 bits packed in
32)? With playback, the driver is explicitly telling the device to playback
a format it can't support. Just a theory.
So, I think the next step for testing is to find a different way to force
both capture and playback to 24 bits. This is how I was setting it in the
quirk before:
.formats = SNDRV_PCM_FMTBIT_S24_LE
Yet it still ends up as 32 bit. Any other formats to try or other methods
for setting the sample rate out there?
--
View this message in context:
http://linux-audio.4202.n7.nabble.com/re-Zoom-R16-tp87487p88128.html
Sent from the linux-audio-user mailing list archive at
Nabble.com.