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. _______________________________________________ Linux-audio-user mailing list Linux-audio-user@xxxxxxxxxxxxxxxxxxxx http://lists.linuxaudio.org/listinfo/linux-audio-user
