[FFmpeg-user] Audio sample format in alsa?

[Simon Roberts]

I'm trying to grab audio from a USB device that's managed by alsa. I find
that if I configure pulse to use this device, then I can grab successfully
with:

ffmpeg -f alsa -i pulse audiotest.mp3

But I need to refer to the hardware device directly, and I'm failing to do
that.

I have these cards:

$ cat /proc/asound/cards
 0 [HDMI           ]: HDA-Intel - HDA Intel HDMI
                      HDA Intel HDMI at 0xf7e34000 irq 44
 1 [UA3FX          ]: USB-Audio - UA-3FX
                      EDIROL UA-3FX at usb-0000:00:14.0-3, full speed
 2 [PCH            ]: HDA-Intel - HDA Intel PCH
                      HDA Intel PCH at 0xf7e30000 irq 43

and these devices:

$ cat /proc/asound/devices
  1:        : sequencer
  2: [ 1]   : control
  3: [ 1- 0]: digital audio playback
  4: [ 1- 0]: digital audio capture
  5: [ 1- 1]: digital audio playback
  6: [ 0]   : control
  7: [ 0- 3]: digital audio playback
  8: [ 0- 7]: digital audio playback
  9: [ 0- 8]: digital audio playback
 10: [ 0- 9]: digital audio playback
 11: [ 0- 0]: hardware dependent
 12: [ 2]   : control
 13: [ 2- 0]: digital audio playback
 14: [ 2- 0]: digital audio capture
 15: [ 2- 0]: hardware dependent
 33:        : timer

But when I try:

$ ffmpeg -f alsa -i hw:1,0 audiotest.mp3

I get:
[alsa @ 0x55ac7254fd80] cannot set sample format 0x10000 2 (Invalid
argument)
hw:1,0: Input/output error

But I'm not sure where to start with configuring this sample format. Can
someone point me in the right direction? I didn't find much in the alsa
section of the online manual.

Cheers,
Simon