uaudio0 -> audacity not recording

Hans Petter Selasky hps at selasky.org
Wed Apr 22 07:44:05 UTC 2015


On 04/21/15 23:08, Russell L. Carter wrote:
> Greetings,
> I am going to take a stab at this again before giving up.
> On 10/stable, I have a Behringer UCA202 that plays fine.
>
> Apr 21 12:54:06 feyerabend kernel: ugen1.9: <vendor 0x1a40> at usbus1
> Apr 21 12:54:06 feyerabend kernel: uhub13: <vendor 0x1a40 USB 2.0 Hub,
> class 9/0, rev 2.00/1.11, addr 9> on usbus1
> Apr 21 12:54:07 feyerabend kernel: uhub13: 4 ports with 4 removable,
> self powered
> Apr 21 12:54:07 feyerabend kernel: ugen1.10: <Burr-Brown from TI> at usbus1
> Apr 21 12:54:07 feyerabend kernel: uaudio0: <Burr-Brown from TI USB
> Audio CODEC, class 0/0, rev 1.10/1.00, addr 10> on usbus1
> Apr 21 12:54:07 feyerabend kernel: uaudio0: Play: 48000 Hz, 2 ch, 16-bit
> S-LE PCM format, 2x8ms buffer.
> Apr 21 12:54:07 feyerabend kernel: uaudio0: Play: 44100 Hz, 2 ch, 16-bit
> S-LE PCM format, 2x8ms buffer.
> Apr 21 12:54:07 feyerabend kernel: uaudio0: Play: 32000 Hz, 2 ch, 16-bit
> S-LE PCM format, 2x8ms buffer.
> Apr 21 12:54:07 feyerabend kernel: uaudio0: Record: 48000 Hz, 2 ch,
> 16-bit S-LE PCM format, 2x8ms buffer.
> Apr 21 12:54:07 feyerabend kernel: uaudio0: Record: 44100 Hz, 2 ch,
> 16-bit S-LE PCM format, 2x8ms buffer.
> Apr 21 12:54:07 feyerabend kernel: uaudio0: Record: 32000 Hz, 2 ch,
> 16-bit S-LE PCM format, 2x8ms buffer.
> Apr 21 12:54:07 feyerabend kernel: uaudio0: Record: 22050 Hz, 2 ch,
> 16-bit S-LE PCM format, 2x8ms buffer.
> Apr 21 12:54:07 feyerabend kernel: uaudio0: Record: 16000 Hz, 2 ch,
> 16-bit S-LE PCM format, 2x8ms buffer.
> Apr 21 12:54:07 feyerabend kernel: uaudio0: Record: 11025 Hz, 2 ch,
> 16-bit S-LE PCM format, 2x8ms buffer.
> Apr 21 12:54:07 feyerabend kernel: uaudio0: No MIDI sequencer.
> Apr 21 12:54:07 feyerabend kernel: pcm0: <USB audio> on uaudio0
> Apr 21 12:54:07 feyerabend kernel: uaudio0: HID volume keys found.
>
> root at feyerabend> mixer
> Mixer vol      is currently set to 100:100
> Mixer pcm      is currently set to  75:75
>
> When I play a vinyl record through the soundcard the output is
> fine. Every other source I've tried plays fine too.
>
> When I start up audacity, I see displayed:
>
> OSS, /dev/dsp, 2(Stereo) Recording Channels, /dev/dsp
>
> When I click on the monitor pane, I get level bars corresponding to
> the monitor levels I'm hearing.  I hit the record button and I seem to
> be seeing sensible record levels.  But when I hit the (re)play
> button, I just hear a bunch of garbled noise, not very loud.  A
> curiosity to me, but perhaps unimportant is that the audacity displays
> /dev/dsp, but only /dev/dsp0.0 exists.
>
> I build audacity using poudriere, and all options are set except
> for lame and debug.
>
> Could be a rookie mistake with audacity, or freebsd recording, or
> something else.  Any ideas?  This setup works fine on linux.
>

Hi,

Are you running a -stable or -current kernel?

Could you try to record using "sox", "play" and "rec" utilities instead?

I have one of these audio adapters myself, and the problem is likely in 
the audio backend which audacity is using. See this PR for example, 
where hardcoded Linux OSS values are used:

https://bugs.freebsd.org/bugzilla/show_bug.cgi?id=199558

If you try to configure jackd to use 24-bit samples under FreeBSD you'll 
simply get a bunch of noise, because the wrong sample FMT is selected.

--HPS



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