Re: Some new music made (mostly) with Linux -- comments on mix?

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



On Mon, Apr 12, 2010 at 10:23:21PM -0400, Monty Montgomery wrote:
> > I found it helpful to listen to the track with JAMIN inline, because it brought out those details I needed to fix. Would be nice if postfish had that capability too. I dunno. Who else is using it?
> 
> A few old-timers on the Audacity lists still use it as I was
> developing it at a time I was also hacking on Audacity.  That was some
> time ago.  I have far fewer hobbies since the first kid arrived.
> 

Yeah, I know how that goes. I stopped playing music entirely from just before my daughter was born, up until she was about 6. If I'd had two or three kids, I'd still not be playing music now.

After poking around in postfish, it's MUCH more powerful than JAMIN, but the impediment to my using it, is that neither it, nor mplayer, nor even aplay, seem to like my FastTrack Pro:


Could not set DSP fragment size; continuing.
Unable to set playback for 16 bits, 2 channels, 44100Hz

I set postfish for 24 bits, and...

Could not set DSP fragment size; continuing.
Could not set DSP fragment size; continuing.
Unable to set playback for 24 bits, 2 channels, 44100Hz


Not postfish's fault; aplay is unhappy too:

aplay -D hw:0,0 foie-A.wav 
Playing WAVE 'foie-A.wav' : Signed 24 bit Little Endian in 3bytes, Rate 44100 Hz, Stereo
aplay: set_params:918: Sample format non available


And neither is mplayer:

Playing foie-A.wav.
Audio file file format detected.
==========================================================================
Opening audio decoder: [pcm] Uncompressed PCM audio decoder
AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400)
Selected audio codec: [pcm] afm: pcm (Uncompressed PCM)
==========================================================================
[AO_ALSA] alsa-lib: pcm_direct.c:905:(snd1_pcm_direct_initialize_slave) requested or auto-format is not available
[AO_ALSA] alsa-lib: pcm_dmix.c:1008:(snd_pcm_dmix_open) unable to initialize slave
[AO_ALSA] Playback open error: Invalid argument
Failed to initialize audio driver 'alsa'
Could not open/initialize audio device -> no sound.
Audio: no sound
Video: no video


If I convert the file to 24-bit using sndconvert, still no happiness:

aplay foie-24.wav 
ALSA lib pcm_direct.c:905:(snd1_pcm_direct_initialize_slave) requested or auto-format is not available
ALSA lib pcm_dmix.c:1008:(snd_pcm_dmix_open) unable to initialize slave
aplay: main:564: audio open error: Invalid argument

Playing foie-24.wav.
Audio file file format detected.
==========================================================================
Opening audio decoder: [pcm] Uncompressed PCM audio decoder
AUDIO: 44100 Hz, 2 ch, s24le, 2116.8 kbit/100.00% (ratio: 264600->264600)
Selected audio codec: [pcm] afm: pcm (Uncompressed PCM)
==========================================================================
[AO_ALSA] alsa-lib: pcm_direct.c:905:(snd1_pcm_direct_initialize_slave) requested or auto-format is not available
[AO_ALSA] alsa-lib: pcm_dmix.c:1008:(snd_pcm_dmix_open) unable to initialize slave
[AO_ALSA] Playback open error: Invalid argument
Failed to initialize audio driver 'alsa'
Could not open/initialize audio device -> no sound.
Audio: no sound
Video: no video

This is whatta I gots:

M-Audio FastTrack Pro at usb-0000:00:1d.0-2, full speed : USB Audio #1

Playback:
  Status: Stop
  Interface 3
    Altset 2
    Format: 0x21
    Channels: 2
    Endpoint: 4 OUT (ADAPTIVE)
    Rates: 44100, 48000
  Interface 3
    Altset 5
    Format: 0x21
    Channels: 2
    Endpoint: 4 OUT (ASYNC)
    Rates: 8000 - 48000 (continuous)


JACK, by the way, works perfectly though:

JACK compiled with System V SHM support.
loading driver ..
SSE2 detected
apparent rate = 44100
creating alsa driver ... hw:0,0|hw:0,1|128|3|44100|0|0|nomon|swmeter|-|32bit
control device hw:0
configuring for 44100Hz, period = 128 frames (2.9 ms), buffer = 3 periods
ALSA: final selected sample format for capture: 24bit big-endian
ALSA: use 3 periods for capture
ALSA: final selected sample format for playback: 24bit big-endian
ALSA: use 3 periods for playback


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

[Index of Archives]     [Linux Sound]     [ALSA Users]     [Pulse Audio]     [ALSA Devel]     [Sox Users]     [Linux Media]     [Kernel]     [Photo Sharing]     [Gimp]     [Yosemite News]     [Linux Media]

  Powered by Linux