Re: [Alsa-user] Stuck with Terratec's PhonoPreAmp iVinyl

2008-02-06 Thread biblonchk



Problem solved by removing all 'pulseaudio' packages,
accordingly this tip on the unbuntuforum:  
http://ubuntuforums.org/showthread.php?t=601522




On Sun, 3 Feb 2008 11:44:08 +0100
B.I. Blonchk [EMAIL PROTECTED] wrote:

Dear all,

This is a newbe question. I've almost no knowledge in sound handling
and can't get further without some hints.

I've shortly acquired a Terratec iVinyl PhonoPreAmp. It's USB and
works  under os-x or window$ without requiring any particular device
driver for recording. An included proprietary software is required for
32 bits recording (which is not in my plans)

Ubuntu 7.10 'gusty gibbon'  hotplugs an usb1 audio device and set alsa
with snd-hwdep, snd-usb-audio and snd-usb-lib for hw:1,0 accordingly.

Alsamixer shows for the iVinyl device a simple 'mic' switch as a
unique controler (no volume or other tuning at all). This seams to be
also the case on the other plateforms.

Here the output of `cat /proc/asound/cards´


 0 [AudioPCI   ]: ENS1371 - Ensoniq AudioPCI
  Ensoniq AudioPCI ENS1371 at 0xdc00, irq 11
 1 [iVinyl ]: USB-Audio - PhonoPreAmp iVinyl
  Terratec PhonoPreAmp iVinyl at
usb-:00:11.0-1, full speed


Having found working parameters for iVinyl by running `arecord´ with all
available formats, i can now run the following command and get a vinyl 
recorded:

$ #
$ # Notice:
$ #   Format is mandatory. And only `S24_3LE´ seams to be working.
$ #   Channels is mandatory.
$ #   Rate default to 32000 if unset. 44100, 48000, 96000 may be set.
$ #
$ arecord -v -D iVinyl -f S24_3LE -r 44100 -c2 test.wav
Enregistrement en cours WAVE 'test.wav' : Signed 24 bit Little Endian
in 3bytes, Taux 44100 Hz, Stéréo
Hardware PCM card 1 'PhonoPreAmp iVinyl' device 0 subdevice 0
Its setup is:
  stream   : CAPTURE
  access   : RW_INTERLEAVED
  format   : S24_3LE
  subformat: STD
  channels : 2
  rate : 44100
  exact rate   : 44100 (44100/1)
  msbits   : 24
  buffer_size  : 22050
  period_size  : 5513
  period_time  : 125011
  tick_time: 1000
  tstamp_mode  : NONE
  period_step  : 1
  sleep_min: 0
  avail_min: 5513
  xfer_align   : 5513
  start_threshold  : 1
  stop_threshold   : 22050
  silence_threshold: 0
  silence_size : 0
  boundary : 1445068800
Terminé par le signal Interruption...
$

But Audacity crash miserabily as soon as AudioI/O preferences are
saved with recording device set to iVinyl (hw:1,0) and S24_3LE format
doesn't seams to be well accepted by jackd. I've played for hours
around .asoundrc trying to fix this issue.:(
Furthermore, this device doesn't exist by Alsa for the moment, isn't
said works with linux by it's manufacturer, nor did I found any
usable tips in linux forums or in google...
In other words i'm stuck.

Q1: How may I fix -f, -r and -c options of the `arecord´ command into
a named setup for capture device in the asoundrc file (for example, in
a way that may be directly used by calling 'arecord -D vinyl48
toto.wav' ) ?
Q2: What should be done in order to get Jackd and/or Pulseaudio
working with this device ?
Q3: Are there other alternatives ?

Many thanks in advance for your tips.
Kindest regards
blonchk

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[Alsa-user] multiple instances of snd-virmidi

2008-02-06 Thread Martin Kemp
Hi
Does anyone know how to set up multiple instances of snd-virmidi or to 
gain more than the standard 4 ports. I found a couple of things from 
Google, neither of which worked for me!
Thanks in advance for any help.
Martin

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[Alsa-user] alsa + festival + card 2

2008-02-06 Thread James Shatto
I can't seem to get festival to use any card other than the default card 0.

$ echo here kitty kitty kitty | aoss festival
Segmentation fault

Without aoss, it generates sounds to card 0.  Even with the following .asoundrc.

#--- START .asoundrc ---#
pcm.atiixp {
   type hw
   card 0
}
ctl.atiixp {
   type hw
   card 0
}

pcm.atiixp-modem {
   type hw
   card 1
}
ctl.atiixp-modem {
   type hw
   card 1
}

pcm.usb_audio {
   type hw
   card 2
}
ctl.usb_audio {
   type hw
   card 2
}

defaults.pcm.card 2

pcm.dsp0 {
type plug
#pcm.slave atiixp
pcm.slave usb_audio
}

ctl.dsp0 {
type plug
#pcm.slave atiixp
pcm.slave usb_audio
}
#--- END .asoundrc ---#

artsdsp -m festival also segfaults.

esddsp -m surprisingly doesn't segfault.  But the text gets all garbled like 
it's multi-threading several festivals at once.  It starts saying several words 
at the same time.

I don't normally run a sound daemon at all.  Except when recording audio, I'll 
manually start jackd.

Any tips to get festival to play nice with alsa?  I want the sound to go to my 
usb device which is piped to my stereo.  If only to annoy the cat.  I could 
reindex the drivers, but I'd rather not go that route.  Any other options?

/proc/asound/cards

 0 [IXP]: ATIIXP - ATI IXP
  ATI IXP rev 2 with unknown codec at 0xd0003400, irq 16
 1 [Modem  ]: ATIIXP-MODEM - ATI IXP Modem
  ATI IXP Modem rev 2 at 0xd0003800, irq 16
 2 [MobilePre  ]: USB-Audio - MobilePre
  M Audio MobilePre at usb-:00:13.0-1, full speed


Thanks,
James

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Re: [Alsa-user] multiple instances of snd-virmidi

2008-02-06 Thread Clemens Ladisch
Martin Kemp wrote:
 Does anyone know how to set up multiple instances of snd-virmidi or to 
 gain more than the standard 4 ports.

modprobe snd-virmidi enable=1,1,1,1

or put the options into /etc/modprobe.conf (or whatever module
configuration file your distribution uses):

options snd-virmidi enable=1,1,1,1

To force the virtual cards to a specific position, use multiple values
for the index option:

options snd-virmidi enable=1,1,1,1 index=4,5,6,7


HTH
Clemens

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