Re: [Alsa-user] MAudio 2496 left-channel crackling

2008-04-28 Thread Dominique Dumont
Matt Garman [EMAIL PROTECTED] writes:

 I did that too---took the card out and visually inspected it for
 anything obvious (leaking or mis-shapen capacitors, burnt-looking
 solder joints, etc).  I didn't see anything though.  But that just
 means nothing obvious is wrong!  ;)  Could still be a bunk component
 in there somewhere.

For the record, only the worst cases of dry solder joint are
visible. When suspecting bad joints, your best chance is to resolder
almost everything until you're bored or you card works
again. Whichever happens first.

When heating a joint, if you hear hissing or poping sounds, it means
that the joint was probably in bad shape.

I usually start by the biggest joints, work 1/2 hour, re-test,
resolder 1/2 hour, re-test... 

 No matter---I got the replacemnt request approved from the retailer,
 so hopefully I'll have a new one in a couple weeks or so.

Good news.

Cheers

-- 
Dominique Dumont 
Delivering successful solutions requires giving people what they
need, not what they want. Kurt Bittner

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[Alsa-user] Multiple things sending to Alsa

2008-04-28 Thread J M Needham
Hi,

Apologies for the apalling subject line. I have a Toshiba satellite pro
[never buy one] which runs Ubuntu Gutsy with an Intel HDA built in
soundcard. I have compilted alsa 1.16 with
the apropriate options (according to the hda-intel ubuntu howto:
https://help.ubuntu.com/community/HdaIntelSoundHowto ), and I get sound
out. What I don't have is that
on my PC, I can play sounds with clients connected to Jack to Alsa which
makes sound, at the same time as playing music through some player
connected directly to alsa (not using jack). Unable to do this on my
laptop.

The problem is that I have no idea where to start looking for the
problem. Other problems include not being able to kill processes using the
soundcard (revealed with lsof /dev/snd/*) that belong to programs that
have crashed, so the
soundcard remains blocked (kill -9 pid doesn't seem to work). If anyone
knows of an apropriate fm that I can
rt then that would be great or any options that I should have included at
compile time. Or even better if anyone can explain what's going on I'd
really appreciate the time.

Apologies if this is stupid; I have tried googling lots as I'm sick of
these two problems.

Jonty


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Re: [Alsa-user] Input feedback on snd-hda-intel with SigmaTel STAC9221 A1

2008-04-28 Thread steven shingler
Hi,
I also have this setup - on a Dell M2010 running Ubuntu 8.04
I'm unable to get any sound from the on-board speakers - only the
headphone jack works.
My alsa-info is at: http://pastebin.ca/1000180
There's also a codecgraph output svg at http://www.box.net/shared/5jpeiw800o
Any advice would be very gratefully received.
Many Thanks,
Steven
attachment: codec#0.svg-
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Re: [Alsa-user] Trying to get arecord/aplay to work with M-Audio

2008-04-28 Thread Takashi Iwai
At Sun, 27 Apr 2008 02:38:06 +0300,
Sergei Steshenko wrote:
 
 On Sat, 26 Apr 2008 10:10:46 +0200
 Takashi Iwai [EMAIL PROTECTED] wrote:
 
  At Fri, 25 Apr 2008 20:45:03 +0300,
  Sergei Steshenko wrote:
   
   On Fri, 25 Apr 2008 16:23:06 +0200
   Takashi Iwai [EMAIL PROTECTED] wrote:
   
At Fri, 25 Apr 2008 16:19:38 +0200,
I wrote:
 
 At Thu, 27 Mar 2008 01:58:50 +0100,
 Rene Herman wrote:
  
  On 26-03-08 23:57, Bill Unruh wrote:
  
   On Wed, 26 Mar 2008, Helge Fredriksen wrote:
   
   Ok, here's what I found out:
  
   Since these cards seem to lack the support for doing frequency 
   conversion in
   HW, you simply need to use the alsa plugin
   plughw instead of hw when referring to the card:
  
   arecord -Dplughw:0  test.wav
   aplay -Dplughw:0 test.wav
   
   AFAIK, alsa frequency conversion is pretty primative ( linear
   interpolation) which produces a huge amount of noise/distortion.
  
  The default is, but if you install the alsa-plugins package (and 
  libsamplerate) and add a line
  
  defaults.pcm.rate_converter samplerate
 
 Or speexrate.  This is usually much faster.

And forgot to write: you don't have to add anything to asoundrc for
speexrate plugin.  Install it, then alsa-lib will pick it up
automatically.


Takashi

   
   After some web search I see that there are several quality settings for
   speexrate.
   
   How does one choose/change the quality settings ?
  
  Use speexrate_best instead of speexrate, for example.
  See alsa-plugins/doc/speexrate.txt for details.
  
  
  Takashi
  
 
 Thanks.
 
 I've just read
 
 /usr/share/doc/packages/alsa-plugins/speexrate.txt
 
 file and I understand its contents, specifically, the
 
 
 Or, more easily, define a global variable defaults.pcm.rate_converter,
 which is used as the default converter type by plug and rate plugins:
 
 defaults.pcm.rate_converter speexrate
 
 Write the above in your ~/.asoundrc or /etc/asound.conf.
 
 
 part this way:
 
 I can edit /etc/asound.conf file and add, say,
 
 defaults.pcm.rate_converter speexrate_best
 
 line anywhere in the /etc/asound.conf file,
 
 Is this understanding correct ?

Yes.

 I read the file on my SUSE 10.3. laptop, hope it's good enough version.
 
 ...
 
 I've just discovered there is no /etc/asound.conf file on my laptop:
 
 
 [EMAIL PROTECTED]:~ ls /etc/asound*
 /etc/asound.state
 [EMAIL PROTECTED]:~
 .
 
 Should I create /etc/asound.conf ? Should it contain just one
 
 defaults.pcm.rate_converter speexrate_best
 
 line ?

Yes.  But I guess you barely notice the difference of output quality
from the default speexrate (although you'll notice the difference of
CPU usage :)


Takashi

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