[Alsa-user] Metallic recording in cs46xx on Thinkpad T22

2008-03-11 Thread Gadi Oron
Hi everyone,

I ran over a lot of Internet search and could not find any clue to this
issue.

I am trying to record sound on an old Thinkpad T22 that uses the cs46xx
sound driver.

Each time you start to record you have a 10% chance of having the recording
completely distorted and having a metallic sound. When you look at the
waveform it looks as though there are small segments with sharp transitions
between them, a little like if these segments were moved a little from their
correct place.

I've tried many many combination and none gives a solution, including

   - Native OSS, 3.92, 4.0
   - kernel 2.4, 2.6.11, 2.6.26 (rh7.3, fedora 1,2,7,8)
   - Knoppix live CD
   - Activate thinkpad=1 for the module
   - Using native ALSA or OSS emulation.
   - Changing fragment size
   - Resetting mixer setting before recording
   - Changing recording format LE,BE, 8bit, 16bit
   - Changing recording frequency
   - Disable ACPI
   - Change IRQ of the sound-card
   - Using a low-latency kernel and giving the recording process
   real-time priority (FIFO_SCHED)

None gave an improvement. It works fine on the Win2k system that is
installed on the same computer.

Does anyone have any clue to what might be going on?

Is there any way to do a hard reset to the sound card (without
unloading-reloading the modules)?

Thank you for the help.
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Re: [Alsa-user] looking for some advise

2008-03-11 Thread Julien Claassen
Hi Romeo!
   I don't have neither. But I have an M-Audio Delta 1010LT, which I suppose is 
RELATIVELY close to the revolution 5.1. If they also use the ICE-chip, then 
they are nice. I also remember, that we had a lot of traffic about the 
revolution cards here. Since I didn't notice those mails for a while, I think 
it's a good one.
   That all sounds very vague, but still... Did you take a look at the alsa 
supported cards? If so and if the Revolution pops up there, then I'd say: Have 
a go at it!
   My Delta has very nice quality, very reasonable lowlatency and the 
BIT-depth/samplingrate should meet your requirements perfectly. Btw.: The 
reasonable lowlatency is mostly due to my system. A better PC than mine 
(1.8GHz CPU, average Harddisk, DDR-RAM) should really get you 64-128 periods 
latency with JACK.
   HTH.
   Kindest regards
   Julien


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Re: [Alsa-user] Missing folder: /proc/asound/card0/pcm0p

2008-03-11 Thread John Sigler
hascii wrote:

 the folder /proc/asound/card0/pcm0p is not there any more.

What is the output of the following command?

gunzip -  /proc/config.gz | grep CONFIG_SND_VERBOSE

You might need to enable CONFIG_SND_VERBOSE_PROCFS in your
kernel configuration.

cf. http://alsa.opensrc.org/Proc_asound_documentation

Regards.

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[Alsa-user] S32_LE format

2008-03-11 Thread John Sigler
Hello,

I experimented with the following command:

$ arecord -vv -M -r 48000 -f S32_LE -t raw -d 60 -D hw /tmp/out

which gave the following output:

Recording raw data '/tmp/out' : Signed 32 bit Little Endian, Rate 
48000 Hz, Mono
Hardware PCM card 0 'HDSPM MADI' device 0 subdevice 0
Its setup is:
   stream   : CAPTURE
   access   : MMAP_NONINTERLEAVED
   format   : S32_LE
   subformat: STD
   channels : 1
   rate : 48000
   exact rate   : 48000 (48000/1)
   msbits   : 24
   buffer_size  : 8192
   period_size  : 4096
   period_time  : 85333
   tstamp_mode  : NONE
   period_step  : 1
   avail_min: 4096
   start_threshold  : 1
   stop_threshold   : 8192
   silence_threshold: 0
   silence_size : 0
   boundary : 1073741824

AFAIU, S32_LE means that each sample is a 32-bit signed quantity 
stored least-significant byte first, right?

e.g. -2 (0xfffe) is stored as fe ff ff ff and 1 is stored as
01 00 00 00 right?

The dump of the beginning /tmp/out follows:

  00 00 a5 f5 08 00 43 f6  08 00 bc f6 00 00 20 f7
0010  08 00 9a f7 00 00 3e f8  00 00 11 f9 08 00 f2 f9
0020  00 00 a9 fa 08 00 42 fb  08 00 e7 fb 08 00 6b fc
0030  08 00 ae fc 00 00 f6 fc  00 00 75 fd 08 00 f3 fd
0040  00 00 31 fe 08 00 41 fe  00 00 62 fe 00 00 b5 fe
0050  00 00 30 ff 08 00 c2 ff  00 00 47 00 08 00 80 00
0060  00 00 65 00 00 00 47 00  00 00 63 00 08 00 ab 00
0070  00 00 f9 00 08 00 0c 01  08 00 c4 00 00 00 60 00
0080  08 00 1a 00 00 00 c3 ff  00 00 2d ff 08 00 7e fe
0090  00 00 d5 fd 00 00 2f fd  00 00 95 fc 00 00 03 fc
00a0  08 00 66 fb 00 00 d7 fa  08 00 6b fa 08 00 01 fa
00b0  08 00 91 f9 00 00 41 f9  00 00 e9 f8 08 00 4e f8
00c0  00 00 c4 f7 00 00 89 f7  00 00 2a f7 08 00 43 f6
00d0  08 00 f9 f4 00 00 81 f3  08 00 0f f2 08 00 ce f0
00e0  08 00 81 ef 08 00 08 ee  08 00 b2 ec 00 00 7e eb
00f0  08 00 27 ea 00 00 d4 e8  00 00 bd e7 00 00 b6 e6

The first sample is stored as 00 00 a5 f5.
Does this represent the value 0xf5a5?

For the majority of samples, the first two bytes are 00 00.
Is this because the input stream was, in fact, a 16-bit stream?

For the rest of the samples, the first two byte are 08 00.
Why don't all the samples start with 00 00?
Does this sequence 08 00 have a special meaning?

-- 
Regards.

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Re: [Alsa-user] Metallic recording in cs46xx on Thinkpad T22

2008-03-11 Thread James Shatto
 Each time you start to record you have a
 10% chance of having the recording completely
 distorted and having a metallic sound.

I know that sound.  And it is quite ugly.  On my snd-hda-intel board(nVidia 
MCP61), I have to increase the number of periods to overcome this sound.  
default of 2 increased to 3 and all was fine, in jackd -n 3.  For arecord you 
might look at different than default buffer/period sizes.  It's basically a 
latency issue.

Other considerations are to give the audio group realtime permissions so things 
like ethernet traffic doesn't cause clicks and other distortions in the sound.  
/etc/security/limits.conf

HTH,
James

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Re: [Alsa-user] Preferred way to disable dmix

2008-03-11 Thread Michael Kiermaier
  To use your second card, you have to specifify the device somewhere. If
  you want just the hardware abilities, just use the hw:1 device instead
  of default:1 or whatever you are using now.
 
  Using hw:1:0, I get the error message sample format non available most
  of the time, I guess the reason is that it is a 24bit sound card and my
  files are 16bit.
 
  So what I want is:
  No dmix, but automatic conversion to 24 bit.
 
  Is there a simple way to do that?

 plughw:1 should do exactly what you want from it, it seems.

Yes, I was not aware of plughw.
Thanks for the hint.

  (your default PCM is defined in $PREFIX/share/alsa/cards/ICE1712.conf)
 
  On my system it is /usr/share/alsa/cards/ICE1712.conf.
  Should I really edit that file? I have some hesitations to touch files in
  the /usr/-directory

 No, that was just showing you were things are. If you'd really want to
 change things you'd redefine the default device in /etc/asound.conf or
 ~/.asoundrc

ok, thank you.

~Michael

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Re: [Alsa-user] One process per audio channel

2008-03-11 Thread John Sigler
Hello Jaroslav,

Jaroslav Kysela wrote:

 John Sigler wrote:
 
 I have an RME AES-32 PCI board which provides 4 stereo input channels 
 and 4 stereo output channels.
 
 (I'm using the hsdpm driver at this time.)
 
 I want to use one process per channel, i.e. process A handles stereo 
 input #1 (on the XLR connector #1), process B handles stereo input #2, 
 etc.
 
 The processes are independent, in that process A might be started, and 
 only several hours later, process B is started, then a few hours later 
 process A is killed and restarted.
 
 Is it possible to do that with the ALSA library?
 
 Yes, look for the dsnoop plugin configuration in alsa-lib.

Thanks for the tip. I will investigate. I must say that the ALSA 
configuration files look like pure voodoo magic to me.

I found the following documentation:

doc/asoundrc.txt
http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
http://alsa.opensrc.org/.asoundrc
http://alsa.opensrc.org/Dsnoop
http://alsa.opensrc.org/AlsaTips

Is there any other good documentation?

How can alsa-lib open a device node several times when the driver
only allows one process to open it?

I see references to ipc in alsa.conf.

My embedded kernel is compiled with
# CONFIG_SYSVIPC is not set

Does the dsnoop plugin require kernel support?

-- 
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[Alsa-user] 64 bit Compilation/Installation problem

2008-03-11 Thread Roger Pryor
Hi:

I seem to have a problem with Alsa 1.0.16 when compiling.

My system is: an Open SUSE 10.2 running on a Intel DP965LT mobo, with an
Intel Core2 Duo E6420 processor.  2 G Ram.  Intel HDA sound card, which
has given me lot of problems.

OpenSUSE 10.2 comes with Alsa 10.0.14a, which does not seem to properly
support the Intel HDA chipset.  So, I wanted to upgrade to Alsa 1.0.16.  I
downloaded all available packages from the Alsa site, built and installed
the driver (using --with-suse=yes), ran alsaconf and rebooted.  So far so
good. I built the library, and installed that.  I try to build the utils,
but it fails with the error no TLV support in alsa-lib.  I check the date
on /usr/lib64/libasound.so.2.0.0, only to find it was NOT updated with the
libray installation, yet the version in /usr/lib was updated.  Temporarily
symlinking libasound.2.0.0 in /usr/lib64 to /usr/lib allows the
compilation of alsa-utils to proceed and complete without error.  BUT, all
other applications that use /usr/lib64/libasound now complain about wrong
ELF type, ELFLIB64,  HuH???  

Not being a real programmer (Hardware engineer, retired), this says to me
that:  

a)  Perhaps the installation of alsa-lib is placing the library in
the wrong directory on 64 bit systems.

b)  If the ELF type in the /usr/lib64 directory IS 64 bit, why is
that causing a complaint?

At that point, my head aches and I need some help and guidance, please.

--
Roger PryorEmail: [EMAIL PROTECTED]
Vancouver, Canada

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Re: [Alsa-user] 64 bit Compilation/Installation problem

2008-03-11 Thread Sergei Steshenko
On Tue, 11 Mar 2008 07:29:11 -0700
Roger Pryor [EMAIL PROTECTED] wrote:

 Hi:
 
 I seem to have a problem with Alsa 1.0.16 when compiling.
 
 My system is: an Open SUSE 10.2 running on a Intel DP965LT mobo, with an
 Intel Core2 Duo E6420 processor.  2 G Ram.  Intel HDA sound card, which
 has given me lot of problems.
 
 OpenSUSE 10.2 comes with Alsa 10.0.14a, which does not seem to properly
 support the Intel HDA chipset.  So, I wanted to upgrade to Alsa 1.0.16.  I
 downloaded all available packages from the Alsa site, built and installed
 the driver (using --with-suse=yes), ran alsaconf and rebooted.  So far so
 good. I built the library, and installed that.  I try to build the utils,
 but it fails with the error no TLV support in alsa-lib.  I check the date
 on /usr/lib64/libasound.so.2.0.0, only to find it was NOT updated with the
 libray installation, yet the version in /usr/lib was updated.  Temporarily
 symlinking libasound.2.0.0 in /usr/lib64 to /usr/lib allows the
 compilation of alsa-utils to proceed and complete without error

The symlinking is senseless in this case - /usr/lib64 is meant to be populated
by 64 bit .so files while /usr/lib by 32 bit ones.

Applications complain because they encounter wrong (64 - 32) files.

The rest will hopefully be addressed by ALSA developers.

Regards,
  Sergei.

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Re: [Alsa-user] Metallic recording in cs46xx on Thinkpad T22

2008-03-11 Thread Gadi Oron
Hi James, thank you for the reply.

 Each time you start to record you have a
 10% chance of having the recording completely
 distorted and having a metallic sound.

 I know that sound.  And it is quite ugly.  On my snd-hda-intel
board(nVidia MCP61), I have to increase the number of periods to overcome
this sound.  default of 2   increased to 3 and all was fine, in jackd -n
3.  For arecord you might look at different than default buffer/period
sizes.  It's basically a latency issue.

 Other considerations are to give the audio group realtime permissions so
things like ethernet traffic doesn't cause clicks and other distortions in
the sound.
/etc/security/limits.conf

Sadly, I've already tried all of these to no avail.

The only thing I can't do is to have the soundcard have it's own IRQ - I
allways get yenta together with it.

Someone knows how to disable it or change it's IRQ?

Ciao
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Re: [Alsa-user] One process per audio channel

2008-03-11 Thread John Sigler
John Sigler wrote:

 Jaroslav Kysela wrote:
 
 John Sigler wrote:

 I have an RME AES-32 PCI board which provides 4 stereo input channels 
 and 4 stereo output channels.

 (I'm using the hsdpm driver at this time.)

 I want to use one process per channel, i.e. process A handles stereo 
 input #1 (on the XLR connector #1), process B handles stereo input 
 #2, etc.

 The processes are independent, in that process A might be started, 
 and only several hours later, process B is started, then a few hours 
 later process A is killed and restarted.

 Is it possible to do that with the ALSA library?

 Yes, look for the dsnoop plugin configuration in alsa-lib.
 
 I see references to ipc in alsa.conf.
 
 My embedded kernel is compiled with
 # CONFIG_SYSVIPC is not set
 
 Does the dsnoop plug-in require kernel support?

# ./arecord -D dsnoop -vv -M -r 48000 -f S32_LE -t raw -d 60 /tmp/out
ALSA lib pcm_direct.c:1605:(snd1_pcm_direct_parse_open_conf) The field 
ipc_gid must be a valid group (create group audio)
arecord: main:564: audio open error: Invalid argument

OK. I need to create the 'audio' group.

# ./arecord -D dsnoop -vv -M -r 48000 -f S32_LE -t raw -d 60 /tmp/out
ALSA lib pcm_dsnoop.c:540:(snd_pcm_dsnoop_open) unable to create IPC 
semaphore
arecord: main:564: audio open error: Function not implemented

No cookie... I suppose I have to recompile a kernel?

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Re: [Alsa-user] Metallic recording in cs46xx on Thinkpad T22

2008-03-11 Thread James Shatto
 Sadly, I've already tried all of these to no avail.

What recording application are you using?  I've had issues where audacity would 
give me that metalic sound and ardour+jackd would not.  And vice versa.  
Depending on versions and whatnot.  

Beyond that I really can't offer any more insight without additional info.  
Like alsa version, kernel version, contents of /proc/asound/cards, .asoundrc, 
and whatever else might apply.  Aside from upgrading to the latest kernel and 
latest version of alsa.  It might be a known and already fixed issue.

For audacity, most times I end up compiling it from source with the 
--with-portaudio=v19 parameter to work around various issues.  Although it 
looks like debian caught on and now supplies a version with that option.

HTH

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Re: [Alsa-user] One process per audio channel

2008-03-11 Thread Darrell Bellerive
On Tuesday 11 March 2008 06:33, John Sigler wrote:
 I must say that the ALSA
 configuration files look like pure voodoo magic to me.

Me too. I really wish someone would write Ultimate ALSA Configuration for the 
Complete Idiot. It would be a book I would purchase.


-- 
Darrell Bellerive

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Re: [Alsa-user] One process per audio channel

2008-03-11 Thread Helge Fredriksen
I'll sign in on that!

Helge F.

On Tue, Mar 11, 2008 at 7:49 PM, Darrell Bellerive [EMAIL PROTECTED]
wrote:

 On Tuesday 11 March 2008 06:33, John Sigler wrote:
  I must say that the ALSA
  configuration files look like pure voodoo magic to me.

 Me too. I really wish someone would write Ultimate ALSA Configuration for
 the
 Complete Idiot. It would be a book I would purchase.


 --
 Darrell Bellerive

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[Alsa-user] asound.conf

2008-03-11 Thread Thierry Bouchard
Hi, I started to play with ALSA 2 days ago and Im trying to figure out how 
ALSA works and how to add PCM devices in the configuration file.

I wrote a simple plugin for my microphone which is supposed to convert the 
data into a 32 bps format. Here is how it looks like : 

pcm.jcb-in-1 {
type hw
card 0
device 2
}
ctl.jcb-in-1 {
type hw
card 0
}
pcm.MicPlug
{
type plug
slave
{
pcm jcb-in-1
format S32_LE
}
}

So now if Im opening the PCM device named MicPlug and start reading on it, Im 
expecting (which may totally be wrong) that the data I read will be in a 
32bps format, which is never happening. In fact, the data I read is exactly 
in the format that I set using  snd_pcm_hw_params_set_format. Is the 
conversion supposed to happen or am I expecting something that is totally 
wrong?

Also is there any good documentation about how the asound.conf file works? I 
found some stuff on the web but it rather sucks.

Thank you!

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Re: [Alsa-user] Metallic recording in cs46xx on Thinkpad T22

2008-03-11 Thread Rene Herman
On 11-03-08 08:24, Gadi Oron wrote:

 I am trying to record sound on an old Thinkpad T22 that uses the cs46xx 
 sound driver.
 
 Each time you start to record you have a 10% chance of having the 
 recording completely distorted and having a metallic sound. When you 
 look at the waveform it looks as though there are small segments with 
 sharp transitions between them, a little like if these segments were 
 moved a little from their correct place.

No insights, but I confirm the bug with a TerraTec DMX XFire 1024 (CS4624).

Rene.

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[Alsa-user] recording from /dev/dsp

2008-03-11 Thread James Shatto
record -D copy -f cd -t wav outfile.wav

This does not seem to capture any of the sounds from /dev/dsp.

ecasound -i:/dev/dsp -o outfile.wav

Nor does this.

It's been a while since I've done this, what am I missing?  Or is there 
something about usb-audio the prevents this from working?  Or some ./configure 
option I missed at compile time?  I know I've recorded what was coming out the 
speakers directly from the /dev/ device before.  I just can't remember how.  
I'm trying to capture some streaming audio.

.asoundrc below

-

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

pcm.atiixp_modem {
   type hw
   card 3
}
ctl.atiixp_modem {
   type hw
   card 3
}

pcm.usb_audio2 {
   type hw
   card 0
}
ctl.usb_audio2 {
   type hw
   card 0
}

pcm.usb_audio3 {
   type hw
   card 1
}
ctl.usb_audio3 {
   type hw
   card 1
}

defaults.pcm.card 0

pcm.copy {
  type plug
  slave {
pcm hw
  }
  route_policy copy
}


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Re: [Alsa-user] recording from /dev/dsp

2008-03-11 Thread James Shatto
 Whether this works will depend on your hardware.  Some devices support
 capturing the audio output, some don't.

Well, I swapped it around so that the onboard sound was card 0 and it works 
that way.  But the quality of what gets recorded is bad, actually hideous is 
more appropriate.  From studio quality to sounding like there's a waterfall in 
close proximity.  After normalizing to bring it to audible levels.  Not quite 
what I want.  Perhaps there's some other way to get flash audio extracted that 
isn't as hideous.

arecord -D hw:0,0 -t wav -f dat tempfile.wav

Works but hideous.  As I notice a button for downloading an .mp3 of the clip in 
the flash.  Which doesn't seem to work in linux.  As thoughts of booting vista 
makes my skin crawl...

http://www.studioauditions.com/jamroomsessions_home.php


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Re: [Alsa-user] Metallic recording in cs46xx on Thinkpad T22

2008-03-11 Thread vsobreip
On Tue, Mar 11, 2008 at 9:27 AM, Gadi Oron [EMAIL PROTECTED] wrote:

 Sadly, I've already tried all of these to no avail.

 The only thing I can't do is to have the soundcard have it's own IRQ - I
 allways get yenta together with it.

 Someone knows how to disable it or change it's IRQ?


Maybe you should prioritize your sound card in the PCI bus if your
motherboard is not PCIe by tweaking the PCI latency, I have very good
results with that:

http://www.sabi.co.uk/Notes/linuxSoundLatency.html
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