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
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
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.
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
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
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
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
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
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
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
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
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
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
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
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 {
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
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
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
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
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