On Fri, Sep 01, 2006 at 03:40:23PM +0200, Tomi wrote:
> Ok hopefully this will get properly replied.
>
> For the subject line I have set this:
> Re: HOWTO post on mailing lists
>
> and for the TO address I've set this:
> alsa-user@lists.sourceforge.net
A real reply will also have In-Reply-To:
On Mon, May 22, 2006 at 12:33:46AM +0200, Ron Smits wrote:
> How hard would it be to make a dialog/druid/wizard thingy
> that would just ask you what speakerset you have and set up
> all the stuff that is needed to have great sound from all
> speakers whatever you throw at it?
Speaking as a (rath
On Sat, Apr 15, 2006 at 01:40:51PM +0200, David Dasenbrook wrote:
> Hello,
> I am trying to do some recording with my SoundBlaster Live, using the
> ALSA emu10k1 module. However, it turns out that I am unable to control
> recording (capture) and playback volumes of the LineIn independently.
> Whe
On Thu, Mar 02, 2006 at 09:32:00PM +0100, Guennadi Liakhovetski wrote:
> This is what "chrt -f 40 arecord" does, isn't it? SCHED_FIFO with static
> priority 40, should be enough...
Should be :-) Sorry, I overlooked the second command.
--
FA
---
On Thu, Mar 02, 2006 at 09:03:29PM +0100, Guennadi Liakhovetski wrote:
> chrt -f -p 50 `pidof "IRQ 5"`
> chrt -f 40 arecord -t wav -f cd -d 600 test.wav
>
> and if there's accidentally an updatedb running in the background, I get
> lots of "Overrun!!! (at least xxxms long)", where xxx can be hun
On Wed, Feb 08, 2006 at 07:57:43PM +0100, Joerg Kampmann wrote:
> I have to play with the mixer (kamix) to get signals to jaaa. This is
> quite strange behaviour. It seems to have changed since 9.0 ...
I recently moved from 9.2 to 10.0 on one of my machines and got
some surprises as well...
Nor
On Wed, Feb 08, 2006 at 12:13:14PM -0500, Paul Fox wrote:
> if it's not clear by now, i'm not looking for professional or
> audiophile quality. i'm hoping someone will tell me that
> something like this turtle beach product works perfectly:
>
> http://www.turtlebeach.com/site/products/audioa
On Wed, Feb 08, 2006 at 06:20:02PM +0100, Joerg Kampmann wrote:
> I have a Siemens Fujitsu Amilo M 7400 with an Intel 82801 AC97 chip onboard.
>
> coming from Suse 9.0 to 9.3 I have a problem with "jaaa" and Alsa:
>
> I do an "strace" like in next line and get the following results
> including t
On Tue, Jan 31, 2006 at 05:34:39PM +0100, Clemens Ladisch wrote:
> snd_usb_audio is broken in both 2.6.14 and 2.6.15 (2.6.15.2 or ALSA
> 1.0.11rc3 are OK).
I installed 2.6.15.2 today - the problem remains. Added the
rt-preempt patch (the latest one applied with a few small shifts),
and modified I
On Mon, Jan 30, 2006 at 07:05:34PM -0500, Lee Revell wrote:
> On Tue, 2006-01-31 at 01:01 +0100, fons adriaensen wrote:
> > > What are the ALSA versions used by these?
> >
> > 1.0.9b
>
> Any chance you can try the latest ALSA version and/or kernel to rule out
>
On Mon, Jan 30, 2006 at 06:39:57PM +0100, Clemens Ladisch wrote:
> Alfons Adriaensen wrote:
> > This weekend I installed SL10.0 on my Thinkpad. Everything runs fine
> > except for USB audio (EDIROL UA-2). I had it working perfectly with
> > -p 256 -n 3 on SL9.2, but now even -p 1024 gives problems,
Sergei,
> Whatever implementation of yours should be limited to simple
> FFT-based equalizer similar to the one I published. That is,
> data from no more than two adjacent FFT buffers can be used.
I will take up this challenge, but not immediately. Right now,
I'm working until almost 3 hours afte
On Tue, Jan 03, 2006 at 03:12:50AM +0200, Sergei Steshenko wrote:
> In the case of DFT limited N means both limited Q and discrete
> central frequencies in the terms of precise signal restoration.
It means limited Q, but no discrete central frequencies.
BTW, I've used such 'interpolated frequency
On Mon, Jan 02, 2006 at 04:52:42PM -0800, Bill Unruh wrote:
> Once you have made teh discrete sampling you have lost the original signal.
> It is gone. You cannot reconstruct it.
> IF you assume that the original signal is frequency limited, then you may
> be able to reconstruct it.
Sigh. Of cou
On Tue, Jan 03, 2006 at 01:22:56AM +0200, Sergei Steshenko wrote:
> - do you agree that if, say, I have an 8 point FFTW, the following
> frequencies are represented in the FFTW output array C (the result of time ->
> frequency conversion, i.e. direct FFT):
>
> C[0] <=> DC (only real part)
>
On Mon, Jan 02, 2006 at 01:24:59PM -0800, Bill Unruh wrote:
> >Even PLLs are often implemented using a VCXO (voltage controlled
> >xtal oscillator). You get the stability of the external reference,
> >and outside the loop bandwidth, the quality of an xtal. Standard
> >practice in all sorts of tele
On Mon, Jan 02, 2006 at 11:37:54PM +0200, Sergei Steshenko wrote:
> So, how are going to implement a central frequency which is
> not a multiple of (Fs / N) in a DFT equalizer ?
>
> That is, what data resulting from direct DFT represents such frequencies ?
A weighted sum of some DFT outputs, ins
On Mon, Jan 02, 2006 at 07:58:57PM +0200, Sergei Steshenko wrote:
> The existence of spectral resolution prevents end user from having
> arbitrary central band frequencies in DFT-based equalizers, central
> frequencies
> can only be a multiple of spectral resolution.
Not true, you can have arbit
On Mon, Jan 02, 2006 at 06:09:17PM +0200, Sergei Steshenko wrote:
> In other words, tunable xtal is a bad xtal by definition.
There are no such things as 'tunable' and 'untunable' xtals.
*Every* xtal behaves has a parallel or series LC circuit near
resonance (depending on how it's used) and can b
On Mon, Jan 02, 2006 at 03:14:14PM +0100, Asbjørn Sæbø wrote:
> What I really would like is a sound card with adjustable sampling
> frequency. That is, small scale tunable on an almost continous scale.
> Practiaclly, I think that one possible way to do it would be to use a
> temperature sensiti
On Fri, Jan 02, 2004 at 11:40:18PM -0800, Mark Rainess wrote:
> You may need to use an impedance matching transformer,
> or a microphone preamplifier, between your guitar and
> the microphone input on your sound card. Your sound
> card probably has a "high impedance" input, and I
> think all elect
On Mon, Dec 08, 2003 at 03:44:53PM -0500, brett holcomb wrote:
> What sound cards are used by the people on this list -
> especially the developers . I currently have a Turtle
> Beach Santa Cruz and want to upgrade. I have been looking
> at cards like the Midiman Delta 44 or others in it's cl
Hello list,
I need some advice on the ALSA sequencer interface. In particular I'd
like to know if the following code can be safely used inside a real-time
audio thread, i.e. is it guaranteed not to block ?
As far as I've been able to find out, it should be safe.
snd_seq_event_t *E;
if (snd_seq_
On Mon, Aug 11, 2003 at 05:51:32PM +0200, Takashi Iwai wrote:
> did you update alsa-tools, too?
> there was a bug in envy24control.
> anyway, the DACs should be controllable via alsamixer, too...
I tried to compile the latest envy24control, but it depends on a lot
of libs or includes that I don'
Hello list,
Another problem after installing my Terratec (but this one should be easy...)
I have in my .asoundrc
pcm.TT {
type hw
card 1
device 0
}
pcm.TT12 {
{
type route
ttable.0.0 1
ttable.1.1 1
slave.pcm TT
}
The first is the raw EWS88MT.
TT12 is supposed to be a 2-channel
On Fri, Aug 01, 2003 at 08:45:00PM +0200, Jaroslav Kysela wrote:
> On Fri, 1 Aug 2003, Fons Adriaensen wrote:
>
> > On Fri, Aug 01, 2003 at 08:13:33PM +0200, Jaroslav Kysela wrote:
> >
> > > Could you try the latest ALSA code? I think that we fixed something
> >
from without a power cycle.
System verison here is 2.4.20 (SuSE 8.2) and ALSA 0.9.0.cvs20030217-23.
I haven't seem traces of a similar problem in the list archives.
Any help or tips appreciated !
--
Fons Adriaensen
---
This SF.Ne
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