On Fri, 30 Dec 2005, Adam Nielsen wrote:
What are you trying to do?
I'm still trying to get stereo output when passing the sound through a
LADSPA plugin (which in the latest version of ALSA converts any incoming
stereo source into mono.)
Could you enter a bug report if the current LADSPA
On Fri, 30 Dec 2005, Adam Nielsen wrote:
ALSA lib pcm_ladspa.c:1283:(snd_pcm_ladspa_parse_ioconfig) Unable to find an
audio port (1) for channel 1
aplay: main:547: audio open error: Invalid argument
So what does this mean? It looks like solving this problem will fix the
mono-only LADSPA
Hello and a very happy new year to you all,
I posted this (or something similar) a while back and had no reply, so
I'm trying again in case someone new spots it.
Trying to route all sound to the spdif output of my onboard nForce4
(NVidia CK804 #9474;
#9474;
ALSA lib pcm_ladspa.c:1283:(snd_pcm_ladspa_parse_ioconfig) Unable to
find an audio port (1) for channel 1
It means that the used LADSPA plugin has no second audio port.
When you say no second audio port is that different to processing a
stereo signal? Because it was my understanding that
On Mon, 2 Jan 2006, Adam Nielsen wrote:
ALSA lib pcm_ladspa.c:1283:(snd_pcm_ladspa_parse_ioconfig) Unable to
find an audio port (1) for channel 1
It means that the used LADSPA plugin has no second audio port.
When you say no second audio port is that different to processing a
stereo
On Mon, 2 Jan 2006, Jaroslav Kysela wrote:
Running speaker-test -c2 -Dplug:test only plays the front right
audio out of both front speakers.
Confirmed. It seems that the problem has nothing with the ALSA LADSPA
plugin code but with the latest mmap code optimizations. I'm still
figuring
The attached patch fixes this bug.
Brilliant! The LADSPA EQ is now in stereo again! Thanks for tracking
this down and fixing it!
Cheers,
Adam.
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On Fri, Dec 30, 2005 at 11:47:41PM -0500, Ross Vandegrift wrote:
On Fri, Dec 30, 2005 at 05:42:32PM -0500, Lee Revell wrote:
LDAS (the low delay audio streamer) does something similar - it has to
compensate for the drift between two machines clocks' that exchange
audio over the network but
I will be away from the office until Tuesday, January Third. I will receive
your email and will get back to you as soon as possible.
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On Fri, Dec 30, 2005 at 05:36:28PM -0500, Ross Vandegrift wrote:
[...]
Say I have two sound cards in my box. Each sound card has an
independent crystal that generates a clock for timekeeping. The
crystals are rated to resonate at some fixed frequency, but there will
always be a margain of
To better measure actual drift one has to use really good LPF, which
means they have a pretty big delay.
So, it doesn't make sense to use a very good sample rate conversion
because we learn bout the drift well AFTER it occurs.
Good sample rate conversion makes sense only if clock frequencies
of
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 sensitive
Even X-tal oscillators
can be tuned over a small range by a voltage controlled capacitor.
one should remember that tunability is the opposite of stability/low jitter.
If we are talking about synchronization, master clock should be as stable as
possible (i.e. xtal, and no caps), and slave
Hi,
I have Sounblaster audigy se, but it didn't work..
My kernel version is: 2.6.14-5
Problem:
Soundcard is recognized, but it's sound is very distorted(you can only
recognize, that something is playing, ...)
on
http://www.alsa-project.org/alsa-doc/index.php?vendor=vendor-Creative_Labs#matrix
[EMAIL PROTECTED] wrote:
But it is impossible to download 1.0.11rc2.
How so?
It was placed on the web site today.
James
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Hello All,
please find the promised multichannel multiband FFT-based equalizer.
Attached is
README.multichannel_multiband_equalizer.txt
files.
Part two to follow will contain source tarball.
Regards,
Sergei.
1. LICENSE.
2. ACKNOWLEDGEMENTS.
3. DFT NUANCES AND THE EQUALIZER.
4. QUICK
Hello All,
please find the promised multichannel multiband FFT-based equalizer.
Attached is
mbeq_119700.tgz
file, it also contains the README file.
Regards,
Sergei.
mbeq_119700.tgz
Description: Binary data
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Hi all,Short time ago i've bought a laptop, this one has a target sound Realtek HD (ALC880). My debian regonizes my realtek while i am using alsa drivers 1.0.11 rc2, but i can't hear anything. I turn on my speakers sound up with the program Alsamixer from Alsa- Utils, but when i'm runnig
On Sun, 01 Jan 2006 13:16:41 -0500
Lee Revell [EMAIL PROTECTED] wrote:
On Sun, 2006-01-01 at 18:01 +0100, Nigel Henry wrote:
On Sunday 01 January 2006 13:12, Lorenz Hopfmüller wrote:
On Sat, 31 Dec 2005 17:45:09 +0100
Nigel Henry [EMAIL PROTECTED] wrote:
On Saturday 31 December
Hi all,
I have been given a Soundblaster X-Fi XtremeMusic as a christmas
present. I understand that the X-Fi boards have a wholly new chipset
and are not supported at all, but I did hear that one can get basic
functions by using the emu10k1 driver. Is this true? Or is the board
completely
Richard Wild wrote:
Hi all,
I have been given a Soundblaster X-Fi XtremeMusic as a christmas
present. I understand that the X-Fi boards have a wholly new chipset
and are not supported at all, but I did hear that one can get basic
functions by using the emu10k1 driver. Is this true? Or is
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
On Mon, 2 Jan 2006, fons adriaensen wrote:
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
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
On Monday 02 January 2006 20:50, Lorenz Hopfmüller wrote:
On Sun, 01 Jan 2006 13:16:41 -0500
Lee Revell [EMAIL PROTECTED] wrote:
On Sun, 2006-01-01 at 18:01 +0100, Nigel Henry wrote:
On Sunday 01 January 2006 13:12, Lorenz Hopfmüller wrote:
On Sat, 31 Dec 2005 17:45:09 +0100
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 ?
On Mon, 2 Jan 2006 22:36:21 +0100
fons adriaensen [EMAIL PROTECTED] wrote:
On Mon, Jan 02, 2006 at 07:58:57PM
Sure,
the bigger is the Q factor of an oscillator, the less tunable it is.
Its tunability depends also on the resonance mode number, actual
material, the way it was cut and G-d knows what else.
Look at the following analogy:
you have, say, parallel LC loop with VERY stable L and C.
Then you
On Mon, 2 Jan 2006 22:34:31 +0100
Nigel Henry [EMAIL PROTECTED] wrote:
On Monday 02 January 2006 20:50, Lorenz Hopfmüller wrote:
On Sun, 01 Jan 2006 13:16:41 -0500
Lee Revell [EMAIL PROTECTED] wrote:
On Sun, 2006-01-01 at 18:01 +0100, Nigel Henry wrote:
On Sunday 01 January 2006
On Mon, 2 Jan 2006 22:34:31 +0100
Nigel Henry [EMAIL PROTECTED] wrote:
On Monday 02 January 2006 20:50, Lorenz Hopfmüller wrote:
On Sun, 01 Jan 2006 13:16:41 -0500
Lee Revell [EMAIL PROTECTED] wrote:
On Sun, 2006-01-01 at 18:01 +0100, Nigel Henry wrote:
On Sunday 01 January 2006
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,
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 telecomm
Regarding
A weighted sum of some DFT outputs
- 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)
C[1], C[7] = 1 * Fs / 8;
C[2],
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)
C[1],
Fons,
when I said central frequencies I meant the frequencies which
can be reproduced exactly using one direct DFT and one inverse
DFT buffer.
That is, I claim that for the 8 points DFT/FFT example I gave if
input signal contains only
0 * Fs / 8
1 * Fs / 8
2 * Fs / 8
3 * Fs / 8
frequencies,
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 course
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'
Fons,
regarding,
It means limited Q, but no discrete central frequencies.
- please implement in the given 8 points example a filter an equalizer,
changing 1.5 * Fs / 8, but not affecting other frequencies.
To be more versatile:
Case_1:
input signal is
sin(2 * PI * 1 * Fs / 8) + sin(2 *
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 after
Fons,
we are making rounds
I want JUST to increase amplitude of 8.5 * Fs / N, and
not to touch anything else.
Your suggestion to also *decrease* the gain in 7 and 10
will change amplitudes of 7 and 10, and that's exactly what
I want to avoid.
I am trying to say that if one uses ONLY (k * Fs /
I am seeing this exact behavior on debian alsa 1.0.10-3 kernel version
2.6.14-3. Only sound from the front channels. I have followed the alsa
wiki page but with no luck for the rear / surround channels. Any help
is greatly appriciated.
-Paul
On Tue, 2005-12-13 at 00:13 +0100, Michal Medvecký
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