Oops, I sent it to a list were I'm not subscribed too.
Here it is, on the right place:
-------- Original Message --------
Subject: Re: At the hands of Professor Keller and Raymond
Date: Mon, 27 Jul 2009 17:48:15 +0200
From: Ralf Mardorf <ralf.mardorf(a)alice-dsl.net>
To: Robert Keller <keller(a)cs.hmc.edu>
CC: laseray(a)gmail.com, LAU <linux-audio-user(a)lists.linuxaudio.org>
References: <200907251343.51403.laseray(a)gmail.com>
<200907261032.12585.laseray(a)gmail.com> <4A6C6DBB.5060601(a)alice-dsl.net>
<200907261227.30470.laseray(a)gmail.com> <4A6CB0DC.4050505(a)alice-dsl.net>
<12BD32E0-7A32-4A2A-AF72-775796462294(a)cs.hmc.edu>
Robert Keller wrote:
> [...]
> As I am not formerly a member of the linux-audio-dev community, I just
> now joined as a result of this.
> [...]
You're welcome :)
excuse me for my first mail. It's awkward manipulative.
LAU isn't a court, so we need not to judge who is the good and who is
the bad guy. I assume that Raymond and you aren't bad.
I don't understand the GPL and I'm sure that I'm not the only one.
Anyway, I do understand what's the idea behind FLOSS.
I'm convinced that it's allowed to do what you are doing, but for FLOSS
and the world it would be good if knowledge would be shared, because
this is the ideology. Everybody would benefit if you forge Impro-Visor.
Most of us understand, Raymond too, that you first want to do your
project for real and that you are the head of this project, but the day
should come, when you share the source code.
I'm not speaking for the list, this are just my 2 Cents.
Thank you for your reply.
Best,
Ralf
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Hi every one.
This is my first mail to the list.
Hope I can help you as well.
I am using the fftw3 library to implement a transform in a array of 5120
elements .
I have this array from a program that I created modifying the arecord
program.
If someone could gimme a little help solving that problem.
I have that array been filled with value in this function:
static void compute_max_peak(u_char *data, size_t count)
{
signed int val, max, max_peak = 0, perc;
size_t ocount = count;
int arrayt[5120][1], l, cc;
int t = 0;
int tcount = 0;
//zero o array
for (cc = 0; cc <= 1; cc++) {
for (l = 0; l <= 5119; l++) {
arrayt[l][cc] = 0;
}
}
signed short *valp = (signed short *)data; //Dados no buffer
signed short mask = snd_pcm_format_silence_16(hwparams.format);
//Frames de silencio
count /= 2; // 5120
printf("Count = %i\n", count);
while (count-- > 0) {
val = *valp++ ^ mask;
val = abs(val);
// armazeno dados nos indices [x][0]
* for (cc = 0; cc <= 0; cc++) {
for (l = tcount; l <= tcount; l++) {
printf("TCOUNT = %i <=> cc = %i\n", tcount, cc);
arrayt[l][cc] = val;
}
}*
tcount++;
if (max_peak < val) //Armazena a maior amostra em max_peak
max_peak = val;
printf("val[%i] = %i\n", t, val);
t++;
}
fftw_calculation(arrayt);
for (cc = 1; cc <= 1; cc++) {
for (l = 0; l <= 5119; l++) {
arrayt[l][cc] = 0;
}
}*/
max = 1 << (bits_per_sample-1); //2#15 = 32768
printf("Max peak (%li samples): %05i (0x%04x) \n", (long)ocount,
max_peak, max_peak);
perc = max_peak * 100 / max;
for (val = 0; val < 20; val++)
if (val <= perc / 5)
putc('#', stdout);
else
putc(' ', stdout);
printf(" %i%%\n", perc);
for (cc = 0; cc <= 1; cc++) {
for (l = 0; l <= 5119; l++) {
printf("arrayt[%i][%i] = %i\n", l, cc, arrayt[l][cc]);
}
}
}
and the transform is calculated by this fucntion:
fftw_calculation(int data[5120][1]) {
fftw_complex *out, *in;
fftw_plan p;
int N, l, cc;
N = 1024;
in = (fftw_complex*) fftw_malloc(sizeof(fftw_complex) * N);
out = (fftw_complex*) fftw_malloc(sizeof(fftw_complex) * N);
in = (fftw_complex *)data;
for (cc = 0; cc <= 1; cc++) {
for (l = 0; l <= 1023; l++) {
out[l][cc] = 0;
}
}
p = fftw_plan_dft_1d(N, in, out, -1, FFTW_ESTIMATE);
fftw_execute(p);
for (cc = 0; cc <= 1; cc++) {
for (l = 0; l <= 1023; l++) {
printf("out[%i][%i] = %i\n", l, cc, out[l][cc]);
}
}
//fftw_destroy_plan(p);
}
The problem is, I can't user N as any number higher than 1024, or I get
segfault.
But if you see, the array been transformed is a array of [5120][1] elements.
Is there anybody here familiar with the fftw3 implementation.
Tks in advanced.
Having a SIP connection can be a god-send when you have lots of strings
to figure out verbally ..
Jeff Pulver used to have a free sip service at pulver.com, but a year
ago (or some such) he warned that the service would be discontinued.
Something to do with Facebook premium service IIRC ...
Problem: Although Ekiga.net works just fine for me, I cannot set up my
significant other on that network. It appears like it only works if you
have an Ekiga client?
There is no Ekiga client for ancient Mac's ...
Question: Any suggestions for an alternative SIP service?
Hi,
I'm working toward getting the UA-101 to work on Linux. I've had a look
at the Alsa drivers, and I after 23s of playback I hit the known glitch
bug:
http://lists.linuxaudio.org/pipermail/linux-audio-user/2009-April/059447.ht…
I've tried fiddling with various alsa buffering settings to get a point
solution that works, but with no luck.
Comparing the USB traffic on Windows and Linux was interesting. The
Linux trace has the sound samples being sent to USB Endpoint 2, but the
Windows trace is weird.
The both traces the sound sample was a long period of silence followed
by a sine wave. There are several things 'odd' under windows:
The data send to the sound card (on USB EP 2) started out as as string
of '0x00 0x00...' as you would expect for a 'silence' sample. After
about 2 seconds, 'silence' is send as lots blocks like
00000000: 00 00 ff ff 00 ff fe ff 00 ff ff ff 00 ff ff ff
00000010: 00 00 00 00 00 ff ff ff 00 00 00 00 00 ff ff ff
in a pattern I haven't got to the bottom of yet.
The second difference I've noticed is that on Windows the UA-101 is
sending data to the PC (on USB EP 81). This is 'random' data (but it
does have a lot of zeros in it), I haven't investigated it yet:
00000000: 00 a9 fb ff 00 e5 00 00 00 65 00 00 00 11 00 00
00000010: 00 5c 00 00 00 15 00 00 00 3f 00 00 00 f9 ff ff
5760 bytes are sent to the sound card, then 9120 bytes are sent back.
This cycle repeats every ~4.6ms. That is about 26 bytes/sample out and
41 bytes/sample back. I'll take some more accurate measurements when I
have a parser for the log format.
On Linux there is no data being read back.
My next move is going to construct some .wav files that differ in a few
samples and compare the traces that they produce.
I have a couple of questions:
* Does anyone know the background behind "I think that's the mentioned
synchronizing-problem in the UA-1000 quirk ('FIXME: playback must be
synchronized to capture')"?
* Any idea what could be causing the glitch?
* Has anyone done any analysis on this problem before?
Cheers,
Phil
Hi,
Looking for a distribution-agnostic, community-maintained place for gathering
linuxaudio-related documentation, wiki.linuxaudio.org seems to be one of the
main resources.
The visual style seemed a bit old-fashioned and cluttered. I took a stab at
cleaning it up a bit:
http://arnout.engelen.eu/files/dev/linuxmusicians/linuxaudiostyle/
How do you like it?
'wiki.linuxaudio.org' and 'apps.linuxaudio.org' have a different logo and a
different start page, but apart from that share the exact same content. This
is pretty confusing. Wouldn't it be more elegant to make 'apps.linuxaudio.org'
redirect to something like 'wiki.linuxaudio.org/apps' ?
Regards,
Arnout Engelen
Hi,
I am running a .mp3 file using 'xine' without any issues on my Fedora
Core 8 machine.
I am reading /dev/dsp and storing it in some file.
i.e. cat /dev/dsp > musicfile
Now I am stopping 'xine' and then writing 'musicfile' to /dev/dsp.
i.e. cat musicfile > /dev/dsp
In this case I can hear the music but with a lot of noise in it.
Is there any way I can remove/reduce this noise from 'musicfile'?
I also tried following program (it records the voice and plays it
immediately) that I got from net to understand how to write to /dev/dsp
but it does not seem to record my voice and hence plays nothing and
there is noise only.
-------------------------------------------------------
#include <unistd.h>
#include <fcntl.h>
#include <sys/types.h>
#include <sys/ioctl.h>
#include <stdlib.h>
#include <stdio.h>
#include <linux/soundcard.h>
#define LENGTH 3 /* how many seconds of speech to store */
#define RATE 8000 /* the sampling rate */
#define SIZE 8 /* sample size: 8 or 16 bits */
#define CHANNELS 1 /* 1 = mono 2 = stereo */
/* this buffer holds the digitized audio */
unsigned char buf[LENGTH*RATE*SIZE*CHANNELS/8];
int main()
{
int fd; /* sound device file descriptor */
int arg; /* argument for ioctl calls */
int status; /* return status of system calls */
/* open sound device */
fd = open("/dev/dsp", O_RDWR);
if (fd < 0) {
perror("open of /dev/dsp failed");
exit(1);
}
/* set sampling parameters */
arg = SIZE; /* sample size */
status = ioctl(fd, SOUND_PCM_WRITE_BITS, &arg);
if (status == -1)
perror("SOUND_PCM_WRITE_BITS ioctl failed");
if (arg != SIZE)
perror("unable to set sample size");
arg = CHANNELS; /* mono or stereo */
status = ioctl(fd, SOUND_PCM_WRITE_CHANNELS, &arg);
if (status == -1)
perror("SOUND_PCM_WRITE_CHANNELS ioctl failed");
if (arg != CHANNELS)
perror("unable to set number of channels");
arg = RATE; /* sampling rate */
status = ioctl(fd, SOUND_PCM_WRITE_RATE, &arg);
if (status == -1)
perror("SOUND_PCM_WRITE_WRITE ioctl failed");
while (1) { /* loop until Control-C */
printf("Say something:\n");
status = read(fd, buf, sizeof(buf)); /* record some sound */
if (status != sizeof(buf))
perror("read wrong number of bytes");
printf("You said:\n");
status = write(fd, buf, sizeof(buf)); /* play it back */
if (status != sizeof(buf))
perror("wrote wrong number of bytes");
/* wait for playback to complete before recording again */
status = ioctl(fd, SOUND_PCM_SYNC, 0);
if (status == -1)
perror("SOUND_PCM_SYNC ioctl failed");
}
}
---------------------------------------------------------------------------------------
Regards,
Girish
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This should settle any doubts people have about there being a GPL violation,
along with understanding how the GPL is properly applied. Here is part of
the header from a file in the previous version of Impro-Visor. No notice to
any change was given with the current preview version and jMusic code is still
used in it. Therefore, it be logically deduced that the GPL still applies:
/**
* This Java Class is part of the Impro-Visor API
*
* Copyright (C) 2005-2008 by Robert Keller and Harvey Mudd College
*
* It has been edited from the jMusic API version 1.5, March 2004.
* Copyright (C) 2000 Andrew Sorensen & Andrew Brown
*
* Impro-Visor is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
...
Raymond
Hi all,
Hermann, don't sell it yet as "clean classes" :lol:
What I did so far is prepare the ground for that but this is not there yet. I split the source files and gathered stuff into different namespaces. But it will make it easier to turn the whole project as a real OO app. Not there yet but soon. I have some experimental code to finish up before this happens. But the project is definitely more readable now.
But I will be away for almost a month, far from any PC or internet for that matter so nothing from my part will happen before I come back from this (well deserved) trip. I may have to restructure a bit but not as drastic as this. Until then, enjoy guitarix 0.05 :)
Cheers!
J.
--- On Sun, 7/19/09, hermann meyer <brummer-(a)web.de> wrote:
> From: hermann meyer <brummer-(a)web.de>
> Subject: Re: [LAD] Faust source code of Guitarix?
> To: "Stéphane Letz" <letz(a)grame.fr>
> Cc: "Linux Audio Developers" <linux-audio-dev(a)lists.linuxaudio.org>
> Date: Sunday, July 19, 2009, 10:15 AM
> Am Mittwoch, den 15.07.2009, 22:19
> +0200 schrieb Stéphane Letz:
> > Hi,
> >
> > I don't see the Faust code (the .dsp file) that has
> been used to
> > produce Guitarix (http://sourceforge.net/apps/trac/guitarix/browser/
> > trunk/src. It the Faust source code available
> somewhere?
> >
> > Thanks
> >
> > Stephane
> > _______________________________________________
>
> Hi, Stephane
>
> So I have upload the faust files to
> http://sourceforge.net/apps/trac/guitarix/browser/tags/faust
>
>
> thanks for remember me on that, I have always forget to
> upload them.
>
> One word one the guitarix source, there have been a drastic
> change now,
> James Warden have rework the hole source and bring it in a
> object
> orientatet form. All is split in to clean classes, witch
> will build as
> objects be for they linked. Those with interest on C++
> sources could
> have a look at the new source form under /trunk
>
> And if you try it out, feedback is very match welcome :-)
>
> regards
> hermann
>
> _______________________________________________
> Linux-audio-dev mailing list
> Linux-audio-dev(a)lists.linuxaudio.org
> http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
>
Hi,
I don't see the Faust code (the .dsp file) that has been used to
produce Guitarix (http://sourceforge.net/apps/trac/guitarix/browser/
trunk/src. It the Faust source code available somewhere?
Thanks
Stephane
