Hi,
Let me inform you about the online demonstration against the
patentability of software (=software patens) which the Commission of
the European Union wants to introduce. As far as I understand the
process of passing the appropriate bill in the institutions of the EU,
the patentability of software could be possible from July 2005 on.
There is a site, run by among others Attac Germany, where you can
protest against this, by uploading a picture of yourself:
http://demo.stoppt-softwarepatente.de/index.php?content=demo&lang=GB
"So far  3883 people have demonstrated here. The more people
participate, the clearer a joint picture of the protest will be shown.
Short before the decision in the European Parliament we will print this
picture on a huge banner and show it in front of the European
Parliament in Strasbourg. This way you can also be a part of the
international demonstration."
Regards,
Jens
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Albert Graef writes:
> that's great news.
Certainly is!
> Has Rosegarden been updated to a newer Lily yet? Last
> time I looked, they were still generating 2.0 code. :(
Rosegarden's been able to generate 2.2 code for ages, and 2.4 for a while - though I can't quite remember whether that was in the 1.0 release or not.
We're partly but not completely up to date with 2.6. I might well look at that this week, unless anyone else offers. Our Lilypond output is not terribly well structured for any version, so a bit of an overhaul might be nice too.
Chris
Hi all,
I've just released my Faust module for the Q programming language. A
realtime synth application based on this module, QFSynth, is also available.
Faust (http://faudiostream.sf.net) is Grame's functional DSP programming
language. Q (http://q-lang.sf.net) is a general-purpose functional
programming language with an extensive collection of multimedia-related
modules. If you're a developer interested in modern FP tools for
multimedia and DSP programming, you should definitely take a look. ;-)
Cheers,
Albert
--
Dr. Albert Gr"af
Dept. of Music-Informatics, University of Mainz, Germany
Email: Dr.Graef(a)t-online.de, ag(a)muwiinfa.geschichte.uni-mainz.de
WWW: http://www.musikwissenschaft.uni-mainz.de/~ag
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hi,
This might be a a newbie question, but i have openal installed and i am
running into problems. i have also posted to openal-dev list about my
concerns.
I have created an ~/.asoundrc file for help with dmix, because i have
had problems playing multiple sound programs at the same time. This
.asoundrc file helped me use xmms and amarok at the same time.
1. My problem is when i am trying to run an openal program when i am
using one of the other sound programs, the program ran last will block
untill the first program that i started had finished. This seems
strange to me that some programs will "dmix" and some will not. Any
ideas on this?
Also, I am thinking about developing and audio app using the openal api,
and i had some questions that maybe someone could clear up for me.
1. I am assuming this api wraps around alsa or oss, i am going to be
using an RME hammerfall dsp and i need to attach some inputs to some
outputs to allow the input from a mic to go to the output which will go
to their headset, (low latency Monitoring). How is this done, as i have
not found any docs talking about how to send sounds out certain channels?
2. Some examples i have found do not compile because they are including
the file #include <strstring.h> , which i do not have. (i am running
gentoo 2005.0, kernel-2.6.11)?
is this a known issue?
thanx in advanced
bj
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Hi,
i added an experimental feature to libconvolve/jack_convolve which might
be able to preserve some of those precious cpu cycles from being burnt.
jack_convolve now has commandline switches --min_bin=bin_no and
--max_bin=bin_no which can be used to specify which bins of the fourier
transformed signal and response to multiply. The range for both is
always 0..periodsize+1, but min_bin must always be < than max_bin..
An example:
assuming jack periodsize of 2048 and samplerate of 48khz:
jack_convolve response_file.wav --max_bin=1200
This leaves out the top 849 bins out of the multiplication and thus
reduces cpu load by ca. 2/5. The cost is that the high frequency
spectrum is cut off. And it's not even a really clean cutoff which
jack_convolve response_file.wav --max_bin=40
would show (you get crackles due to edge effects, etc).
This feature is probably most useful to get some preview with degraded
quality but less cpu consumption. Although at some samplerates it even
makes sense.
Assume a samplerate of 96khz, then there's quite a bit of signal which
doesn't need to be processed since it's far out of the range of human
perception.
the min_bin parameter is not really useful and only included for
completeness sake :) It cuts out low frequency bins out of the equation.
Due to the crackle effects when throwing away bins that are audible one
should not use these settings for filtering of the convolution in the
audible frequency range..
In one of the previous releases (which went unannounced i think) i also
added a gain argument which can be used to boost the level of the
convolution output.
Example:
jack_convolve response_file.wav --gain=4.0
to get a 300% increase in level..
Regards,
Flo
P.S.: I am also currently working on a qt app which uses libconvolve,
but it might still take a while until initial release (due to my time
constraints induced by studying). Preliminary screenshot here:
http://www.affenbande.org/~tapas/wordpress/?page_id=27
Help would, of course, be appreciated :) Anyone know of waveform or vu
meter widgets for qt which are reusable?
Flo
--
Palimm Palimm!
http://affenbande.org/~tapas/
> (*) Recent experiments by prof. Angelo Farina (Univ. of Parma, Italy)
> suggest strongly that when the DA conversion is done properly, there is
> no audible difference between a sample rate of 48 kHz and any higher value.
> OTOH, he only found one type of DAC that was good enough in order to be
> completely free of audible artefacts at 48 kHz (by Apogee, and they are
> quite expensive).
>
Is there an online publication of Mr Farina's findings? Google is not turning
up anything definitive for me.
-Ben
Hi.
I am sure this sounds a bit crazy to some of you, but be ensured,
I am not (yet) completely out of my mind and there are actually
real reasons for doing what I do :-).
I am now running jackd from within Emacs, using jack.el.
Find it here: http://delysid.org/emacs/jack.el
Basic usage:
* Install the elisp file and put (require 'jack) in your .emacs.
* Customize startup options: M-x customize-group RET jack RET
* I.e. jack-sample-rate and jack-period-size are two important candidates.
* Start jack: M-x jack-start
You will see the verbose output in a special buffer called *JACK output*.
* If you want/need to get rid of your jack instance, simply do
M-x jack-kill RET
* If you changed sample rate or any other startup parameter, do
M-x jack-restart RET
to make them effective.
Plans:
* Do something useful with xrun output, maybe some stats (suggestions?).
* Add support for dummy and oss drivers, and add remaining ALSA
params.
* Write a patchbay based on jack_lsp and jack_{dis,}connect.
Why am I doing this?
You might have noticed I am recently seriously working on
Emacs extensions related to Sound and audio. osc.el is working
nicely and om.el, a client to drobillas om-synth is just waiting
for a polish up. midi.el is parsing binary data already and
needs a bit more careful planning before it can really grow.
The idea behind jack.el was that I basically got fed up by jack
wasting either a virtual console, or a screen window. Both dont really
offer confortable scrollback, so the idea to make jack run from
within Emacs, and collect all its output into a Emacs buffer was born.
Some nice sideeffects: Load statistic is collected in a variable,
so if you are into elisp, you can do fancy stuff like:
(with-current-buffer (jack-output-buffer) jack-load)
to retrieve the current load :-)
Combine all this with things like csound-x (Csound support for Emacs)
and the SuperCollider Emacs frontend, and you get a truly
powerful composition and experimentation environment.
One day I'll learn CLM and CM, and add it to the mix as well :-)
--
CYa,
Mario
Hi all,
Linuxtag[1] 2005 is over. It took place from 20050622 to
20050625 in Karlsruhe, Germany.
We had a booth there populated by the following people:
Arnold Krille
Christoph Eckert
Frank Neumann
Gerd Flaig
Joachim Schiele
Moritz Dressler
Reinhard Katzmann
We had 4 presentation places with a keyboard and a notebook
each, additionally an acoustic guitar and an electric violin
so we have been able to show soft synths as well as realtime
audio processing. Needless to say that there have been lots
of USB audio devices :) .
There was a lot of interest of the visitors in what we can do
with free audio software, and my observation is that most
people have been simply surprised by the amount of available
applications as well as the quality of the apps and the
processed audio material.
Furthermore I had the occasion to hold two talks about linux
audio, one for musicians[2] (en) and one for desktop users[3]
(de).
Because it has been a successful show with really great
software, I'd like to thank all who helped and especially
those who do the hard background work on free audio software.
Best regards
ce
[1] http://www.linuxtag.org/typo3site/8.0.html?L=1
[2] http://www.linuxtag.org/typo3site/
freecongress-details.html?&L=0&talkid=145
[3] http://www.linuxtag.org/typo3site/
freecongress-details.html?&L=0&talkid=300