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[ Apologies for cross-posting. Feel free to forward around ]
Florence, 21 September 2004
+++ Media Innovation Unit Workshops at Firenze World Vision
Information for streaming
On Sep 23/24/25, 2004, Media Innovation Unit - Firenze Tecnologia will
host three workshops (on "Audio/Video streaming with Free Software",
"Custom Debian Distributions" and "Migrating to Free Software") during
Firenze World Vision 2004, which will be held in Florence (Italy).
It will be possible to remotely follow the workshops connecting to:
http://stream.bononia.it:8000/miu-fwv04.ogg
starting from 9:00 a.m.
During all seminars an IRC channel (server: irc.freenode.net, channel:
#miu-fwv04) will be open, where you will be able to ask your questions
to the speakers.
Special thanks to the Bononia team (http://www.bononia.it/) for their
technical support and for offering otheir bandwidth.
All presentations will be given in the english language. An
italian-english translation service will be available for the Q&A
session, after each talk.
Further information on the workshop will be published on:
http://www.miu-ft.org/what/events/20040923.fwv.str/
For more details, feel free to contact:
Nicola Giosmin <n.giosmin(a)miu.firenzetecnologia.it>
and/or our office in Florence:
Media Innovation Unit - Firenze Tecnologia
Borgo Albizi 15/a
50122 Firenze
Italy
Tel. no: +39 055 2001561
Fax. no: +39 055 2345762
+++
About Firenze Tecnologia: Firenze Tecnologia is the special agency of
the Chamber of Commerce of Florence devoted to study and facilitation
of technological innovation in the florentinian and tuscanian areas.
Please check: <http://www.firenzetecnologia.it/>.
About Media Innovation Unit: Media Innovation Unit is the research
unit of Firenze Tecnologia, devoted to research, development and
promotion of Free Software, Libre Content, Open Networks and New
Media.
About Firenze World Vision: Firenze World Vision is the event which
annually gathers the protagonists of innovation and of contemporary
communication processes, who compare their "visions" of the future.
Firenze World Vision deals with theory and practice, with ideas and
products, with global change and the ways our daily life is going to
change, well aware that the keys to read the contemporary world can be
found only through a multiform approach, without preconstituted views.
Please check: <http://www.firenzeworldvision.it/>.
Sorry, sent personal emails instead of list. Reposting...
> Hydrogen sounds like what you need:
> http://hydrogen.sf.net/
Thank you. I know :) But hydrogen just uses samples.
I can download TR-909 or TR-808 samples and use them in Specimen (or
Hydrogen). The thing is to generate my own different(!) sounds. The
only thing I know for now which can help is Stomper, but it does not
work under Wine :(
Drumatic-VST also segfaults.....
Uh...
> > Hydrogen sounds like what you need:
> > http://hydrogen.sf.net/
> not exactly, but is the best candidate i can think of for such a
> feature as being dicussed in this thread... i mean hydrogen could do
> quite well with some synthesis channels using the interface it has
> allready.
As far as I understand... Hydrogen is sample-player + drum-sequencer.
Do we need a kitchen sink there?
For me it would be better to have a separate drum-synth. So I can
drive it with seq24.
> Comix what do you say? :)
> show us the modular API to hook sample channels in hydrogen! :))
:)
It's time to port Stomper...I suppose.
JACK RELEASE 0.98.16
JACK is a low-latency audio server, written primarily for the GNU/Linux
operating system. It can connect a number of different applications to
an audio device, as well as allowing them to share audio between
themselves. Its clients can run in their own processes (ie. as normal
applications), or can they can run within the JACK server (ie. as a
"plugin").
JACK is different from other audio server efforts in that it has been
designed from the ground up to be suitable for professional audio work.
This means that it focuses on two key areas: synchronous execution of
all clients, and low latency operation.
JACK is available at http://jackit.sf.net
--CHANGES--
Buffer resizing enabled by default.
Added jack_ringbuffer_peak() to API.
Added jack_last_frame_time() to API.
--verbose will print the maximum usecs used on jackd termination.
Better compatibility with NPTL.
--version output changed for easier parsing.
New --unlock/-u option so that large libraries (gtk, qt, fltk, wine)
aren't memlocked.
Jack's tmp files now have the uid appended to them, so if there is a
crash, and then another user tries to use jack, it will still work.
New jack_create_thread() cleans up threading for portability. Available
for use by jack clients too.
New CoreAudio driver from the Jackosx project included in jack tree.
Prettier configure output.
and of course, updated documentation, better error reporting, and misc
internal fixes and cleanups.
I've released the pksampler and pkaudio packages.
PKAudio is an audio lib/daemon that makes asynchronous communication with a
realtime audio engine from high-level languages like python easy. It is
incredibly simple to write extension modules (subclass, implement two simple
methods, link with executable), and is a very nice place to start for someone
looking to get right to the important part of their dsp application. A Python
module is included.
PKSampler is a DJ sampler intended for a touch screen. It uses python and qt
to animate real 3D widgets created with PoVRay. Check out the screen shots!
I'm also working on PKStudio, which is essentially meant to be an open
replacement for Propellerhead Software's Reason. It focuses on tools for
extending pkaudiod to encourage others to write more dsp modules for it.
Read the pkaudio release notes for compilation help. should be simple.
PKSampler is written in python, so there isn't any setup.
http://pkaudio.sf.net/http://pksampler.sf.net/
There's lots of directions for this app to take, and I'd like to know if
anyone can get pksampler running on their machines. Works best with more than
one soundcard! Thanks!
>From: Jens M Andreasen <jens.andreasen(a)chello.se>
>
>Here is an early paper:
>
> http://www.cs.washington.edu/homes/oskin/thompson-micro2002.pdf
>
>It is about general-purpose-computing of which audio-processing is a
>subset. (BTW: Check out the aging references on the last page.)
I found nothing on the audio processing. So, the paper may be next to
nothing. But it was said that the GPU can be used as general purpose
computational unit for other applications.
In Europe, e.g., an apparatus, in which analog parts are replaced
by their digital equivalents, is not patentable. (Source: a patent
officer in a magazine.) So, if that big change is not patentable,
then why would a move of the audio application to GPU? Specially after
it is already said it could be used as an alternative to CPU.
We should just ignore the patents and keep up with the audio research.
Prior the patent the research has moved to GPU. We should not wait
20 years before we take first steps in researching the possibilities
of the GPUs.
Juhana
--
http://music.columbia.edu/mailman/listinfo/linux-graphics-dev
for developers of open source graphics software
The following proggi does the job:
#include <xmmintrin.h>
#define _MM_DENORM_ZERO_ON 0x0040
main()
{
// enable flush to zero
_mm_setcsr(_MM_FLUSH_ZERO_ON | _MM_MASK_UNDERFLOW | _mm_getcsr());
// enable denormals are zero
_mm_setcsr(_MM_DENORM_ZERO_ON | _mm_getcsr());
}
Note: you will need compiler option -mfpmath=sse and -march=pentium4
(works for both P4 and Athlon64)
After reboot, both FZ and DAZ flags are reset to zero, so you should run this
program in your /etc/profile. But as said earlier, works only for
applications that use (are compiled to use) the SSE unit instead of the old
style X86 FPU.
Here is one of music-dsp articles (critical date becomes important).
http://aulos.calarts.edu/pipermail/music-dsp/2004-May/027009.html
Perhaps they read that article, patented and implemented during the
summer. I don't know.
Juhana
== cut ==
Bill Baldwin wbaldwin at austin.rr.com
Sat May 1 15:18:11 PDT 2004
It's definitely possible, but far from convenient or optimal with today's
hardware.
The h/w is steadily getting better, but you'd likely run into FP precision
issues on many cards.
The new pixel shader 3.0 h/w is starting to appear and will become more
common over the next year - this extends the instruction length of shaders,
adds better flow control, and lots of other stuff that increases the
potential for doing interesting audio processing.
Getting the data in and out is another issue - you could treat an audio
channel as a 1-dimensional texture, turn off h/w filtering, etc. - but you'd
end up locking resources very frequently to write your input and read back
your output, which tends to stall the graphics pipeline and thus slow things
down.
Still, it's worth exploring given that all that GPU power that your
sequencer is ignoring...
-Bill
-----Original Message-----
On Behalf Of vesa norilo
Hi all,
Has anyone thought about it? I bet someone has.
There's plenty of horsepower sitting idle in PCs with modern video
cards. The in them shaders are completely programmable, but I don't
really know about streaming data to and fro. I just thought that it
would be very neat to write a VSTi synth that runs on, say, Radeon X800
or GF6800. Is it possible at all?
Vesa
http://shoko.calarts.edu/musicdsphttp://ceait.calarts.edu/mailman/listinfo/music-dsp
== end ==