Hacked some test code here and discovered something "interesting" with
lo_server_add_method() and method handling. If I try to do it like in
the examples and add the default/debug ("match all") method first, it
gets called for every incoming message, before the real handler is
called. That is, I see an error message - and then the correct method
handler is invoked anyway.
Looking quickly at the code, I'd actually expect this behavior, as
lo_server_add_method() does indeed add methods at the end of the
list.
Trying the examples again, I realize that they demonstrate this
behavior as well, so maybe it's not just my code doing something
funny. :-)
This is with liblo 0.22 on Gentoo/AMD64.
//David Olofson - Programmer, Composer, Open Source Advocate
.------- http://olofson.net - Games, SDL examples -------.
| http://zeespace.net - 2.5D rendering engine |
| http://audiality.org - Music/audio engine |
| http://eel.olofson.net - Real time scripting |
'-- http://www.reologica.se - Rheology instrumentation --'
Hello everybody !
I've just added a resampling function to my code thanks to the
excellent work of Erik de Castro Lopo (thanks a lot !). Combined with
libsndfile (thanks again) it is really easy to load any sound file I
want. But I'd like to make sure I'm using it correctly.
I process my input data in one pass using src_simple() and I have to
compute the length of the output data buffer beforehand. So I did
somehting like this :
out_len = (long int) ceil((double) in_len * ratio);
It seems that my output buffer is always one frame too big (I checked
this by reading the output_frames_gen field of the SRC_DATA structure
after the processing is done).
Is it safe to assume that using floor() instead of ceil() will not lead
to a too short output buffer in some cases ?
I can live with wasting a malloced float but I'd like to be sure it
cannot be done in a prettier way.
Thanks.
--
David
Announcing the 20060122 release of WhySynth, a DSSI softsynth
plugin.
New since the last major release:
* A new oscillator mode, based on Nasca O. Paul's gorgeous
PADsynth algorithm.
* A new filter mode, essentially the low-pass filter from amSynth.
* A new dual delay effect.
* Improved and extended wavetables.
* More patches.
* Lots of cleanups and bug fixes, including fixes for more stable
operation especially under Rosegarden, and for compilation on
Mac OS X 10.4 'Tiger'.
Find WhySynth here:
http://home.jps.net/~musound/whysynth.html
More information on the DSSI plugin standard, available hosts
and plugins can be found here:
http://dssi.sourceforge.net/
WhySynth is written and copyright (c) 2006 by Sean Bolton,
under the GNU General Public License, version 2.
Hi all,
is there a recommended way to write / read additional chunks in
WAV files, using libsndfile (assuming it's possible at all - I
didn't find any hints to this in the docs) ?
What I need in particular is some way to calibrate the time
axis - i.e. to say frame #N corresponds to t = 0, and some
other similar info, mostly sample indices.
TIA,
--
FA
Hi all,
I am new to this list, living in Switzerland and working mainly with
electronic. I have done it was some time ago a monophonic realtime note
recognition software on an embeded dsp56k system and I want to adapt it
to linux.
It will take some time to me to do that because I know almost nothing
about c and c++ programming, as about the alsa and jack libraries. But I
know my algorytm, and it is working just fine on my dsp.
The latency is very low, 1+(~1/4) period of the sound, the (~1/4) term
depend of the harmonic content of the sound. I believe
at it is worth to do a jack-alsa software with it.
A gui will be needed, giving the possibility to change, save and recall
some parameters of the recognition loop on an per instrument basis.
Can you recommand me a good, and if possible simple to use and fast,
MIDI library? I will use only basic functionalities as the possibility
to send MIDI notes to the sound server.
Ciao,
Dominique
A while ago the list moderator requested that all of us gmail users
"fix" our mail by setting the UTF-8 option. Which I did.
However, I'm now on another mailing list on Sourceforge where the
utf-8 thing causes junk to be added after my signature. This is some
sort of bug in sorceforge's base64 handleing.
I've been asked by why I had it set that way. So I was just wondering
what the issue with non- utf8 mail was on lad.
--
Richard A. Smith
hello... i just wanted to announce, that i have written a jack binding
for ninjam.
ninjam is a network jam session software.
www.ninjam.com
the ported consoleclient for ninjam is available here:
http://galan.sf.net/ninjam-with-jack.tar.bz2
and yes... there is a Makefile.. find -name Makefile it...
i just added the jack support...
more io ports will follow...
also dont use with too low period sizes, it does not behave so well..
--
torben Hohn
http://galan.sourceforge.net -- The graphical Audio language
Hello,
maybe the c++ gurus out there have an idea. currently i think it is
unpossible.
i try to use a template a a class member variable, like that:
asume a template matrix class
template < int cols, int rows>
class matrix {
// implementation
}
everything works fine when i'm doing something like
matrix<<3,4> mymat;
matrix <5,5> quad;
but now i try to do it a little more generic i try to build a class, which
have a member of this matrix like
class MyClass
{
MyClass();
MyClass(n,n);
private:
matrix<n,n>
};
this means, at runtime, i want to set the size of the matrix. is this
possible? this are divers concepts (templates and runtime) , aren't they?
thanks c~
A minor update of hexter, the Yamaha DX7 modeling DSSI plugin,
is now available at:
http://sourceforge.net/project/showfiles.php?
group_id=104230&package_id=134428
Changes include:
* The coarse frequency of each operator now can be controlled in
real time via MIDI control changes.
* Fixes for RPM, gcc 2.9x and 4.x, and Mac OS X 10.4 'Tiger'.
More information about hexter and DSSI can be found at:
http://dssi.sourceforge.net/hexter.html
hexter is written by Sean Bolton, and copyright (c)2006 under
the GNU General Public License, version 2 or later.
just to let you guys know .. following up on the GP2X synthesis
thread from earlier, the author of littleGPtracker for the GP32
(ARM-based handheld predecessor to the GP2X) has this to say about
synthesis on ARM ..
j.
>Delivered-To: 29-jayv(a)synth.net
>From: "M-.-n" <nostromo(a)arkaos.net>
>To: <gpx-dev(a)ampfea.org>
>Subject: RE: [gpx-dev] hardware interfaces
>Date: Thu, 12 Jan 2006 09:20:05 +0100
>X-Priority: 3 (Normal)
>Importance: Normal
>X-BeenThere: gpx-dev(a)ampfea.org
>X-Mailman-Version: 2.1.5
>Reply-To: gpx-dev(a)ampfea.org
>List-Id: gpx-dev.ampfea.org
>List-Unsubscribe: <http://www.ampfea.org/mailman/listinfo/gpx-dev>,
> <mailto:gpx-dev-request@ampfea.org?subject=unsubscribe>
>List-Archive: <http://www.ampfea.org/pipermail/gpx-dev>
>List-Post: <mailto:gpx-dev@ampfea.org>
>List-Help: <mailto:gpx-dev-request@ampfea.org?subject=help>
>List-Subscribe: <http://www.ampfea.org/mailman/listinfo/gpx-dev>,
> <mailto:gpx-dev-request@ampfea.org?subject=subscribe>
>Sender: gpx-dev-bounces(a)ampfea.org
>
>
>
>On littleGPtracker (still GP32; running @133Mhz), I do 8 monophonic stereo
>voices (sampled based) using about 50% of the CPU. This includes possibility
>of arpegiation, pitch shifting, legato & volume envelope control but no
>filters yet. Sure the sample engine is pretty crude at the moment but u will
>have PLENTY enough power in the 2x to do nifty things.
>
>-----Original Message-----
>
>MHz does not say much, I think, but it should be plenty to do a lot of
>fun with :)
>
>--
>
>Joost Schuttelaar
>
>_______________________________________________
>gpx-dev mailing list
>gpx-dev(a)ampfea.org
>http://www.ampfea.org/mailman/listinfo/gpx-dev
>
>
>
>
>_______________________________________________
>gpx-dev mailing list
>gpx-dev(a)ampfea.org
>http://www.ampfea.org/mailman/listinfo/gpx-dev
--
;
Jay Vaughan