On Sat, 2007-03-31 at 13:17 +0200, Dennis Schulmeister wrote:
Thanks Paul and Pau for pointing me to GStreamer. I took a glimpse on
some tutorials yesterday and my impression is that GStreamer is easier
to handle than I thought.
whatever you do, don't forget to start using the playbin element
On Fri, 2007-03-30 at 16:28 +0200, Dennis Schulmeister wrote:
Hi,
My question is if there's any high-level sound API for python. I don't
want to process audio. Instead I'm looking for an API (or a combination
thereof) which could be used to implement a small media player feeding
several
of music audio files. It contains advanced waveform and
spectrogram viewers, as well as editors for many sorts of audio
annotations.
What does annotations mean in this context?
--
Paul Winkler
http://www.slinkp.com
On Thu, 2007-03-15 at 13:12 +0100, Malte Steiner wrote:
I wonder what are the alternatives, if there are any I would jump
instantly. Windows Vista in which nothing works and wastes CPU cycles to
spy and torture you? OSX which comes with a hefty price, limits your
choice (and money) and
for 'Device drivers', among countless other
resources.
Paul
[EMAIL PROTECTED]
If I were a commercial HW developer needing to develop a driver for the
first time, this would certainly be an issue.
Greets,
Pieter
On 3/14/07, Gordon JC Pearce [EMAIL PROTECTED] wrote:
After a few days of careful consideration, I've decided that I no longer
want to be involved in developing software for Linux. It's been a
difficult decision to make, having used Linux as my main desktop OS for
around 10 years now, but I
On Wed, 2007-03-14 at 08:56 -0400, Paul Coccoli wrote:
Besides, what you want is probably impossible. You can't have
pre-comiled, binary-only drivers *and* a custom kernel.
in theory, you certainly can. but the kernel development team, and linus
in particular, are not interested
On Wed, 2007-03-14 at 10:21 -0400, Lee Revell wrote:
Why should Linux sacrifice stability just so vendors can keep their
hardware interfaces secret?
although i broadly agree with lee on most things, i think that this way
of approaching this issue is unnecessarily confrontational. just flip it
.
Let me know if you need it.
--
Paul Winkler
http://www.slinkp.com
On Mon, 2007-02-19 at 14:18 +0100, Stefano D'Angelo wrote:
How often are more than one plugin with the same control inputs used in
paralel? I was rather thinking of colapsing (or swapping) plugins in
series. They'd have to be linear and time invariant, of course.
Or maybe plugins could
On Mon, 2007-02-19 at 18:10 +0100, Stefano D'Angelo wrote:
nope. thats not a linear arrangement of the two mono plugins, but a
parallel arrangement. the signal going to each instance of the mono
plugin is different.
I'm obscure even in Italian, I can just imagine how it can sound like
On Mon, 2007-02-19 at 13:33 -0800, Jonathan Ryshpan wrote:
On Mon, 2007-02-19 at 13:18 -0800, vreuzon wrote:
Jonathan Ryshpan a écrit :
The above recording session was done while jack was Stopped. Would
jack work better if it were Rolling?
This play button refers to jack transport
On Mon, 2007-02-19 at 19:49 -0300, Camilo Polyméris wrote:
Julien Claassen wrote:
Hi!
I'm sorry to ask that here, but it seems I can't get an anser anywhere
else.
Does the libstdc++ support UTF-8 strings? Or is there some simple example
code snippet somewhere to derive/modify
using to
implement a lot of pypy?
http://codespeak.net/pypy/dist/pypy/doc/coding-guide.html#restricted-python
--
Paul Winkler
http://www.slinkp.com
for lots of jobs. This isn't one of
them.
--
Paul Winkler
http://www.slinkp.com
On Wed, 2007-01-31 at 21:35 +0100, David Olofson wrote:
On Wednesday 31 January 2007 21:02, Michael Ost wrote:
[...]
We have a 32 sample setting (.7 msecs) in Receptor which I have yet
to see in a Windows driver. And it actually works with some plugins,
even a large sampler like Synthogy
?
/relurk
--
Paul Winkler
http://www.slinkp.com
On Sat, 2007-01-27 at 16:05 +1100, Fraser wrote:
Hi All,
I've been converting my old VST plugins over to LADSPA and have come
across something in the api which I really miss - the inability separate
the algorithmic to the displayed value of a parameter.
I'm finding this inability is leading
On Wed, 2007-01-24 at 16:06 +0100, Jay Vaughan wrote:
At 20:08 +0100 22/1/07, Stefano D'Angelo wrote:
What I'd like to work on is a sound processing architecture (LADSPA,
VST, DSSI, etc.) wrapper, which hides the details of a particular
implementation to audio program developers.
Nice idea.
http://www.freshpatents.com/Low-latency-real-time-audio-streaming-dt20060406ptan20060074637.php?type=description
in which Microsoft patents designs partially implemented by OSS 10 years
ago and fully implemented by ALSA 5 years ago. Wrapping this up in
windows API nonsense obscures the basic fact
On Tue, 2007-01-23 at 14:31 +, John Rigg wrote:
On Tue, Jan 23, 2007 at 08:53:13AM +1100, Erik de Castro Lopo wrote:
Hi all,
SecretRabbitCode was recently included in a test of a number of
commercially available sample rate converters and while it wasn't
the best, it certainly
On Tue, 2007-01-23 at 08:53 +1100, Erik de Castro Lopo wrote:
Hi all,
SecretRabbitCode was recently included in a test of a number of
commercially available sample rate converters and while it wasn't
the best, it certainly didn't disgrace itself either.
congrats Erik. as you said, not the
On Fri, 2007-01-19 at 11:54 +0100, Marc-Olivier Barre wrote:
Ok, Apparently forum addicts have decided to hijack this thread
though, the original question was : do you want LA* _lists_ (LIST is
the keyword here, can't stress it enough) to be moved to
linuxaudio.org.
I have noting
On Tue, 2007-01-16 at 07:30 -0500, Dave Phillips wrote:
Damon Chaplin wrote:
What are the recommended books to read for people new to audio
development? (Covering things like synthesis techniques, effects
processing and basic acoustics stuff.)
On the bottom of the Documentation section of
ALSA, Jack, LADSPA/LV2, DSSI LASH.
Damon
--
Paul Winkler
http://www.slinkp.com
On Mon, 2007-01-15 at 20:00 +0100, Richard Spindler wrote:
2007/1/15, Dave Phillips [EMAIL PROTECTED]:
This link may have been posted here before now, but it bears repetition :
http://www.cs.auckland.ac.nz/~pgut001/pubs/vista_cost.txt
Nasty stuff planned in Redmond.
It seems as
On Mon, 2007-01-08 at 23:23 +0100, Milan Mimica wrote:
Hello!
We're designing a new sound subsystem for allegro game programming
library, and we would like to take advantages of multiple hardware voice
capabilities.
very few audio interfaces have this feature anymore. most h/w makers
seem
On Wed, 2007-01-10 at 18:02 +0200, Jussi Laako wrote:
Paul Davis wrote:
in general, you should forget about the h/w capabilities of an audio
interface. for every user that has a device with some interesting
qualities, there will be 10 who do not.
welcome to winmodem for audio
On Wed, 2007-01-03 at 16:28 -0500, kind king knight wrote:
I need a framework of a sequenced sample player. This is becouse I want
to start my own project, and don't want to invent everything from
scratch. Ofcourse there is lot of this kind opensource applications, but
I need the simplest.
-
On Wed, 2006-12-20 at 15:43 -0800, Anthony Green wrote:
I understand that LADSPA and friends specifically exclude any
functionality around how to find and load plugins, but it seems that a
lot can be gained by introducing some standards in this area.
As a package of audio apps/plugins for
On Thu, 2006-12-21 at 02:02 +0100, Leonard Ritter wrote:
Hi Anthony,
I guess most of us use the sample enumeration c code included with the
LADSPA sources as starting point. This code expects a LADSPA_PATH
variable to be set. As a fallback, I suppose most programmers
added
On Sun, 2006-12-17 at 19:43 -0800, oscar si wrote:
Hello:
Could anyone please tell me if there are sample driver codes for PCI
based sound card that are not using either alsa or oss?
why would you want such a thing?
to be
here.
A toast to the host who brought the most!
For my part, I apologize that I several times proposed an interactive
replacement to the static site and never delivered anything. Sometimes
I miss being underemployed :-)
--
Paul Winkler
http://www.slinkp.com
On Wed, 2006-11-22 at 15:40 +0100, conrad berhörster wrote:
Hello all,
is there a way to restart jack out of an application, or is this the reason,
why ardour need a running jackd.
i want to write an application with a reinit app button and need a way to
restart jackd, if it has been
On 11/22/06, Christian Schoenebeck [EMAIL PROTECTED] wrote:
Am Mittwoch, 22. November 2006 10:27 schrieb Thorsten Wilms:
i'm looking for a sampler instrument file format similar to .nki, .sf2
or akai instruments. is there an open standard existing already, perhaps
even accompanied by some
On Thu, 2006-11-16 at 18:50 +0100, Takashi Iwai wrote:
Hi,
is anyone interested in participating in the third Desktop Architects
Meething (DAM) on December 7-8 in Beaverton near Portland, Oregon?
I'm not able to attend it (I'll be in vacation in Japan exactly at
that time), and hope someone
On Sun, 2006-11-12 at 22:02 +0700, Patrick Shirkey wrote:
They might pay Stephane Letz some money to port JACK though ;-)
Stephane has already ported JACK to windows.
On Wed, 2006-11-08 at 23:19 +0100, Elthariel wrote:
Thk you for testing it.
You build problem looks quite strange,
is there any error message about other EV_XXX identifier missing.
EV_SW is a macro defined in linux/input.h
can you send me your version of the kernel and your
On Mon, Nov 06, 2006 at 01:39:43PM -0500, Lee Revell wrote:
Real linux drivers reside in the mainline kernel. Out of tree stuff is
is irrelevant.
I don't think that's a fair blanket statement given that drivers often
begin life outside the mainline kernel tree. ALSA, for example.
--
Paul
On Mon, 2006-11-06 at 16:38 -0500, Paul Winkler wrote:
On Mon, Nov 06, 2006 at 01:39:43PM -0500, Lee Revell wrote:
Real linux drivers reside in the mainline kernel. Out of tree stuff is
is irrelevant.
I don't think that's a fair blanket statement given that drivers often
begin life
(no mids, no highs) there was some easily
audible distortion that I couldn't get rid of.
When turning off the solo switch, it either went away or was masked by
the mids and highs.
Not sure which version of JAMIN, this was early 2006 and
I haven't used it lately.
--
Paul Winkler
http
On 11/3/06, Orlarey Yann [EMAIL PROTECTED] wrote:
Paul Coccoli a écrit :
On 11/2/06, Orlarey Yann [EMAIL PROTECTED] wrote:
Paul Coccoli a écrit :
The alsa-gtk build does not run on my FC4 box, because it requires
libpangocairo-1.0.so.0.
alsa-gtk applications are build against gtk+-2.0
delurk
regarding portability...
portaudio?
/delurk
--
Paul Winkler
http://www.slinkp.com
On 11/3/06, Orlarey Yann [EMAIL PROTECTED] wrote:
Paul Coccoli a écrit :
Here's pkg-config on my FC4:
[EMAIL PROTECTED] ~]$ pkg-config --libs gtk+-2.0
-lgtk-x11-2.0 -lgdk-x11-2.0 -latk-1.0 -lgdk_pixbuf-2.0 -lm
-lpangoxft-1.0 -lpangox-1.0 -lpango-1.0 -lgobject-2.0 -lgmodule-2.0
-ldl -lglib
On 11/3/06, Orlarey Yann [EMAIL PROTECTED] wrote:
Not to drag things out, but the problem isn't really with FC4. Anyone
who doesn't have libpangocairo installed will not be able to run those
binaries. Period. You said the compiler server isn't linking the
binaries against that lib, but I
On 11/2/06, Orlarey Yann [EMAIL PROTECTED] wrote:
Dear All,
A completely rewritten Faust website is available at
http://faudiostream.sourceforge.net or at http://faust.grame.fr . Its
main feature is the possibility to use the Faust compiler online, via
the web pages, without having to install
On Thu, 2006-11-02 at 20:58 +0100, Jens M Andreasen wrote:
On Thu, 2006-11-02 at 10:33 +0100, lemmel wrote:
Lots of people have been wondering, but this is the meat I think:
and even randomly truncated,
The difference in behaviour /might/ arise from differences in philosophy
of
On 11/2/06, Orlarey Yann [EMAIL PROTECTED] wrote:
Paul Coccoli a écrit :
The alsa-gtk build does not run on my FC4 box, because it requires
libpangocairo-1.0.so.0.
alsa-gtk applications are build against gtk+-2.0 which requires
pangocairo-1.0. Have you gtk+-2.0 installed ?
Yes, I have gtk
On Wed, 2006-11-01 at 22:32 +0100, lemmel wrote:
Well, all the files, that I will play, will have the same charactistics, do I
really need to bother with an hi-level API [2] ?
[1] I noticed that a lot of applications still use OSS, and I thought it was
because the migration to ALSA was
On Mon, 2006-10-30 at 18:52 +0100, Fons Adriaensen wrote:
- hardware presents itself as 2 * 96 kHz
- user wants to see a device with 4 * 48 kHz.
interestingly, ADAT devices do the opposite to get to SR's above 48kHZ:
- hardware runs as N * 48 kHz channels
- data is
On Fri, 2006-10-20 at 22:44 +0200, Tim Goetze wrote:
[Fons Adriaensen]
Input the vertical video sync signal via the audio card and analyse
its timing in terms of audio samples (e.g. using a DLL). This will
enable you to predict where the next sync will be in the audio input.
Back in the
On Tue, 2006-10-17 at 11:56 +0200, Fons Adriaensen wrote:
'THE SAMPLES ARE NOT THE SIGNAL'. The real peak level of a
signal when converted to the analog domain can be several
dB above that of the highest sample.
indeed. there are people who are coming to believe that this error is
responsible
, because
the compressor is effectively trying to increase the gain at the end of
every half-cycle. This is considered a case of giving the engineer
enough rope to hang himself, and hoping that he uses it responsibly. See
http://www.fmraudio.com/FAQ.htm#question4
for more.
--
Paul Winkler
http
microseconds at 44.1 kHz? I don't think there are any
analog compressors that react anywhere near that fast. Don't worry about
it :-)
--
Paul Winkler
http://www.slinkp.com
On Wed, 2006-09-06 at 22:11 +0100, John Rigg wrote:
Hmm. The manufacturer's web page describes it as a 32 channel interface
but I could only count 16 :(
On Wed, 2006-09-06 at 22:11 +0100, John Rigg wrote:
Hmm. The manufacturer's web page describes it as a 32 channel interface
but I could only count 16 :(
many companies count input + output separately. RME, for example.
several years ago, i wrote an system to control an ICube MIDI Sensor
interface, described here:
http://equalarea.com/paul/icube
unfortunately, the actual software has gone missing, even google cannot
find it.
if anyone has a copy of the software, i'd like to get a copy of it.
On Fri, 2006-08-25 at 17:02 +, carmen wrote:
i guess everyone has to pay the rent somehow...but do a indeed/simplyhired
search for linux audio, or similar. and check out the names of the top 10
entriesSony, Avid, Qualcomm. id rather work at starbucks than give them
more intellectual
On Sat, 2006-08-19 at 05:16 -0700, Stephen Cameron wrote:
On Fri, 2006-08-18 at 20:39 -0400, Stephen Sinclair wrote:
Audio doesn't use setitimer()-driven sleeping. It's interrupt-driven,
not timer-driven.
Yes, the driver is interrupt driven, but the driver interrupt handler
is only
On 8/16/06, carmen [EMAIL PROTECTED] wrote:
i assume this Xscale is more or les x86.. definitely be interested int hese if
theyre priced reasonably.
Nope: ARM. See http://en.wikipedia.org/wiki/Intel_XScale
On Sun, 13 Aug 2006 15:07:04 +0200
Dominique Michel [EMAIL PROTECTED] wrote:
From http://www.jacklab.net/phpBB2/viewtopic.php?t=103 :
Model 1: Want to start a synth from time to time
SUSE Standard kernel with PAM from Rui (also available in the jacklab
repository)
Model 2 Want to work
On Tue, 15 Aug 2006 10:02:25 +0200
Florian Paul Schmidt [EMAIL PROTECTED] wrote:
Given that you have setup rlimtis and rt lsm in equivalent ways, there
must be no measurable difference in achievable latencies. If you do
measure differences, some other factor of your setup differs, too.
Let me
The db inside jack are dbFS with a maximum possible signal of 0 db.
Now, both jacqeq and jackmix give you a maximum conrol level at +6dB.
It mean at +6dB in those EQ is equal to 0dbFS in jack.
there are no dB units inside of JACK. some JACK applications use dBFS,
that much is true. however, it
On Sun, 13 Aug 2006 12:46:35 +0200
Dominique Michel [EMAIL PROTECTED] wrote:
A rt kernel will be of almost no use if you don't fix the priorities.
You can use PAM-rlimits or the realtime-lsm module for that, and you
have to be in the audio group.
The audio group will have a higher priority
:
- support for realtime stretching (as a player)
- ogg vorbis support for input
- improved the stretching algorithm and now it can
do unlimited stretching
- other
Paul
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam
Someone called illiac wrote:
Linux as it is ordinarily distributed is not a small-footprint real-
time operating system. You will notice that your cell phone does not
run Linux. There is a reason for that.
i am sure nokia will be interested in your reason, since they don't seem
to have
On Wed, 26 Jul 2006 15:07:13 +1000
Erik de Castro Lopo [EMAIL PROTECTED] wrote:
My own rather feeble first attempt is here:
http://www.mega-nerd.com/tmp/sndfile.hh
but I am not a fan nor a great user of C++. The wrapper should
really be written by someone with a love for the
On Wed, 26 Jul 2006 11:26:00 +0100
Chris Cannam [EMAIL PROTECTED] wrote:
vote++, i never cared for the more java style methodName convention.
I think if your class is named LikeThis, then your method should be
named likeThat (Java-style). If your method is named like_this, then
your
On 7/26/06, Erik de Castro Lopo [EMAIL PROTECTED] wrote:
Taybin Rutkin wrote:
I prefer the unix-y open_read(). I don't think method names should
ever start with a capital, unless it's the ctor or dtor.
I'm actually tending towards openRead().
I vote for Java (SmallTalk?) style too.
I
On 7/26/06, Steve Harris [EMAIL PROTECTED] wrote:
On Wed, Jul 26, 2006 at 03:41:24PM +0200, Richard Spindler wrote:
2006/7/26, Thorsten Wilms [EMAIL PROTECTED]:
You should not depend on particular plugins, if the app could work
without just fine. Hasn't Debian a 'Recommends' thing going for
On 7/20/06, Loki Davison [EMAIL PROTECTED] wrote:
On 7/20/06, Erik de Castro Lopo [EMAIL PROTECTED] wrote:
Loki Davison wrote:
There are quite a few c++ 'not fans' on LAD. C and python all the way ;)
I used to be a Python fan but for anything larger than a couple
of hundred lines I now
On Fri, 2006-07-14 at 13:27 +0200, conrad berhörster wrote:
Hello maarten and dmitry and the rest,
thanks for the quick answers.
Faster means, that the workerthread is called more often than the jackthread
no, the workerthread should *not* be called more often than the
jackthread. either
On Thu, 2006-07-13 at 21:11 +0200, Dirk Jagdmann wrote:
Libsndsfile is plain C, but will do what you want without any fuss.
You could write a WAV specific C++ wrapper on top of this in a few minutes.
libsndfile is superb, but sometimes you don't want to link against
external libraries in
On Fri, 2006-07-14 at 00:56 +0400, Dmitry Baikov wrote:
If you need to stream a file, mmap'ed variant will eat memory up to
file size. Given large enough file, it will eat all you memory and
then will begin to page out unused portions of the file.
Of course, details on when and where will vary
On Fri, 2006-07-14 at 06:48 +1000, Erik de Castro Lopo wrote:
but I am not a fan nor a great user of C++. The wrapper should
really be written by someone with a love for the language.
LOL! that's pretty great. not a fan translates in real world terms
into one of LAD's most persistent critics of
On 7/11/06, Stefan Westerfeld [EMAIL PROTECTED] wrote:
Since we've been comparing different methods here, I thought I might as
well write a benchmark, to look at the performance, too. I wrote a
little test which repeatedly switches between two threads, which wakeup
eachother using a pipe, cond
On Tue, 2006-07-11 at 17:06 -0400, Dave Robillard wrote:
Semaphores seem about perfect for this to me.. am I missing something?
Why doesn't anyone ever recommend them?
i think mostly because in 2000-2001, they were very slow.
On Sun, 09 Jul 2006 14:03:50 +0200
Jens M Andreasen [EMAIL PROTECTED] wrote:
On Sun, 2006-07-09 at 12:06 +0700, Huu Phuoc wrote:
Hi everybody!
I am a newbie to alsa programming.
I am trying to follow the article which locates at
http://www.suse.de/~mana/alsa090_howto.html to develop a
On Sat, 2006-07-08 at 13:34 +0100, James Courtier-Dutton wrote:
Hi,
Is there a standard way of converting a 24bit sample to 16bit?
I ask because I think that in different scenarios, one would want a
different result.
1) scale a 24bit value to a 16bit by simple multiplication by a fraction.
On 6/26/06, Jack O'Quin [EMAIL PROTECTED] wrote:
OK, I see it now. All those uninitialized write complaints are due to
the fact that jack_request_t is a union. Most requests don't need (or
want) to fill in all the bytes, just the ones that matter for that
RequestType.
There are jack_request_t
On Tue, 2006-07-04 at 07:48 +0200, Andreas Kuckartz wrote:
I have a simple question:
Which companies are (or have been) distributing LinuxSampler as part of
a package also including hardware and/or proprietary software?
as noted liontracs do, and that means that is incumbent upon me to
On Mon, 3 Jul 2006 12:33:11 +0200
Alfons Adriaensen [EMAIL PROTECTED] wrote:
On Fri, Jun 30, 2006 at 11:53:01AM +0200, Tom Szilagyi wrote:
Aqualung: Music Player for GNU/Linux
Release 0.9beta5
This looked like the player I've been wanting for some
On Mon, 2006-07-03 at 02:26 +0700, Patrick Shirkey wrote:
If they really want to get people to give money then they should just
make it so that you have to pay or contribute code/time for a while to
get access to the newest downloads from their site. Keep the stable
version far enough
On Tue, 2006-07-04 at 10:22 +1000, Ryan Heise wrote:
On Mon, Jul 03, 2006 at 05:55:11PM -0400, Paul Davis wrote:
both guesses are wrong. i think it will be precise enough to say that a
company expressed what appeared to be a serious interest in leveraging
the existence of LS for its own
On Fri, 30 Jun 2006 11:53:01 +0200
Tom Szilagyi [EMAIL PROTECTED] wrote:
Aqualung: Music Player for GNU/Linux
http://aqualung.sf.net
Release 0.9beta5
Damn you beat us to it. See these two mailing list posts:
On Thu, 2006-06-29 at 18:35 -0400, Forest Bond wrote:
I've been looking at fst, and was going to package it for Ubuntu.
you cannot legally package FST. please do not do this. its not likely
that steinberg will come after you, and neither torben nor I are likely
to either, but its a violation of
understand well, it could be a good idea to have more than
one moderators. preferably in different time zones...
greetings, Paul
On Sat, 2006-06-24 at 23:26 +0100, [EMAIL PROTECTED] wrote:
Hi peeps.
I've just been running an app through valgrind and I'm getting a few
of these:
==11955== Syscall param write(buf) points to uninitialised byte(s)
==11955==at 0x4D51BDB: (within /lib64/libpthread-2.4.so)
==11955==
On Tue, 2006-06-20 at 15:26 +0100, Steve Harris wrote:
On Tue, Jun 20, 2006 at 09:39:30 -0400, Dave Robillard wrote:
I can make the plugin validating host check the latency primitively (eg
run a single sample through the buffer) and fail if it isn't reported
correctly, so we're sure the
svn should once again have working (better than before the libsndfile
changes) support for tape tracks. what a total pain this has been, but i
think the end result is worth it - a standard library shared with other
apps. let me know if you have issues with it. it may not work with
existing
On Tue, 2006-06-20 at 00:57 +0200, Fons Adriaensen wrote:
It's not beyond the realms of the possible to describe the mathematical
relationship between the octave pitch unit and Hz, but it's probably
excessive.
A well-designed set of tags like the ones you show above would
probably solve
On Thu, 2006-06-15 at 16:32 +0900, David Cournapeau wrote:
I am in no way as experienced as most people on this list for audio
programming, but I don't see why C/C++ should be the only way to write
software to handle audio stream, neither do I see why GC would be the
only useful feature. For
On Thu, Jun 15, 2006 at 12:02:34PM +0200, Dominique Michel wrote:
On Wed, Jun 14, 2006 at 10:48:54AM -0400, Paul Winkler wrote:
Pyrex is good for making faster python libraries, which is a great
thing, but it won't help with the problem that you really
don't want to be running a python
action happening somewhere
that I'm not aware of? I was always disappointed that there
didn't seem to be a lively community around saol.
--
Paul Winkler
http://www.slinkp.com
On Thu, 2006-06-15 at 10:49 -0700, Michael Ost wrote:
As we looked over the Jack docs, it seems like a natural for supporting
this kind of architecture. We would break out our VST support into a
separate app and connect them to our Host app via Jack. This seems to be
how FST is implemented and
libraries, which is a great
thing, but it won't help with the problem that you really
don't want to be running a python interpreter inside a realtime
dsp system.
--
Paul Winkler
http://www.slinkp.com
On Tue, 13 Jun 2006 18:11:05 +0200
Lars Luthman [EMAIL PROTECTED] wrote:
You should maybe add that this will only work when jack is already
started or JACK_START_SERVER is set in the environment.
Hm, true. I never even considered that option. Is anyone still starting
jackd manually?
Yah,
On Mon, 12 Jun 2006 13:56:09 +0200
Lars Luthman [EMAIL PROTECTED] wrote:
To use the applet, simply run the program. If you have a
standards-compliant
system tray on your desktop a small LASH icon (a cardboard box with a
soundwave on it) should appear there. It is probably insensitive (greyed
On Wed, 2006-06-07 at 11:12 +0300, Jussi Laako wrote:
Paul Davis wrote:
writing to a pipe is not 100% RT safe, but if the pipe is created in a
shm filesystem, its as close to it as you will get without ...
Nowadays, there's also available a very good interface from POSIX RT
extensions
On Mon, 2006-06-05 at 01:21 +0200, Stefan Westerfeld wrote:
Hi!
I am trying to notify a high priority (nice -20 or nice -19) thread from
a realtime thread (from a jack callback to be precise). Of course I want
the realtime thread to not block, but I want the high priority thread to
react
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