The latest release of caps, a collection of LADSPA plugins, contains
various fixes necessitated by upgrades to gcc and libc.
Other than that, only slight adjustments and fixes have been applied
to various plugins, with the notable exception of major improvements
to the Compress unit.
http://qui
[Fokke de Jong]
>I’m processing 32 sample-blocks at 48KHz but roughly every 0,6
>seconds I get a large spike in cpu usage. This cannot possibly be
>explained by my algorithm, because the load should be pretty stable.
>
>I am measuring cpu load by getting the time with
>clock_gettime(CLOCK_MONOTONI
[David Robillard]
>I believe #1 is the intent. The hint serves to distinguish plugins that
>have some dependency on real time, from those that do not (only
>processing their input). The latter can be used in a more functional
>way, being called to process a block whenever, where the actual time r
[Guy Sherman]
> Would the approach to use a sample-rate converter to essentially interpolate
> samples, then do the processing, and then sample back down?
The principle is indeed the same, and you could use a converter
library for this purpose. However, those converters are designed to
work over
[Guy Sherman]
>And the code is at: https://github.com/guysherman/si-plugins
For high-bandwidth input or high-gain clipping, you'll need to run the
nonlinear operator at substantially elevated sample rates unless you
want synthesised harmonic content to alias audibly. I'm fine with 4x
oversampling
http://quitte.de/dsp/caps.html
The latest release of the caps audio plugin suite contains a number of
bug fixes and improvements as well as changes intended to make it
more suitable for low-power systems.
http://quitte.de/dsp/caps.html#CHANGES
Most notably, support for a variety of little-used f
Good morning lads,
among the people still using LADSPA, either directly or wrapped into
more fashionable plugin APIs, is anyone actually dependent on
run_adding()?
(Asking, of course, because I intend to drop support for it.)
Thank you,
Tim
___
Linux-
[William Light]
>I'm avoiding blocking, of course, but I'm also worried about the
>potential scheduling implications of jumping into kernel-mode and back,
>and also the potential non-determinism of system call execution time.
>Are these things I should actually worry about?
I've been using pipes f
[W.Boeke]
> Compared to the available technical possibilities of the past, software
> designers nowadays have a much easier life. A computer and a MIDI keyboard is
> all you need, you can try all kinds of sound creation, so why should you stick
> trying to reproduce the sounds of yore?
I definitel
[hermann meyer]
> However, this list here is the wrong place to discus that any further,
> I just hope that some people here get interested in such a approve
> (Tim you are more then welcome to join us),
> if, please join us at
> guitarix-develo...@lists.sourceforge.net
Thank you kindly Hermann, I
[Paul Davis]
>the OP(s) should by looking at both soundtouch and librubberband. any
>others?
A few years ago, I've published one which employs Dolson's original
phase vocoder (Richard Dobson's streaming C++ adaptation to be
precise) but I reckon I'm the only one to actually use it. It's only
a co
[Paul Davis]
>On Sat, Mar 1, 2014 at 7:33 AM, Fons Adriaensen wrote:
>[...]
>in addition, fftw3 has already had years of optimization applied.
Two algorithms of wide and likely lasting use that might be able to
benefit strongly from vectorisation come to my mind: sample-rate
conversion and time-
CAPS 0.9.19
===
http://quitte.de/dsp/caps.html
The latest release of this collection of LADSPA plugins extends the
dynamics modulation capabilities of the 'virtual guitar amplifier'
AmpVTS and fixes a nasty bug in the Noisegate circuit that had been
causing spurious gain fluctuations in c
CAPS 0.9.17
===
http://quitte.de/dsp/caps.html
The latest release of CAPS, a collection of LADSPA plugins, contains
two important bugfixes and minor sonic improvements.
http://quitte.de/dsp/caps.html#CHANGES
James Morris insisted that something was wrong with the PlateX2 stereo
reverb,
[Burkhard Ritter]
>I've considered different approaches (separate data for each thread,
>various versions of shared data), but I believe all of them make at
>least two assumptions: That reading and writing byte-sized data (e.g.
>a bool flag) and that updating pointers is atomic. I got a bit
>confus
The new version of CAPS comes -- among other, minor changes -- with
control input smoothening for Eq4p, establishing gentle manners in the
face of even the most drastic parameter changes:
http://quitte.de/dsp/caps.html#Download
Due to an unrelated change, the plugin will even run slightly quicker
[Fons Adriaensen]
>On Sun, Aug 25, 2013 at 12:21:52AM +0200, Tim Goetze wrote:
>
>> I'm planning to evade this problem by crossfading between two parallel
>> static filters. Some phase mismatch issues can probably be expected
>> when the parameter sweep covers a lar
[Fons Adriaensen]
>I've learned to mute monitoring while doing such tests...
I was suspecting you had :)
>> It adds to the evidence that implementing thorough control
>> smoothening is inevitable in the long run.
>
>That may or not solve the problem. You have 3 input parameters
>(P), and 5 biqua
[Fons Adriaensen]
>On Sat, Aug 24, 2013 at 03:14:14PM +0200, Tim Goetze wrote:
>> Eq4p, a four-way parametric equaliser
>> http://quitte.de/dsp/caps.html#Eq4p
[...]
>The F slider switching between its extremes when using the
>mouse wheel is of course a bug in A3. But it
CAPS Audio Plugin Suite 0.9.12
http://quitte.de/dsp/caps.html
New in this release:
Eq4p, a four-way parametric equaliser
http://quitte.de/dsp/caps.html#Eq4p
(Unlike Fons', this one uses parallel processing but lacks control
smoothening other than that provided by a continuous IIR filter
history.)
[Julien Claassen]
> I have been recording music for 11 years now, with Linux. [...]
> I'm not sure, if my luck is due to the commandline, it tends to eliminate
> a lot of problems, or if I'm just lucky with my Linux. :-)
I think the small applications with narrow focus you find used on the
comman
CAPS 0.9.10
===
http://quitte.de/dsp/caps.html
The CAPS Audio Plugin Suite, a LADSPA library comprising classic sound
effects, various signal generators and guitar tone processing, sees
another update containing some bug fixes, minor sound improvements in
various places and a new plugin,
Hi Aurélien,
>- What method of clipping is used will give a "personality" to the
>module: hard clipping, soft clipping, the method used for soft
>clipping, etc...right?
from my experience, there are two main factors determining the
character of waveshaping distortion: one is the hardness of the
c
[Paul Davis]
>On Sun, Jun 9, 2013 at 5:55 AM, Tim Goetze wrote:
>> Some MIDI devices do not employ 14-bit CCs or the NRPN mechanism.
>> Access' Virus for example maps all CCs as single 7-bit values, and a
>> forced 14-bit mode will break communication.
>>
>
>
[David Robillard]
>I really think that, in Jack (and LV2), it should just be mandated that
>these (and all other multi-) events be shipped as one, for the same
>reason that running status is forbidden: it's annoying and entirely
>pointless.
Some MIDI devices do not employ 14-bit CCs or the NRPN me
[Fons Adriaensen]
>Exactly the same with the form I proposed, w, a, b need to be computed
>just once, not for every gain change. In fact only w depends on the
>sample rate, a and b are fixed constants.
Ah yes, sorry, I see that now.
If that extra operation comes around to bite hard enough, I'l
[Fons Adriaensen]
>On Tue, Mar 19, 2013 at 03:02:19PM +0100, Tim Goetze wrote:
>> Surely you realise this version executes exactly as many additions and
>> multiplications per sample as a biquad?
>
>Yes. In this case it's possible to remove one multiplication
[Fons Adriaensen]
>On Tue, Mar 19, 2013 at 04:26:21AM +0100, Tim Goetze wrote:
>> A 2nd-order IIR filter is often called a "biquad"; at musicdsp, look
>> for that instead.
>
>Not really. A biquad is one way to implement a 2nd order IIR, and
>in many cases related
[Tim Goetze]
>[Harry van Haaren]
>>How is the "rise time" determined here?
>
>As a function of the filter's damping (zeta = 2*Q) and frequency:
>http://en.wikipedia.org/wiki/Rise_time
Sorry, zeta = 1 / (2*Q).
___
L
[Harry van Haaren]
>On Mon, Mar 18, 2013 at 11:45 PM, Fons Adriaensen wrote:
>> A critically damped
>> second order lowpass with a rise time of 30 ms or so will eliminate all
>> audible artefacts. It's very low on CPU and you only need to run it while
>> the gain is changing.
>
>Although I understa
[Tim Goetze]
>[Bart Brouns]
>>I have a similar problem as the one reported earlier:
>>In AmpVST, all amp-models sound the same to me except the BassMan.
>>Also: they all have a lot more gain/distortion then the BassMan.
>>This is with v 0.9.7.
>>
>>If you wa
[Bart Brouns]
>I have a similar problem as the one reported earlier:
>In AmpVST, all amp-models sound the same to me except the BassMan.
>Also: they all have a lot more gain/distortion then the BassMan.
>This is with v 0.9.7.
>
>If you want I can provide a small Ardour session to demonstrate the
>p
[Bruno Gola]
>http://github.com/portalmod/caps-lv2/
>
>We simply modified some basic files, so it's easy to keep in track
>with upstream changes. For more info about the port you can read the
>PORT section of the README.md file.
That seems to be pretty much what I had in mind; good to see it works
[Bruno Gola]
>Is there any special reason for you to set the Properties variable
>inside ChorusI::setup() method? The setup() will always call the
>autogen() method that will set Properties = HARD_RT anyway.
Perfectly OK to remove that line, it's only there because ChorusI got
backforwardported, a
[Gianfranco Ceccolini]
>One more thing. The new Cabinets have their gain limited to 0 dB.
>
>The old ones had 24 dB as the upper limit, which was very practical, since
>enabling the plugin would give me the tone improvement with a cost os some
>atenuation which would be compensated with some extra
Ciao Gianfranco,
>I think the AutoFilter has something wrong.
[...]
>According to the webpage doc, in default settings it should work as a
>AutoWah but it doesn't.
You're right, the last round of changes have reduced the range of
automatic filter modulation to the point of it being virtually
inau
This release of CAPS, likely to be the last for some time unless major
bugs surface, brings
* performance and sound improvements for AmpVTS,
* selection of peak or RMS measurement in Compress,
* 4x128 saturating mode for Compress,
* 2x,4x and 8x oversampling for AutoFilter, and
* more filter types
[Tim E. Real]
>OK makes sense to me. Thanks.
>My mistake, I "turned it down, man" and can now hear the differences
> with all controls.
>After that, even with gain at full I can now recognize the differences.
Heh, I'm the one boasting about "meaningful control interfaces" when
in reality the effec
[Tim E. Real]
>Applied patch. The model seems to still make very little difference.
>Hard to tell, but it seems like only model 7 sounds different than the rest.
>Also hard to tell but the brightness seems to make no difference.
>Should there be so little difference in these settings?
Situated rat
[Gianfranco Ceccolini]
>when running AmpVTS in JackRack I hear no difference when changing the amp
>models. when plugging the Tonestack plugin the difference is perfectly
>audible.
>
>Something worng with AmpVTS?
Yes, it's a bug, thanks for pointing it out. The attached patch fixes
the problem.
New in CAPS 0.9.4:
* selectable oversampling ratios for AmpVTS (2x,4x,8x)
* selectable sounds for Click (box, stick, beep; the second being
very close to the sound of the unit in 0.4.x)
* further smoothening of ChorusII modulation
* selectable oversampling ratios for Compress (2x,4x)
Also, a
CAPS 0.9.3's guitar amplification contains four new tonestacks,
distortion asymmetry dependent on output power and other sound
improvements. The modulation of the mono chorus has been smoothened,
and the Plate default parameters have been restored to be nearer the
more reverberant response of the
[Jörn Nettingsmeier]
>> http://quitte.de/dsp/caps.html
>
> that's excellent documentation! i'm looking forward to listen to the
> saturation
> functions in particular, looks like there is much to learn. thanks for
> creating
> and sharing this!
Thanks Jörn, glad you like the first look :)
It's
[R. Mattes]
>
>Builds fine but doesn't load:
>
> error opening shared object file '/usr/local/lib/ladspa/caps.so':
>/usr/local/lib/ladspa/caps.so: undefined symbol: _ZN3DSP10Polynomial5clip9Ef
>
>Template problem?
>
>This happens with compilations with all of
> g++, clang++ and clang++-libc++
Th
CAPS 0.9.1
==
The C* Audio Plugin Suite is a selection of popular effects, unique
filters and generators. For the digital guitarist, CAPS offers a
range of processors recreating the formation of tone in traditional
instrument amplification. Beyond sound quality, central design
considerat
[M Donalies]
>I have a few other questions that aren't directly related:
>The first has to do with manipulating midi ctlr events to simulate various
>musical techniques like legato and palm muting. What I've tried so far doesn't
>sound very good. I need legato in polyphonic mode, so just overlapp
[Julien Claassen]
> If I know, that my synth uses all those not recommended controls and has no
> other obvious 14bit controls, could another solution be, to allow these
> controls just as any other? Or would that again break something else?
> I mapped all the controls from my NordLead3 - and unl
[Paul Davis]
>On Mon, Dec 10, 2012 at 7:44 AM, Tim Goetze wrote:
>> To know what the manufacturers intended, I guess you'd have to ask
>> them, or purchase the official MIDI spec from midi.org (print copies
>> only!), hope it answers the question and hope that the man
[Julien Claassen]
> I've got a question about MIDI controller handling in software. I heard, that
> MIDI CCs 32-63 have special meaning, or are used as part of 14-bit controllers
> or something. Now I have a synth, which submits those. I'm sure, that other
> people might also have such devices and
[Bruno Gola]
>On Fri, Nov 30, 2012 at 9:31 AM, Tim Goetze wrote:
>> [Bruno Gola]
>>>So... http://github.com/portalmod/caps-lv2/
>>
>> After having had a quick glance, it seems the structural changes made
>> to the source tree will make reflecting upstream deve
[Bruno Gola]
>So... http://github.com/portalmod/caps-lv2/
After having had a quick glance, it seems the structural changes made
to the source tree will make reflecting upstream development a bit
difficult, or is my perception failing?
Cheers, Tim
___
Li
[Aurélien Leblond]
>Quick question: does the CAPS* plugins suite exist in LV2 format (I'm
>almost sure the answer is no, as I have looked everywhere, but you never
>know...).
>
>If not, could somebody who knows the code of these plugins let me know how
>hard it would be to port them to LV2? Are the
[Lucas Takejame]
>I was using CAPS AmpVTS and notice that some tone knobs didn't work well. I
>end up trying tonestack plugin itself and it seems that the problem (is
>this a bug or its the way it really works?) is there, the Bass filter
>apparently stops working about 0.5 to 1.0 , the Mid filter,
[Fons Adriaensen]
>Once synths became polyphonic they mostly turned from a instrument
>in its own right into something just used to imitate other instruments.
>
>I'm working on a SW modular synth that at least in its first version
>will be strictly monophonic, precisely to emphasize it's meant to b
[Harry van Haaren]
>Thanks all for the replies, I've certainly learnt a lot.
>
>On Thu, Aug 2, 2012 at 9:11 PM, Tim Goetze wrote:
>>
>> I think it's almost always a better idea to add an inaudible DC offset
>> or a square wave at the block interval or at Ny
[David Robillard]
>On Thu, 2012-08-02 at 20:31 +0200, Martin Homuth-Rosemann wrote:
>[...]
>> // denormals are zero
>> static inline float daz( float f )
>> {
>>// define an aliasing type to perform a "reinterpret cast"
>>typedef __u32 __attribute__ (( __may_alias__ )) u32bit;
>>if ( *
[Robin Gareus]
>On 07/06/2012 10:07 PM, jer...@autostatic.com wrote:
>
>> Ran a backtrace, attached the output.
>
>use
>(gdb) thread apply all backtrace
>
>to produce backtraces of threaded apps.
or equivalently,
(gdb) thr ap al bt
to reduce wear on those precious fingertips. ^_^
Cheers, Tim
__
[David Robillard]
>On Sat, 2012-06-02 at 11:37 +0200, Tim Goetze wrote:
>> As a plugin author, I consider all assumptions or guarantees about
>> the block size useless unless the plugin can tell the host exactly
>> what it wants, and get it everytime (and I'm not even su
[David Robillard]
>On Fri, 2012-06-01 at 20:22 -0400, David Robillard wrote:
>[...]
>> I am attempting to farm
>> precisely one piece of information: whether or not the above callback
>> can adequately express any reasonable block length situation; i.e.
>> whether or not any plugin could know "yes
[Adrian Knoth]
>Iain Duncan wrote:
>
>Hi
>
>>- I could pre-allocate a giant list of messages and pluck the data off
>>that
>>list when I need to make a new one
>>- I could pre-allocate a block of memory and allocate off that
>
>I'm writing from my phone, so for the sake of brevity, I will only
>
[Paul Davis]
>On Tue, Dec 20, 2011 at 8:24 AM, Tim Goetze wrote:
>
>> Alternatively, out-of-order queuing can be dealt with by sorting the
>> contents of the FIFO in the consuming thread (since the write position
>> only ever increases, sorting everything before it is
[Paul Davis]
>On Mon, Dec 19, 2011 at 8:10 PM, Iain Duncan wrote:
>
>> However, I would like to add the ability for the user to send a message and
>> have it get executed later, where later gets figured out by the engine ( ie
>> on the top of the next 8 bar phrase ). To do this, I need some way o
[Fons Adriaensen]
>Jack_delay can be used to measure the round-trip latency of a soundcard.
My M-Audio Audiophile 24/96 PCI card's roundtrip latency at 64|2|44100
(jack 0.118.0 so no -E measurement):
189.623 frames4.300 ms
The result confirms my favourable opinion of the card's perf
[David Robillard]
> No, the pragmatic thing to do is not deliberately break your plugin when
> several knowledgeable people have pointed out that doing so can cause
> countless
> problems.
Again: not the plugin is broken, but the host that assumes the port
signature not to change over different
[Jeff McClintock]
>What happens when you modify version-1 of your plugin and remove a port
>(making Version2), then later re-add a new (unrelated) port with different
>semantics? (Version 3)... Then load a project created with version 1.
>
> Does the host in THIS situation set the new port to it's
[Robin Gareus]
>On Apr 10, 2011, at 3:52 PM, Tim Goetze wrote:
>>
>> Surely you will understand that I'm not inclined to maintain two
>> versions of a plugin whose code differs in only one line.
>>
>
>That sounds like a good case of an #ifdef, no? Just gen
[Paul Davis]
>On Sun, Apr 10, 2011 at 3:43 AM, Tim Goetze wrote:
>> Following your suggestion and changing the plugin's UniqueID would
>> break perfectly good saved session files in these and other properly
>> designed host applications - instead of *preventing* breakag
[David Robillard]
> On 07/04/11 01:00 PM, Tim Goetze wrote:
>> The new port for the AutoWah plugin will indeed be appended to the
>> list of input ports; however, it will still precede the audio output,
>> adhering to the CAPS port order convention which puts inputs before
>
[Stefano D'Angelo]
>2011/4/5 David Robillard :
>> On 03/04/11 04:34 AM, Stefano D'Angelo wrote:
>>>
>>> 2011/4/2 Tim Goetze:
>>>> const int * caps = (const int *) dlsym (h, "__caps_version__");
[...]
>> Global identifiers beginnin
[Stefano D'Angelo]
>2011/3/29 Tim Goetze :
>> It is very unfortunate that such a change might break the way your
>> bridge code works, Stefano, and I would like to apologise in advance.
>> (If the addition of a 'version' symbol exported by caps.so is an
[Philipp Überbacher]
>Excerpts from Stefano D'Angelo's message of 2011-03-28 22:59:46 +0200:
>>
>> This means, if you change the port signature and maintain the same
>> UniqueID, we would have incompatibilities in the LV2 world. If you
>> create a new plugin or don't touch ports, instead, everyth
[Stefano D'Angelo]
>Aehm... there is a bug in the LRDF file, plugin CabinetI has two more
>model entries than the port's minimum-maximum range supports.
Thanks for spotting that. Fixed locally (along with filter mode ports
being incorrectly labeled 'model' in the RDF).
>> I'm planning to add a
[Paul Coccoli]
>On Sat, Mar 26, 2011 at 8:41 PM, Julien Claassen wrote:
>> Hello Tim!
>> thanks for the new release. I love your plugins. Wouldn't know, where I'd
>> be without the chorus (1767) or the amp (1786) and occasionally a few of the
>> other ones as well. :-) Great suite of plugins. I
The C* Audio Plugin Suite reaches version 0.4.5.
http://quitte.de/dsp/caps.html
http://quitte.de/dsp/caps_0.4.5.tar.gz
CAPS is a collection of refined LADSPA units covering a wide range of
applications, from stompbox classics to experimental oscillators.
CAPS is distributed as open source u
[James Morris]
>On 15 July 2010 18:23, Harry Van Haaren wrote:
>> http://en.wikipedia.org/wiki/Denormal_number
>>
>> A nice "quick" fix for these is:
>> if ( number < 0.1)
>> number = 0.0;
>>
>> ;-)
>
>Don't forget about negative numbers ;-)
>
>n = ((n > 0.0 && n < 0.1) || (
[Chris Cannam]
>Rubber Band Library v1.5.0 is now available.
>
> http://breakfastquay.com/rubberband/
I finally got around to set up an entirely unqualified and
unscientific comparison between rubberband and the 'stretch' utility
from the pvoc package (which basically is not much more than a d
[Olivier Guilyardi]
>On 12/17/2009 01:03 PM, Tim Blechmann wrote:
>>> +#if defined(__APPLE__)
>>> +#include
>>> +#define MEMORY_BARRIER() OSMemoryBarrier()
>>> +#elif (__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 1)
>>> +#define MEMORY_BARRIER() __sync_synchronize()
>>> +#else
>>> +#warni
The C* Audio Plugin Suite reincarnates as version 0.4.4.
http://quitte.de/dsp/caps.html
http://quitte.de/dsp/caps_0.4.4.tar.gz
CAPS is a collection of LADSPA plugins enjoying worldwide favour for
its instrument amplifier emulation. In addition, it provides a
sizeable assortment of acclaime
[Conrad Berhörster]
>On Wednesday 19 August 2009 12:37:36 Tim Goetze wrote:
>>
>> It'd be a lot more helpful to see the full source but from what you're
>> writing I'd be willing to bet you're encountering denormals.
>
>geeh, good point. So there is
[Conrad Berhörster]
>i have written a channel class, which collects data from (file) sources and
>copies it to a buffer. Jack will get this buffer and put it into his streams.
>So far, I think, this is a normal design with the following code
>
>for (unsigned int n = 0; n < nframes; ++n)
>{
>
[Paul Davis]
>> So will it link on OSX if I remove -nostartfiles?
>
>i'd suggest copying what swh's makefile does, which is something like this:
>
>gcc -flat_namespace -undefined suppress -o .libs/ringmod_1188.so
>-bundle .libs/ringmod_1188.o -lm -march=i686 -nostartfiles
>
>clearly, i was talkin
[Paul Davis]
>> I've added these attributes to _init() and _fini() sometime ago but
>> hopeful users of caps on OSX tell me linking the .so fails:
>>
>> $ g++ -nostartfiles -O2 -ffast-math -funroll-loops -Wall -fPIC -DPIC
>> -shared -o caps.so [list of object files]
>
>why are you using -nostartfil
[Paul Davis]
[...]
>and the cleanup function formerly known as fini() as
>
> __attribute__((destructor))
>
>failure to do this pretty much guarantees plugin crashes on OS X and
>other platforms that dropped init+fini support years ago.
I've added these attributes to _init() and _fini() sometime a
[Damon Chaplin]
>I can confirm that initializing the "model" and "tone" variables does
>fix the problems in CAPS. Though I'm not sure what they should be
>initialized to, so I'll leave that to you!
Thanks, that is rather helpful, and the issue may have been the cause
of real problems for some use
[Damon Chaplin]
>caps memory errors in 3 plugins
Thanks for pointing out the make invocation. I haven't used valgrind
before so my cluelessness may show again in what follows.
Anyway, when running this:
$ valgrind --tool=memcheck --leak-check=full --show-reachable=yes
./test-ladspa -p caps
[Damon Chaplin]
>I've been having problems with a few LADSPA plugins recently, so I've
>written a little test app that loads all LADSPA plugins, connects the
>ports and runs them for one cycle. (I've attached it here.)
If you're using libraries like glib, it would be very kind of you to
supply a
[Bengt Gördén]
>Den Monday 13 July 2009 23.25.39 skrev Fons Adriaensen:
>>
>> Anyone able to explain why console-kit-daemon is running
>> more than 60 threads ?
>>
>>
>> System is Fedora 10.
>
>It's being discussed at lkml.
>
>http://lkml.org/lkml/2009/6/29/487
I see there are multiple wait option
[Stefano D'Angelo]
>
>Note: a weird host could copy make a copy of the descriptors (I see
>nothing claiming this should not happen in the header file), thus ABI
>would be broken... this could maybe happen in LADSPA hosts written in
>non-C languages (why does Java come to my mind?).
No, it's not a
[Fons Adriaensen]
>
>This makes is backwards compatible, as no new field is required
>in the descriptor struct.
AFAICS, binary compatibility is not compromised by an expanded
descriptor struct, as long as additional members are appended to the
struct.
(That is not to say it's hugely preferable
[Fons Adriaensen]
>Does anyone know of a counting semaphare class/module
>in Python ? Given the lock provided by the built-in
>thread module it seems impossible to implement this
>(it does support multiple waiters which I don't need,
>but definitely is not counting). This also means that
>whatever
[Tim Goetze]
>[victor]
>>I was told to revert to 2.6.24.17 (not possible in my specific
>>case, but there you go), in this list. Or to join the tuner's list.
>
>http://www.kernel.org/pub/linux/kernel/projects/rt/patch-2.6.26.3-rt2.bz2
>
>worked for me. Applied
[victor]
>I was told to revert to 2.6.24.17 (not possible in my specific
>case, but there you go), in this list. Or to join the tuner's list.
http://www.kernel.org/pub/linux/kernel/projects/rt/patch-2.6.26.3-rt2.bz2
worked for me. Applied cleanly and compiled well after turning off
some RCU-rel
[Fons Adriaensen]
>I don't know the details of GTK, but this is a classic problem
>with high-level GUI toolsets: they don't allow you to wait
>for X events (which trigger the loop, behind the scenes) and
>anyhting else at the same time in the same thread. They don't
>even separate _waiting_ for an
[Paul Davis]
>On Fri, 2008-02-01 at 08:14 +0100, Yann Orlarey wrote:
>
>> The minimal combination to activate FTZ mode seems to be :
>>
>> "-O3 -march=native -mfpmath=sse -ffast-math"
>>
>> when you are on a sse capable cpu. Here it works, you don't have
>> denormals any more :-).
>
>Ardour tak
[Chris Cannam]
>On Monday 10 December 2007 23:36, Tim Goetze wrote:
>> Unfortunately it also has to be noted that after the onset, the
>> voice body in rubberband's output is not sounding quite as good as
>> with 'stretch' -- there's a faint choru
[Chris Cannam]
>Rubber Band is an audio time-stretching and pitch-shifting library and
>utility designed for musical applications.
>
> http://www.breakfastquay.com/rubberband/
>
>It includes a library that supports a sample-accurate multithreaded
>offline mode and a real-time lock-free streaming
[Rémi Thébault]
>I join my source code, my Makefile and the compiled library (x86).
>If somebody could have a look or just give some piece of advice on how
>to make a plugin recognized, this would be a great help for me.
Since you are compiling as C++, you need
extern "C" {
/* your exported sym
[Juuso Alasuutari]
>In my optimistic vision, the authors of some respected existing plugin
>collections would decide to join forces and set up a common playground. But
>things may not always work according to one's wishes -- especially if he
>isn't a plugin author himself...
If you take a look
[Steve Harris]
>
> My preferred solution would be if host authors fixed their sample-rate-hint
> implementations, but it's not perfect even when it does work.
>
> In general I want to be able to say something like minimum 10Hz, maximum 0.45
> x
> samplerate.
True, that would be nice. In the mean
[Steven Chamberlain]
>Unfortunately jack-rack doesn't clamp the minimum frequency, so it goes
>right down to zero, and 0-20Hz takes up three-quarters of the slider's
>adjustable range (which is rather tedious).
>
>According to the XML file the minimum frequency is set as 0.0 * sample
>rate. Why do
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