On Mon, Oct 30, 2006 at 10:07:06AM -0500, Lee Revell wrote:
Some ALSA plugin? Of course, doing it in JACK's ALSA backend is also
reasonably simple, but would require small changes in many places to
override samplerates, etc.
I don't think it even needs to be a plugin, just tell ALSA
On Wed, Oct 04, 2006 at 10:51:08PM -0500, Andres Cabrera wrote:
I've written a paper analyzing the characteristics of some software
dynamic range compressors. The interesting part concerning linux, is
that all my results show jaggedness on the gain reduction curves. I did
the tests using
On Thu, Oct 05, 2006 at 09:14:26AM -0500, Andres Cabrera wrote:
Just to clarify the problem I'm encountering, here's a zoom in on the
processed wave, exactly at the point where gain reduction starts
occuring. This screenshot doesn't apply any of the methods proposed in
the paper, it shows
On Thu, Oct 05, 2006 at 09:14:26AM -0500, Andres Cabrera wrote:
Just to clarify the problem I'm encountering, here's a zoom in on the
processed wave, exactly at the point where gain reduction starts
occuring. This screenshot doesn't apply any of the methods proposed in
the paper, it shows
On Wed, Jul 26, 2006 at 01:25:38PM +0300, Ari Kauppi wrote:
Good try but it still has at least one potential problem: according to
MIDI spec running status should be set only with channel messages.
Sysex/common messages should reset it to undefined (0).
I don't have a midi spec at hand
On Wed, Jul 26, 2006 at 01:16:27PM +0100, Chris Cannam wrote:
The channel is part of the status byte.
Yes, of course. It's really too hot here.
--
FA
Lascia la spina, cogli la rosa.
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 time
(all formats, jackified), so I installed in on a fairly
standard SuSE 10.0 this
On Mon, Jul 03, 2006 at 12:52:42PM +0200, Florian Paul Schmidt wrote:
Just a guess. Did you choose a different samplerate converter than the
fastest? That was what it is choking on here sometimes. The fastest
works fine here [though of course with reduced sound quality]..
I used the 'medium'
On Mon, Jun 26, 2006 at 09:07:11AM +0100, Steve Harris wrote:
On Sun, Jun 25, 2006 at 04:30:40 -0400, Dave Robillard wrote:
Plus, it's completely useless for GUIs in a separate process, while LV2
is not (it's just a data file, anything can load it, it's not even
architechture dependent).
On Mon, Jun 26, 2006 at 01:19:21PM +0100, Steve Harris wrote:
On Mon, Jun 26, 2006 at 01:53:13 +0200, Alfons Adriaensen wrote:
If the GUI is in a separate process and connected by e.g. OSC, it
could as well be on a machine that doesn't have the plugin files.
Or that has a different
On Mon, Jun 19, 2006 at 12:04:56PM +0200, Thorsten Wilms wrote:
It's rather LADSPA which shot itself in the foot:
With LADSPA, fixed data fields providing information about the plugin (like
the number and type of ports) are inside the plugin binary, resulting in a
number of problems:
All
On Thu, Jun 15, 2006 at 09:07:35PM +0200, Albert Graef wrote:
This is hard to get right for low-level DSP programming because
efficiency is of utmost importance, but I think that Faust does this
pretty well. The only roadblock right now is that it is not suitable for
multirate processing
On Thu, Jun 15, 2006 at 08:53:11AM -0400, Paul Davis wrote:
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,
I've had this request on a non-Linux list:
Can you provide more feedback (about) ... multimedia players capable of
decoding Dolby AC3, DTS, DVD-Audio, etc. on a Linux machine?
What about software players running on a CAR-PC and surround-capable?
I know very little about the whole field of
On Sat, Apr 29, 2006 at 01:00:04AM +, carmen wrote:
It's not possible for a host to know how to scale a port from just the unit
labeling. Unit labeling and input value scaling are independent, in fact
are completely orthogonal except in certain conventional cases like
IEC for some
I've gone through the docs a number of times, but can't find
a way to read / write the channelmask and guid in a WAVEFORMAT
extensible file.
Is it possible ?
--
FA
Follie! Follie! Delirio vano e' questo!
On Fri, Apr 14, 2006 at 07:46:38PM +1000, Erik de Castro Lopo wrote:
Alfons Adriaensen wrote:
I've gone through the docs a number of times, but can't find
a way to read / write the channelmask and guid in a WAVEFORMAT
extensible file.
I thought the guid was determined by the data
On Mon, Apr 10, 2006 at 03:11:49PM +0200, Julien Claassen wrote:
Hi all!
I know it's completely OT, but I think there maybe people here, who could
help me.
Problem is: I'm still on my libcui (character user interface) project and I
wonder:
I push a button, slide a slider... How does
On Mon, Apr 10, 2006 at 03:11:49PM +0200, Julien Claassen wrote:
Hi all!
I know it's completely OT, but I think there maybe people here, who could
help me.
Problem is: I'm still on my libcui (character user interface) project and I
wonder:
I push a button, slide a slider... How does
On Thu, Apr 06, 2006 at 02:36:34PM +0400, Dmitry Baikov wrote:
Hello!
I'm planning to get RME Multiface (II) and have a question to it's owners.
Can you, please, post its latency in audio loop, measured with jaaa in
2x64 buffer setup.
Better use jdelay for that, that's why it exists !
--
On Thu, Apr 06, 2006 at 04:50:37PM +0400, Dmitry Baikov wrote:
Better use jdelay for that, that's why it exists !
Oops, my bad - I mixed up the names...
P.S. that's why I get no responces - they are still measuring
It shouldn't take more than a minute...
--
FA
Follie! Follie!
On Wed, Mar 01, 2006 at 11:07:17PM +0100, David Kastrup wrote:
I am not asking for a solution. I am asking for a clue. The man page
to aplay does not mention what a PCM actually is. It just tells you
to list them with -L.
This is my main gripe with ALSA documentation: it often uses terms
On Thu, Mar 02, 2006 at 12:59:12PM +, James Courtier-Dutton wrote:
If you don't like the current documentation, you are welcome to improve it.
Just update the wiki.
I'd be happy to, if only I could just be a bit more confident
about my own knowledge. Currently I'm really in no position to
On Wed, Jan 25, 2006 at 10:32:17PM -0500, Lee Revell wrote:
On Wed, 2006-01-25 at 22:27 -0500, Paul Coccoli wrote:
In Programming with POSIX Threads by David R. Butenhof,
pthread_mutex_unlock is said to do this:
Unlock a mutex. The mutex becomes unowned. If any threads are
waiting
On Thu, Jan 26, 2006 at 04:02:45PM +0100, Jens M Andreasen wrote:
On Thu, 2006-01-26 at 11:54 +, James Courtier-Dutton wrote:
Thread A will only get the lock if the kernel happens to do a task
switch between Thread B to Thread A.
Not according to posix. Perhaps all this talk about
On Fri, Jan 20, 2006 at 12:31:34PM +0100, Carlo Capocasa wrote:
new to this list and knowing absolutely nothing about C++ audio
programming I would like to ask for a little bit of help.
I want to create a console keyboard to MIDI application and for this I
would like to read the physical
On Fri, Jan 20, 2006 at 02:35:47PM +0100, Carlo Capocasa wrote:
Yes, vkeybd is what I am using now. I set out to create a console
alternative that is also more stable when pressing many keys at once.
Computer keyboards are not designed to handle this. If some key
combinations do not work in
On Mon, Jan 16, 2006 at 11:38:24AM -, Dave Griffiths wrote:
thinkpads are good laptops for live use, as they are pretty robust and
reliable.
I can confirmn that. I'm quite happy with mine. Only problem is
that at 1.7G it's rather hungry (for battery power) - more so if
for reliable audio
On Thu, Jan 12, 2006 at 04:07:35AM +0300, Andrew Gaydenko wrote:
convolve.cc: In constructor `Convdata::Convdata(size_t, size_t, int)':
convolve.cc:50: error: array bound forbidden after parenthesized type-id
convolve.cc:50: note: try removing the parentheses around the type-id
convolve.cc:
On Thu, Jan 12, 2006 at 02:45:14PM +0300, Andrew Gaydenko wrote:
Yes, I did it just now. I have tried to use the demo.conf config, all works.
Great ! I'll upload a corrected version later today.
The most impressive is a few seconds period after a 'pause' pressed :-)
Yes, it's a nice reverb.
On Wed, Jan 11, 2006 at 12:14:07PM +, Steve Harris wrote:
The maximum peak performance of a modern CPU is 1 or 2 cycles per
multiply, but in practice memory bandwidth throttles that. 10 might be
more typical at a rough guess.
For most intensive DSP applications, PCs are limited by memory
On Wed, Jan 11, 2006 at 09:09:46PM +0900, Srinivas Reddy wrote:
I am having an audio engine with MIDI PLAYER and MIDI SYNTHESIZER which is
running in real time mode . It means that
the player layer send the MIDI events directly to the synthesizer that are
contained in the current frame.
On Wed, Jan 11, 2006 at 01:53:25PM +, Steve Harris wrote:
I agree about ARM assembly, I have written some (not DSP related) many
years ago and it was quite straightforward.
Another ex Acorn user ???
Does this ARM chip have real fixedpoint hardware, or do you have to do bit
On Wed, Jan 11, 2006 at 02:08:05PM +, Steve Harris wrote:
Damn, does it show ;)
You're not alone :-)
What helps enormously on the ARM is that all arithmetic instructions can
include a (no overhead) shift on one of the operands. There are some
other unique things, such as the 16
On Thu, Jan 05, 2006 at 06:19:05AM -0500, Bill Allen wrote:
fons adriaensen wrote:
AMS should handle multiple patches without requiring a separate instance
for each.
That would be great, as ams is my favorite synth, but I haven't found a
way to do it. When you say should are you saying
On Tue, Nov 15, 2005 at 02:12:55PM +0100, Jens M Andreasen wrote:
(good to see you're back on line :-)
On a related subject: How is level one cache replaced with new data,
should one (or ones compiler) decide to use some of the prefetch
instructions available from Intel PII and up? It would
On Thu, Nov 10, 2005 at 04:24:25PM +0200, Juhana Sadeharju wrote:
Fixing the hardware input and output levels makes sense to me.
The input devices all have fixed SNR -- it does not help to
crank up the soundcard input level as it brings the noise up.
The output would be fixed for the same
On Mon, Nov 07, 2005 at 08:37:22AM -0500, Fred Gleason wrote:
Similarly, it seems that virtually no 'recording' style board sports a cue
buss, while the very thought of trying to run a radio operation without such
a buss is enough to induce anxiety in this old 'master control' operator.
On Wed, Nov 02, 2005 at 11:05:34AM +0100, St?phane Letz wrote:
Le 31 oct. 05 ? 02:18, fons adriaensen a ?crit :
A big advantage of using futexes in shared memory would be
that they don't have to be recreated each time the callback
order changes - unlike the pipes, they are not bound to a
On Wed, Nov 02, 2005 at 11:56:59AM +0100, St?phane Letz wrote:
I must be missing something essential here. Access to named things
that have to be opened is normally by a file descriptor, and file
descriptors are bound a process. How then can you give *all* clients
access to the named pipe or
On Wed, Nov 02, 2005 at 12:38:49PM +0100, St?phane Letz wrote:
Yes, clients use open *once* when the new client opens. This is done
in a non RT thread (what we call the notification thread that also
handle all non RT events like callback...)
This means that changing the graph order can
On Wed, Nov 02, 2005 at 01:57:47PM +0100, St?phane Letz wrote:
So if there are N clients, each of them needs N file descriptors open
all the time. System wide the complexity grows as N^2. Not really a
good way to tackle an O(N) problem IMHO.
Yes but in the jackdmp data flow kind of model,
On Tue, Oct 25, 2005 at 08:38:18AM -0500, Richard Smith wrote:
I haven't found any RIAA filters yet so I guess I'm looking at
writeing one. So does anyone have any information on where to find
the official RIAA curve to make a plugin from?
The curve as used in most preamps is the combination
On Wed, Oct 26, 2005 at 01:07:33AM +1000, Loki Davison wrote:
Is there some sane reason for not just using a turntable preamp? The
phono signal level is quite low and i'm assuming recording it as line
level and amping in software isn't the nicest or easiest way to do it.
I really fail to see
On Wed, Oct 05, 2005 at 07:48:10PM +0200, Derek Holzer wrote:
* MCP Plugins 0.3.0
These lack a proper ./configure file, and I did not edit the Makefile at all.
Compile ends with:
g++ -shared mvclpf24.o mvclpf24_if.o exp2ap.o -o mvclpf24.so
powerpc-apple-darwin8-g++-4.0.0: unrecognized
On Wed, Oct 05, 2005 at 11:23:06PM +0200, nescivi wrote:
DH [More spew available on request]. Also cannot get ./configure to recognize
DH fftw.h and the other FFTW headers, no matter how many PATH combinations I
try.
I had this same problem on Linux (though I did not try as much PATH
On Mon, Oct 03, 2005 at 07:59:59PM -0400, Paul Davis wrote:
i consider that the CV never considers changes
to have been carried out just because it asked. anything else is not
really MVC. maybe the change requested by CV is not possible for M at
this time.
Good point. So this means that when
On Mon, Sep 19, 2005 at 10:46:26AM +0200, Mario Lang wrote:
I've feared this effect of half-hearted accessibility support
for graphical desktops under Linux, and it seems my fears have come
true: Just because there *is* an attempt to make GUIs accessible
doesnt necessarily mean that all
On Thu, Sep 15, 2005 at 10:26:43PM +0200, Magnus Hjorth wrote:
The model shouldn't know about which buttons etc exist in the
GUI and how to display things, so the event should be more like
('user wants to perform action #123') and the response should
be more like ('the model state changed to
On Thu, Sep 15, 2005 at 02:54:50PM +0200, Richard Spindler wrote:
I actually use some some text based applications quite often, and I
really like that the provide some kind of command language so I only
type in what I wan't to do and here we go. This however is a totally
different approach
On Thu, Sep 15, 2005 at 04:28:52PM +0100, Martin Habets wrote:
Protocol wise, it would be interesting to just use the X protocol.
i.e. create an X server that writes to the console. This would work
for any gui application, and you could ignore uninteresting graphics
stuff.
Not sure how to
On Thu, Sep 15, 2005 at 07:12:14PM +0200, Esben Stien wrote:
All apps should really use this style. I'm much more comfortable
giving direct commands to programs, even when 3d modelling, editing
sound files or pictures, whatever I can think of, really.
Couldn't agree more. One good example is
On Wed, Aug 10, 2005 at 11:34:39AM +0200, Florian Schmidt wrote:
a] is it possible to use threading in a DSSI?
I've done this in some LADSPAs, it works.
b] would a RT prio of 1 (for the convolution thread) be an OK
compromise? It will be lower than all audio stuff on a typical jack
system?
On Wed, Aug 10, 2005 at 12:28:01PM +0200, Florian Schmidt wrote:
Let's play this through with an example. For simplicity's sake let's
assume the host always calls the plugins run() method with a constant
buffersize of 1024 frames (there's still no requirement for this though
...
Sorry, I
On Mon, Aug 08, 2005 at 04:39:32PM -0500, Andres Cabrera wrote:
Just tried it, and it looks very cool. Could you explain how a room can
be tuned using japa, or point to some reference for this?
You can use JAPA as the analyser when equalising a sound system.
This is not exactly 'tuning a
On Thu, Aug 04, 2005 at 08:34:59AM -0500, Andres Cabrera wrote:
Maybe it's denormal problems? You can try adding a noise generator set
to a very low level before the plugin, and see if this fixes it.
But I would think the tap plugins were denormal safe...
That was my first idea as well, since
On Thu, Aug 04, 2005 at 03:44:57PM +0100, Simon Jenkins wrote:
That FLUSH_TO_ZERO macro doesn't always work though:
http://music.columbia.edu/pipermail/linux-audio-dev/2003-August/004581.html
Unfortunately the fix I suggest at the bottom of that mail doesn't always
work either (it turns
Hello all,
I'm trying to clear up my mind as to what conventions to follow
in a GUI for the actions of zooming in and out e.g. a spectrum
or an impulse response window. It's not the intention to launch
a debate about this, just to collect other people's ideas.
The accepted model for scrolling
On Tue, Jul 26, 2005 at 09:45:45AM +0100, Steve Harris wrote:
Yes, note that its not much good as a random number generator as
its easily predictable,
All 'psuedoramdom' generators are predictable, that doesn't make
them less 'good', except for cryptographic applications.
but in this case
On Mon, Jul 11, 2005 at 02:49:22PM +0300, Artemio wrote:
Youre passed it as a paramter when your instantiated, just stash it
in the struct.
Thanks for your help!
But... I have added:
typedef struct {
unsigned long SampleRate;
...
} MyPlugin;
and then in runMyPlugin:
On Thu, Jun 30, 2005 at 12:20:06PM +0200, Olivier Guilyardi wrote:
Needs checking indeed... Spawning an additionnal thread may be required.
The reason why I don't just do it is because discussing implementations
ideas before coding has been of great benefit to me in the past, especially
On Thu, Jun 30, 2005 at 02:57:51PM +0200, Olivier Guilyardi wrote:
I recognize that MVC is very useful to big applications, but it's
not so important for small ones IMHO.
That's exactly the idea I want to warn you for. One of the cases
where I tried to make the GUI the 'center of control' and
On Wed, Jun 29, 2005 at 01:57:58PM +0300, Artemio wrote:
I have a problem with creating a plugin with more than one control
port. In the attachment you'll find booster-simple.c which has one
Gain port and my attempt to add a second port in booster.c. For
some reason the latter cannot be
On Wed, Jun 29, 2005 at 03:43:39PM +0200, Florian Schmidt wrote:
This has a subtle bug afaict. Let's assume the host called several
process() with numframes != 512 first, then one with numframes == 512, i.e.:
1. 123
2. 432
3. 234
4. 512
The 4th process call disregards data already in
On Wed, Jun 29, 2005 at 10:46:55AM -0400, Paul Coccoli wrote:
On 6/28/05, Olivier Guilyardi [EMAIL PROTECTED] wrote:
I believe there could exist a library with which :
1 - you instantiate a core object (providing the alsa midi port as an arg)
2 - you attach to some widgets : sliders, spin
On Wed, Jun 29, 2005 at 05:45:12PM +0200, Benno Senoner wrote:
So a 2nd output ringbuffer buffer would be required.
Yes. In my lib, the two ringbuffers are part of the convolver,
and the FFT and MAC operations operate directly on them.
The API to write/read them is similar to ALSA's memory
On Tue, Jun 28, 2005 at 03:05:10PM +0200, Jens M Andreasen wrote:
On Tue, 2005-06-28 at 08:38 -0400, Dave Phillips wrote:
J. Chowning, The Synthesis of Complex Audio Spectra by Means of
Frequency Copulation
s/copulation/modulation
Are you sure ?
s/Frequency/Frequent/ :-)
--
FA
On Tue, Jun 21, 2005 at 01:17:21PM +0100, Dave Griffiths wrote:
* high dc offset
* very low frequency
* very high frequency
Add 'High level at higher frequencies'. In most speakers
systems, the HF unit will sustain considerably less power
than the LF/MF parts.
High-end PA systems (using
[Lachlan Davison]
... from a dance music making perspective i'm not sure if i
understand this customer concept. Does your fav rock band use
best practice and professional conduct? are they not professional?
[Dave Phillips]
I'm a music professional. I make my living teaching music, writing
On Fri, Jun 17, 2005 at 12:33:20AM +0100, Damon Chaplin wrote:
Out of interest, what APIs do you think GNOME and KDE should provide for
sound?
None. Why should a window manager / desktop provide its own API for
such things ?
--
FA
On Sat, Jun 11, 2005 at 09:46:23PM +0200, Mickael Vardo wrote:
Just try this simple experience: sample a 1 Hz pulse that
is triggered after a random non-quantized delay that is less
than four seconds. Sample it at a rate of 2 sps and then,
try to get the original signal back with all its
On Fri, Jun 17, 2005 at 12:48:11PM +0100, Damon Chaplin wrote:
On Fri, 2005-06-17 at 09:57 +0200, Alfons Adriaensen wrote:
On Fri, Jun 17, 2005 at 12:33:20AM +0100, Damon Chaplin wrote:
Out of interest, what APIs do you think GNOME and KDE should provide for
sound?
None. Why
On Wed, Jun 15, 2005 at 08:22:17AM -0400, Paul Davis wrote:
i don't think thats entirely fair. when jaroslav started ALSA i think he
was intent on a set of ideas that looked like the best choices at the
time. the goal was to improve lots of issues with OSS, including its
requirement for all
On Thu, Jun 16, 2005 at 10:30:29AM -0400, Paul Davis wrote:
true, but i take it you get the way CoreAudio is doing it: it means you
can drive audio processing from a different interrupt source (e.g.
system timer) because you have very accurate idea of the position of the
h/w frame pointer. In
On Mon, Jun 13, 2005 at 04:59:57PM +1000, Erik de Castro Lopo wrote:
With libsamplerate, I can state that the three sinc based converters
have the following characteristics:
SNRBandwidth
SRC_SINC_FASTEST 102.42 dB 80.23 %
On Mon, Jun 13, 2005 at 10:49:38PM +1000, Erik de Castro Lopo wrote:
The SNR and bandwith cannot be determined by reading code.
Measurement is the only option.
It's perfectly possible to calculate this, and it isn't even
very difficult. In the case of a sinc filter, everything is
determined
On Mon, Jun 13, 2005 at 11:22:57AM -0400, Paul Davis wrote:
ALSA's biggest problem was that people like me shaped its design too
much. I was trying to ensure that ALSA was useful for pro-audio setups,
and I had little interest in the desktop story. There were no
(sufficiently) vigorous
On Wed, Jun 08, 2005 at 10:20:01AM +1000, Dave Robillard wrote:
Premature optimization is the root of all evil.
That's by Donald Knuth IIRC. Most of wht I know about programming
(I mean relevant things, not language or system nitty-gritty), comes
from hist ACP series of books, and I'd agree
On Wed, Jun 08, 2005 at 09:50:42AM +0100, Simon Jenkins wrote:
Suppose I sum a vector of 5 million integers and it takes 6 seconds. And
assume - (generously![1]) - that I switch to using an array and now it
only takes 1 second. Hmmm... a 6 * speedup! So I look to see where else
my code could
On Wed, Jun 08, 2005 at 07:19:56AM -0400, Paul Davis wrote:
the nice thing about a
design pattern like STL containers is that you can toggle back and
forth between any all of them with almost no work. i can't count how
many times in ardour i have changed:
typedef vectorFoo Foos;
to
On Wed, Jun 08, 2005 at 09:12:05AM -0400, Paul Davis wrote:
SAWstudio is a pretty full-featured DAW that is, AFAIK, written almost
entirely in x86 assembler. Its blazingly fast and yet dinosaur like at
the same time, from what I hear.
Reminds me of the original version of Sibelius (the music
On Mon, Jun 06, 2005 at 09:17:54AM +, vanDongen/Gilcher wrote:
As I recently found out, this can be very messy :)
Indeed :-)
The basic algorithm is this:
Each horizontal pixel represents n samples. Usually n is pretty big.
Of those n samples you take the min and the max, and then you
On Mon, Jun 06, 2005 at 12:05:24PM +0200, Olivier Guilyardi wrote:
And I now think that this trigger-detection engine should indeed output
midi, so that you can plug into many different devices/applications. Now,
once a midi signal is issued, and ends up playing a sample or synthetizing
a
On Mon, Jun 06, 2005 at 12:14:41PM +, vanDongen/Gilcher wrote:
What kind of interpolation is required to visualize the DAC output of a
sampled waveform?
This depends mostly on the maximum frequency you want to display or
measure accurately, and on the level of accuracy required.
You can
On Mon, Jun 06, 2005 at 01:53:10PM +0400, Andrew Gaydenko wrote:
Will anybody be so kind to suggest steps to find this difference
reason?
Andrew
// .asoundrc fragment
pcm.!default {
type plug
slave {
pcm 2x4
On Mon, May 23, 2005 at 11:03:30AM +0200, Richard Spindler wrote:
if (*p_A0 *p_B0) {
*p_output =(*p_A+1)*(*p_B+1)-1;
} else {
*p_output =2*(*p_A+*p_B+2)-(*p_A+1)*(*p_B+1)-3;
}
What is this
On Mon, May 23, 2005 at 12:18:58PM +0200, Viceic Predrag wrote:
Well, you're right. But if you want to finish with signal thats limited to
[-1:1] you got to have some normalization somewhere..Or I'm totaly wrong?
Since everything in LADSPA (and I assume in your app) is floats, there is
nor
On Thu, May 19, 2005 at 04:06:19PM +0200, Florian Schmidt wrote:
I don't know of any _reliable_ constant delay (jitter free) way to
schedule events happening during period N for playback during period
N+1. If anyone does, please enlighten me.
Call jack_frame_time() when you get the event, and
On Thu, May 19, 2005 at 06:12:56PM +0300, Juhana Sadeharju wrote:
I dislike that the Jack buffersize must be turned up for all
clients when one client does not perform well. It well could
be that I would like to use the buffersize 32 for
A/D -- EQ -- M -- D/A
and the buffersize 256 for
On Thu, May 19, 2005 at 05:25:37PM +0200, Florian Schmidt wrote:
...
The keypress cannot be scheduled for period N+1 (with constant delay) as
the process_n() (which prepared the buffer that will be audible during
period N+1) is already done. It can be put into the buffer by
process_n+1().
On Wed, May 18, 2005 at 11:29:47AM +0200, Mario Lang wrote:
I will try to illustrate, but I dont use SCUM, so its untested,
but I am pretty sure you'll get the idea:
SynthDef(onetwoonetwo,{ arg out=0, freq=440;
Out.ar(out,
SinOsc.ar(freq, 0, 0.5)
)
}).send(s);
On Wed, May 18, 2005 at 10:32:14AM -0400, Ivica Ico Bukvic wrote:
If I have a statement that does:
Out.ar(out, SinOsc.ar(freq, mul: 0.5))
Then it works.
If I have a statement that does:
Out.ar(out, SinOsc.ar(freq * 100, mul: 0.5))
This also works.
Now if I have:
On Fri, May 13, 2005 at 02:18:38AM +0100, Steve Harris wrote:
My preferred form would be something like
/std_prefix/inst_name/base_freq f base-frequecy
/std_prefix/inst_name/note_on iff note-id octave velocity
/std_prefix/inst_name/note_off if note-id velocity
What's the octave param
On Fri, May 13, 2005 at 10:16:41AM +0200, Albert Graef wrote:
/set voice gate freq gain
...
The disadvantage of this fairly basic scheme is of course that the
client has to dispatch the voices himself.
It's possible to have the best of both worlds and remain close
to midi, something
On Fri, May 13, 2005 at 08:23:34AM -0400, Paul Davis wrote:
Camel Audio has released PhatSpace, a bundle of two multi-effects
designed for musicians rather than engineers.
now check out the screenshots:
http://news.harmony-central.com/Newp/2005/CamelSpace-CamelPhat-30.html
The irony
On Wed, May 11, 2005 at 07:48:20PM -0400, Pete Bessman wrote:
On Wed, 2005-05-11 at 00:04 +0200, Fons Adriaensen wrote:
but I've no desire to live in the US
until at least the next regime change.
Hmm... in that case, I'll vote Republican again the next election.
A wise decision. My
Hello all,
I'm planning the OSC-fication of Aeolus, and would like to have
some comments / feedback on the current ideas (they could well
be braindead, in which case you are kindly requested to say so).
The setup I have in mind is as follows:
- There will be one UDP server socket. This socket
On Thu, May 12, 2005 at 11:01:15AM -0400, Jesse Chappell wrote:
Have you considered process separation for the engine and the
local GUI ? At least an option to run aeolus with no gui/x.
Not process separation, but the local user interface will be
a configuration and comannd line option: --x11
On Thu, May 12, 2005 at 11:40:51AM -0400, Jesse Chappell wrote:
No, i mean the arguments the engine will use in the callbacks
are conventionally specified in the OSC API for these comms.
Sorry but this still escapes me... Could you give an example ?
--
FA
On Thu, May 12, 2005 at 05:54:15PM +0200, stefan kersten wrote:
On Thu, May 12, 2005 at 05:22:43PM +0200, Alfons Adriaensen wrote:
One thing I forgot to mention regarding /addclient : the response
to this will include a client ID (integer) that is a required
parameter to all polled
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