are emergency features that are not used normally.
Again I tend to agree with David. Stopping the stream of events should
not imply that all sounds stop, even if this may be what you want most
of the time.
--
Fons Adriaensen
Alcatel Space
invited !!
--
Fons Adriaensen
= start time, and the
buffer if pre-filled before starting, then the k-th sample will reach
the DA converter at T + (k + B) / Fs. The time the sample was computed
doesn't matter.
--
Fons Adriaensen
Alcatel Space
.
I was just trying this out (on Solaris ...) when this message came in.
It works, and it's simple (if you have the habit of working with X
directly).
--
Fons Adriaensen
ALCATEL SPACE
David O'Toole writes:
Probably. I think it was Bjarne Stroustrup who said
something along the
lines of Every use of a define is an instance of a programmer not
programming correctly. But I was just wondering if it was some
portability thing or something.
I think
could avoid all the
memcpy() calls by just manipiulating buffer pointers. This does not
increase required buffer sizes, as you need two buffers anyway.
--
Fons Adriaensen
ALCATEL SPACE
using different blocksizes
at a time (assuming sample frequencies are coherent) ?
--
Fons Adriaensen
ALCATEL SPACE
Frank Barknecht writes:
Hallo,
with the LAD meeting getting closer, I'm getting a bit curious about,
what the plans are for the open Linux Sound Night on 15.3.? Will we
hear some of you guys perform and Paul records it?
And will there be a Ladies' Programme, as at the AES conventions ?
Martijn Sipkema writes:
According to the mLAN spec you need a buffer of around ~250us (depending
on format) to collate the packets.
Still there is no guarantee that 10 packets always have exactly the same
number of samples. You say the mLAN spec says you need a buffer of
around
variables.
--
Fons Adriaensen
ALCATEL TELECOM
(), which takes a struct timespec * referring to
an absolute time, the same you can get with clock_gettime(). I had the
impression this gave me almost microsecond resolution timing (but I did
not look closesly at the precision at the time).
--
Fons Adriaensen
ALCATEL SPACE
Roger Larsson writes:
This can easily be avoided by making these 'free-running' threads wait
on an event that has the right frequency, e.g. a counting semaphore
that is incremented by the output module each time a block is processed.
This way you can eleminate all buffering between the
Steve Harris writes:
On Fri, Feb 28, 2003 at 03:49:59 +0100, Fons Adriaensen wrote:
Is this really new ? I was porting some C++ thread classes (originally
developed for Solaris) to 2.4.19 this week. The ITC mechanism uses
pthread_cond_timedwait(), which takes a struct timespec
Paul Davis writes:
the kernel gives non-SCHED_FIFO/RR threads roughly 1/HZ
resolution.
Seems to be 20 ms on my unpatched standard SuSE 2.4.19.
I imagine this can be lowered by modifying the kernel config ?
SCHED_FIFO/RR threads can get better than that if the
delay is very small, but its
On Thu, Jun 19, 2003 at 11:23:07PM +0200, Tom Weber wrote:
!
Instead of doing a discrete fourier transform when reading a small
frame of the sound, do a dense transform (every 0.1 Hz?) and pick out
the peaks. Then assume that a similar enough frequency in the next
frame comes from the same
On Sun, Jun 22, 2003 at 10:50:39PM +0100, DG Malham wrote:
Not that I know of, yet.
Dave
On Fri, 20 Jun 2003, oliver thuns wrote:
hello,
i'm looking for a solution to encode an ambisonics signal to 2 channel
UHJ in real time. is there for example a LADSPA or (linux
On Sun, Jul 06, 2003 at 05:12:08PM -0500, Tom Felker wrote:
VLevel is written in C++. I have two questions. First, why do most
other plugins allocate and free copies of their strings and structures,
instead of just passing the literal (as I do)? The declarations in
ladspa.h don't allow
On Sun, Jul 06, 2003 at 05:35:41PM -0300, Juan Linietsky wrote:
4-Interface abstraction for plugins.: We all know how our lovely X11 does not
allow for a sane way of sharing the event loop between toolkits
This is a problem of those toolkilts, not of the X system. They all combine
two things
On Thu, Jul 17, 2003 at 05:22:19AM -0300, Juan Linietsky wrote:
On Wednesday 16 July 2003 22:21, Nasca Paul wrote:
Hi.
I released ZynAddSubFX 1.4.2
I can understand about metal lyrics being anti-christian, we've seen
that countless times in groups from megadeth to iron maiden,
But I
Hello list,
I hope there are some other users of the Terratec EWS88MT card out
there, because I seem to be in serious trouble - the card 'works'
but not in a way that makes it useful.
I've got a test program that outputs the same signal (1 kHz sine at
-6 dB below peak level) to all eight
On Fri, Jul 25, 2003 at 10:38:40PM -0400, Paul Winkler wrote:
there is/was a bug in some version of envy24control that caused this
behavior. I forget which version of alsa that corresponded to.
0.9.2 or something? check the alsa-user archives...
Thanks for the info. I've found out a bit
On Wed, Nov 05, 2003 at 10:43:17AM +, Steve Harris wrote:
Nope, polyphases only filter at fs/(2^n), they are used for resampling and
reconstruction.
You seem to use a very odd definition of 'polyphase filter'. In fact any
FIR you like can be
Hi Paul,
have you considered the possibility that Xlib is not statically
thread-safe? if there are any globals in the implementation of Xlib,
this scheme will fail as soon as the host is using Xlib as well.
Yes, and if X does its bookkeeping in terms of 'processes' and not 'clients'
the same
On Wed, Nov 19, 2003 at 08:03:06PM +0200, Juhana Sadeharju wrote:
Hello. Who of us are working on a modular synth GUI where user
grabdrag modules and connects them with cables? I'm myself
interested in the editor GUI development --- there already
are many modular audio engines, but not
On Wed, Jan 14, 2004 at 06:42:58PM +, Mike Rawes wrote:
With the exception of samples or other 'blob' data (e.g. IR impulses), I'm
fairly confident that a good modular could be constructed with LADSPA as it is.
I'm prepared to be proved wrong though!
You're probably right about this. But
On Wed, Jan 14, 2004 at 01:46:52PM -0500, Dave Robillard wrote:
Quick digression about LADSPA in ams: is there a reason exported
control ports on LADSPA plugins don't work (at least for me anyway)? I
realize control ports run at a different rate than the audio, but since
the ports are
On Fri, Jan 16, 2004 at 09:08:33PM +0100, Joost Diepenmaat wrote:
... most plugins just let the GUI be generated by the host, and
of those that don't, only a couple actually improve the situation -
actually, the only one I can think of is the default mixer plugin,
which has 16 channels with
On Fri, Jan 16, 2004 at 02:46:25PM -0500, Dave Robillard wrote:
I'll just assume you have your reasons. But just because AMS is going
to have metadata doesn't mean some modular synth can't use LADSPA
without doing so.
Nor did I say so. To illustrate this point, let me mention the problem of
On Fri, Jan 16, 2004 at 02:49:05PM -0500, Dave Robillard wrote:
On Thu, 2004-01-15 at 08:52, Alfons Adriaensen wrote:
On a more general tone: I think we should be less afraid of complexity
and learning curves. Dumbing down maybe required in order to be popular,
but that's not my aim. And
On Sat, Jan 17, 2004 at 10:20:00AM +, Steve Harris wrote:
Thats true, but the voice controller is generally part of the host
environment, not a plugin. It would typically be the think wihich is
resonsible for splitting the MIDI data to multiple sub-patches and routing
the CV streams
On Mon, Jan 19, 2004 at 08:56:52PM +, Simon Jenkins wrote:
Worse:
JACK - A - JACK - C - JACK - B - JACK
Where C is in a separate client.
Now...
+---+
| |
+---|- C -|--+
| | | |
| +---+ |
|
Thanks a lot Kai and Eric (see below) for your help.
On Tue, Feb 03, 2004 Kai Vehmanen wrote:
ecasound -a:1,6 -f:32,12,44100 -i alsa \
-a:1 -f:32,1,44100 -o t1.wav \
-a:6 -erc:6,1 -f:32,1,44100 -o t6.wav \
-a:2 -i t2.wav -ea:200 \
-a:5 -i t5.wav -erc:1,5
On Fri, Feb 06, 2004 at 04:08:31PM -0500, Dave Robillard wrote:
Just click on Export control ports as module ports.
Sure, but they don't actually work. At least not for me, from the AMS
discussion on this list I thought it was a known limitation (since
everything in AMS is audio rate).
Hello List,
I've finally found the time to put my Linux Audio things online, at
http://users.skynet.be/solaris/linuxaudio.
You will find there
* the latest releases of the MCP, REV and VCO plugins (previously on
the alsamodular site),
* some things that are under construction,
* two ogg
On Fri, Mar 05, 2004 at 02:10:30AM +0100, Tim Goetze wrote:
dancing forever around the S in ladspa, yelling 'heretic' at any
extension proposal, is only going to make us the fools of the
universe. we only have this standard and things are evolving, and so
it also must.
I couldn't agree
On Mon, Mar 08, 2004 at 05:58:48PM +, Steve Harris wrote:
I dont think that scale markings, / enumerations, defaults and units are
logically connected to the same extent that upper and lower bounds are.
That is a matter of opinion of course.
Well, for me ENUMs on an integer param that
On Mon, Mar 08, 2004 at 10:06:25PM +0100, Tim Goetze wrote:
[Fons Adriaensen]
Can you point me to some document that contains that requirement ?
not that i knew of any. simple reasoning tells us you cannot, for
example, connect a LFO (-1 .. 1) to, say, a phaser modulation depth
(0 .. 1
On Mon, Mar 08, 2004 at 11:23:20PM +0100, Fons Adriaensen wrote:
On Mon, Mar 08, 2004 at 10:06:25PM +0100, Tim Goetze wrote:
while we're at it, i'd also like to know why limiting the value-label
mapping to integers is beneficial.
It's two different problems that happen to map to similar
On Tue, Mar 09, 2004 at 12:46:09AM +0100, Tim Goetze wrote:
[Fons Adriaensen]
You can, and you're responsable for the result. I wouldn't want a
host stopping me from doing this, and in fact do similar things all
the time in AMS.
ok, then you've probably already connected LFOs to TOGGLED
On Tue, Mar 09, 2004 at 06:49:46PM +0100, Tim Goetze wrote:
thanks for providing the code. i'll not argue it is not complete
seeing that it seems impossible to obtain ladspa.h or gcc from your
workplace. :)
It's not impossible to get them, and in fact I'm using five versions
(for different
Hello list,
After much debate, I propose to extend the current LADPSA 1.1 interface
specification in the way documented below. Two new LADSPA_HINT bits are
introduced. Both can be ignored by existing hosts without any ill
consequences.
The purpose of the first new bit is to allow simple hosts
On Tue, Mar 09, 2004 at 09:05:16PM +0100, Tim Goetze wrote:
if you don't even see compactness as a virtue, there's only elegance
remaining to recommend your proposal. elegance at the cost of
increased complexity in understanding and implementation, no matter
how you look at it.
It *is*
On Fri, Apr 09, 2004 at 04:51:57PM +0100, Steve Harris wrote:
On Fri, Apr 09, 2004 at 04:47:59 +0200, Alfons Adriaensen wrote:
The advantage of having a real dictator is that no concensus is needed :-)
I think the word benevolent was used ;)
Yes, but a benevolent dictator, seeing his
On Fri, Apr 09, 2004 at 10:04:16PM +0200, Samuel Abels wrote:
There is a need for a GTK2 midi sequencer though. *hint,
hint* :)
Actually, I have been thinking about that, also. Actually, I have even
created a (not completely finished) display widget for MIDI tracks using
GTK2-perl,
On Sat, Apr 10, 2004 at 06:08:31PM +0200, Kjetil Svalastog Matheussen wrote:
Thats not what I said. Or ment at least. I said; use a high-level language
for high-level operations. I'm not saying: Do computer-intensive/realtime
critical operations with lisp/python/ruby/etc. Ardour consist of
afternoon, probably along the route
Karlsruhe - Frankfurt - Koeln - Aachen - Brussels - Antwerp.
So maybe we can share part of the trip.
As I want to be back before midnight, I'll have to leave around 15:00
to 16:00. If that would correspond to your plans, let me know.
--
Fons Adriaensen
On Sun, Apr 18, 2004 at 04:32:44PM +0200, Julien Claassen wrote:
Hi Fons!
It would be very cool, if we could take a ride to Colone together. You
wuoldn't need a ticket for that passage. I'd like to see the panel discussion
though. But, well I think four PM would be ok.
Hello Julien,
I
Hello all,
The first release of FIL-plugins is now available at
http://users.skynet.be/solaris/linuxaudio
There's one plugin in this first release, a four-band parametric
equaliser. Each section has an active/bypass switch, frequency,
bandwidth and gain controls. There is also a global bypass
On Tue, May 11, 2004 at 06:01:04PM -0400, Paul Davis wrote:
there are fundamentally different approaches to handling i/o when it
involves hardware. one of them is based on the traditional unix
read/write model, the other is based on a callback/interrupt
model. its not easy to reconcile these
On Wed, May 12, 2004 at 01:09:37AM +0200, Tim Goetze wrote:
The first release of FIL-plugins is now available at
first impression: sounds real good, nice to have smoothened controls
too. could do without the global and section bypass though. cpu is ~
3.5 % @ 64/44.1 on this 1.7G athlon,
On Thu, May 13, 2004 at 11:39:27AM -0400, Jesse Chappell wrote:
I assume you meant 4 and 8 respectively. What about 3?
Yes, of course.
Incidentally, I notice that in 0.1.0 that when you specify values 5-7 it segfaults,
and 8 causes jackd to hang up entirely on my system.
Shame on me. I
On Fri, May 14, 2004 at 05:55:07PM +0200, Marcus Andersson wrote:
Alfons Adriaensen wrote:
Another point. I've defended the adoption of simple integer enumerations
(corresponding to a C switch) using the argument that it is the single
missing essential feature in the port information. At
On Fri, May 14, 2004 at 12:01:01PM -0400, Paul Davis wrote:
I don't mind *IFF* the metadata file has a simple, human readable
syntax (no XML please) that can be parsed line by line.
no XML, and yes, parsable line by line, and yes, human readable. *but*
the plan should be to use the supplied
On Fri, May 14, 2004 at 04:53:53PM +0100, Steve Harris wrote:
Yes, but xrm misses most of the desirable feaures of metadata languages
(agreed semantics, extensibility and so on). We could just use the syntax,
but its pretty complex for non-X11 apps that want to parse it.
Actually, it's very
On Fri, May 14, 2004 at 11:31:01AM -0500, Jack O'Quin wrote:
I'm having trouble figuring out Fons' original point here, though I'm
sure he has one. Simple and human readable are worthwhile goals, but
hard to reconcile.
Strange.. I'd think these two would go hand in hand...
Whit 'simple' and
On Fri, May 14, 2004 at 01:15:37PM -0400, Dave Robillard wrote:
On Fri, 2004-05-14 at 13:07, Fons Adriaensen wrote:
no XML, and yes, parsable line by line, and yes, human readable. *but*
the plan should be to use the supplied library to get and set
values. nobody should be doing
On Fri, May 14, 2004 at 01:19:11PM -0400, Paul Davis wrote:
On Fri, May 14, 2004 at 12:01:01PM -0400, Paul Davis wrote:
I don't mind *IFF* the metadata file has a simple, human readable
syntax (no XML please) that can be parsed line by line.
no XML, and yes, parsable line by line, and
The first release of Aeolus is now available at
http://users.skynet.be/solaris/linuxaudio
Enjoy !
--
Fons
On Fri, May 14, 2004 at 08:10:32PM +0100, Steve Harris wrote:
On Fri, May 14, 2004 at 07:07:59 +0200, Fons Adriaensen wrote:
On Fri, May 14, 2004 at 12:01:01PM -0400, Paul Davis wrote:
I don't mind *IFF* the metadata file has a simple, human readable
syntax (no XML please) that can
On Fri, May 14, 2004 at 04:03:55PM -0400, Dave Robillard wrote:
On Fri, 2004-05-14 at 14:09, Fons Adriaensen wrote:
I suppose the question is _why_ would you fiercely resist this good
design practise?
System interfaces are often defined by an API (and even that is
questionable since
On Fri, May 14, 2004 at 10:50:31PM +0100, Steve Harris wrote:
I dont really see the avantage of this - control and description are
seperate tasks, and not even closely related.
Very closely related if you consider that the ultimate purpose of a
description is to control something. Why should
On Sat, May 15, 2004 at 07:10:30AM +0200, Marcus Andersson wrote:
Interesting interpretation. This means that the mapping between the
slider and the parameter will be
f(x) = k*a^x
with k=f(0) and a = f(1) if the slider goes form 0 to 1.
You probably meant a = f(1)/f(0).
This also
On Sun, May 16, 2004 at 09:14:46AM +0200, Marcus Andersson wrote:
This also means that it is illegal to include 0 in the parameter range.
In that case you can use f(x) = x*a^x, with a = f(1).
How do I invert this function? I am stuck.
You need the inverse only to set the slider to a
On Wed, May 19, 2004 at 06:46:18AM +1000, Erik de Castro Lopo wrote:
Once libsndfile gets Ogg Vorbis and Speex support, FLAC support is
also high on the list. Since libsndfile accepts WAV-EX (stupid fscking
microsoft idea) it will be able to transcode WAV-EX - FLAC without
a problem.
On Thu, May 20, 2004 at 08:00:31AM +0200, Jens M Andreasen wrote:
I went to ambisonic and read the FAQ. I do not agree with them when they
say that 2-channel stereo is only good for imaging between the speakers.
I do not see such a statement in the FAQ.
It is possible by using phase
On Thu, May 20, 2004 at 07:07:52AM -0400, Paul Davis wrote:
this will get you started:
http://www.soundonsound.com/sos/Aug01/articles/surroundsound1.asp?session=dec5645986353a3e68c8439720360f53
Hi Paul, interesting pointer, it's in my bookmarks now ! Thanks !
--
Fons
Hi Joern,
not superior, but equivalent, less snake-oil-infested, cheaper, more
general and more elegant. ahem, yes, superior :-D
you can express any spatial sound with just 4 channels, where one is
the mono component, and 3 are the x, y, and z-axis difference
signals (similar to m/s
On Fri, May 28, 2004 at 12:38:03PM -0700, Fernando Pablo Lopez-Lezcano wrote:
Hmm, it would be a fun project then to come up with a profiler of various
audio cards by recording and then capturing a specific buffer of audio data.
Then by comparing them (assuming that this drift is constant)
On Tue, Jun 08, 2004 at 01:54:58PM +0200, Marek Peteraj wrote:
Fons' Moog HP filter is a complex piece of DSP i suspect.
No, it's actually quite simple :-) The most complex one is
the four-band parametric filter I released recently, and
that's also the only one that is not intended as an AMS
On Thu, Jun 10, 2004 at 12:26:07AM +0200, Marek Peteraj wrote:
I've never seen such inapt community btw, which is totally ignorant in
organizing itself. See the gnome community which started to exist the
same year. They have more conferences per year, one of them being
huge(guadec) with
New releases of Aeolus and Jaaa are now available at
http://users.skynet.be/solaris/linuxaudio
Aeolus-0.2.0
- bugfixes,
- some new stops,
- added tuning and temperament controls,
- added controls for tremulant speed and intensity.
Still no manual :-( but it's coming...
This
Already some bugfixes to the things I announced less than 24 hours ago...
jaaa-0.1.1.tar.bz2
clthreads-0.0.3.tar.bz2
as usual to be found on http://users.skynet.be/solaris/linuxaudio
Thanks to Jesse Chappell for pointing out the problems.
--
FA
Hi Fernando,
First of all, if you package Aeolus and Jaaa please use the most recent
versions that I released only yesterday evening (libclthreads-0.0.3
and jaaa-0.1.1).
I noticed something different. I used to be able to start
Aeolus and click on [Next] and then I would get a preset (you
On Thu, Jun 17, 2004 at 07:04:42PM +0200, Dr. Matthias Nagorni wrote:
On Thu, 17 Jun 2004, Alfons Adriaensen wrote:
- I have the same SL 9.0 and ALSA version (but other soundcards)
- The ALSA code is a near copy of the ALSA code in JACK, in
both cases memory mapped access is used.
On Thu, Jun 17, 2004 at 08:26:51PM +0100, Chris Cannam wrote:
One way to do this is to use pthread conditions. Have a ring buffer
between your decoder and RT threads, with the decoder sleeping for
short periods of time between reads using pthread_cond_timedwait or
similar. This causes
around the bug you reported I suggested -d hw:0.0
This should be -d hw:0 (tested, this works). I also found the
cause of the problem, it's not in Aeolus but in one of the shared
libraries.
Kind regards,
--
Fons Adriaensen
On Fri, Jun 25, 2004 at 06:54:20PM +0200, Thorsten Wilms wrote:
Requiring the user to read documentation to learn about functionality
he would not even expect is not an option.
Have education levels gone down *that* far ?
--
FA
On Fri, Jun 25, 2004 at 06:15:24PM -0400, Pete Bessman wrote:
I have a very simple request for everybody who loathes
plug-and-drool usability: show me the tunes. That's all. Lemme
hear the avant garde music enabled by avant garde interfaces
The most avant-garde music is enabled by very dull
On Fri, Jun 25, 2004 at 03:38:10PM -0400, Lee Revell wrote:
On Fri, 2004-06-25 at 12:00, Fons Adriaensen wrote:
On Fri, Jun 25, 2004 at 06:54:20PM +0200, Thorsten Wilms wrote:
Requiring the user to read documentation to learn about functionality
he would not even expect
On Fri, Jun 25, 2004 at 08:29:44PM +0200, Thorsten Wilms wrote:
On Fri, Jun 25, 2004 at 06:00:42PM +0200, Fons Adriaensen wrote:
On Fri, Jun 25, 2004 at 06:54:20PM +0200, Thorsten Wilms wrote:
Requiring the user to read documentation to learn about functionality
he would not even
On Fri, Jun 25, 2004 at 07:55:50PM -0400, Pete Bessman wrote:
At Fri, 25 Jun 2004 23:28:35 +0200,
Fons Adriaensen wrote:
so that I can compare it against the mouth-breathing crow-magnon
music created with shiny-quarter interfaces. I'm sure the results
will speak for themselves
On Sat, Jun 26, 2004 at 01:33:59AM -0400, Lee Revell wrote:
Designing for usability is not rocket science. For the phone example,
the options (in decreasing order of desirability) are:
1. A self-explanatory pictorial representation.
2. A text label.
3. An
On Sat, Jun 26, 2004 at 01:41:17AM +0100, Dave Griffiths wrote:
Having worked professionally on related things, I just can't stand the
I HAVE to understand everything about an interface in 5 SECONDS!
attitude to gui design. People can learn things, it's part of playing
music on real
On Sat, Jun 26, 2004 at 11:50:29AM -0400, Pete Bessman wrote:
Great, well, I made the observation that the intelligentsia have
microscopic genitalia. (What, you want my data? Surely you jest.)
Ergo, the smarter a person claims to be, the greater the magnification
they require at the urinal.
On Sat, Jun 26, 2004 at 09:58:19AM +0200, Thorsten Wilms wrote:
With regards to widgets, I stated that requiring the user
to read documetation in order to use a widget is not an
option. It was especialy about visualizing / hinting at
functionality. The fan-sliders without the fan graphics
On Mon, Jun 28, 2004 at 08:47:48AM +0200, Jens M Andreasen wrote:
Actually, in my house we speak latin only when we feel the urge to make
fun of people in badly need of a good argument. Out of house, of course,
we speak latin to make people feel stupid and stop argueing with us.
I don't live
On Sun, Jun 27, 2004 at 11:03:15PM -0400, Pete Bessman wrote:
That's a straw man. The original point was something to the effect of
a volume knob which can only be operated after studying a manual is
an indication that the UI designer is a failure, although my
rendition is probably more
On Tue, Jul 13, 2004 at 11:37:12PM +0100, Martijn Sipkema wrote:
IMHO it is the lack of a mutex implementation with priority ceiling
or inheritance and the stories about relying on either being a design
problem that have caused the Linux audio community to not use
mutexes and declare them
Hello LAD,
New releases of aeolus, jaaa, and the required shared libs are
now available at http://users.skynet.be/solaris/linuxaudio.
Aeolus:
* The AEOLUS_DIR environment variable is no longer used. There is now
a command line option -S directory to select the stops directory.
This
On Sun, Aug 15, 2004 at 01:58:49PM +0200, Benno Senoner wrote:
If you absolutly have to have multiple machines doing i/o then you will
need some complicated resampling stuff. Fons has been working on it, to
allow soft-sync between 2 jack systems, but I've not tried it yet.
Between a JACK
On Sun, Oct 03, 2004 at 09:56:24AM -0400, Dave Phillips wrote:
I'm hoping that you're thinking of a realtime display, in which the
peaks roll off to create a true waterfall effect.
Hi Dave,
I've been thinking of adding such a mode to JAAA. How do you think it
should look ?
1. For each new
Hello,
Is there any way to stop qjackctl from re-ordering a client's ports
in its connection dialog ? IMHO the author of a client probably had
her/his good reasons for the order chosen, and qjackctl should leave
this alone.
--
FA
On Sat, Oct 09, 2004 at 06:48:29PM +0200, Florian Schmidt wrote:
On Sat, 9 Oct 2004 15:13:16 +0200
Fons Adriaensen [EMAIL PROTECTED] wrote:
Is there any way to stop qjackctl from re-ordering a client's ports
in its connection dialog ? IMHO the author of a client probably had
her/his good
On Sat, Oct 09, 2004 at 10:03:11PM +0100, Rui Nuno Capela wrote:
Florian's suggestion makes sense, of naming the ports like something in
the lines of:
out_1L
out_1R
out_2L
out_2R
I tried a number of different schemes, and here are the results:
(creation order 1L, 1R, 2L, 2R, )
-
On Sun, Oct 24, 2004 at 11:12:36AM +, [EMAIL PROTECTED] wrote:
what is ams ? just try to search about that with no luck...
See http://alsamodular.sourceforge.net.
AMS is probably your best bet. If you want the ultimate in control
(but a much much steeper learning curve), Supercollider is
On Mon, Nov 08, 2004 at 10:47:34PM +0300, Andrew Gaydenko wrote:
The aim is to make measurement mini-laboratory, in particular, to measure
distortions of audio amplifier. Using JACK, I can route line ins/outs, find
clean sine signal and get amplified one, and then route last to... Well,
the
On Thu, Nov 18, 2004 at 06:56:12PM -, Rui Nuno Capela wrote:
Why don't you just start with the SUSE supplied kernel-sources and config?
You'll find all the necessary stuff on the distro CDs/DVD. I know 'coz I
do a also run SUSE 9.2 Pro around here :)
Did you apply any patches ? Having 9.2
Hello LAD,
New releases of Aeolus and JAAA are available at the usual place :
users.skynet.be/solaris/linuxaudio.
From the Aeolus-0.3.1 README :
* Added 'instability'. Each pipe is individually phase modulated
in order to emulate the random fluctuations in a real one. This
provides
On Mon, Nov 22, 2004 at 01:21:07PM -0500, Lee Revell wrote:
Yes, my thoughts exactly. You don't even have to use the ALSA sequencer
API, you can use the ALSA timer API directly. You get multiple timer
sources (system, RTC, sound card). It should also be more portable.
Yes, but using the
On Mon, Nov 22, 2004 at 07:00:59PM -0500, Paul Davis wrote:
how far ahead can you queue without getting into trouble when the user
does realtime edits?
think about this for long enough, and i think you will come to same
conclusion: deliver events in a process-callback-style fashion, more
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