New releases and updates at http://www.kokkinizita.net/linuxaudio
Aliki-0.0.3 Impulse response measurement.
- Many bug fixes, should be a bit more stable now...
- Added flexible export options.
- Manual updated.
Jace-0.0.4Low-weight convolution engine for JACK and ALSA.
- Now
On Tue, Mar 27, 2007 at 04:17:16PM +0200, Frank Barknecht wrote:
Maybe one day there will be a Linux version of Live, but it's
not something I particularily look forward to, as I wouldn't
use it anyways unless it gets opensource'd.
There are probably many of us thinking the same way.
But
On Tue, Mar 27, 2007 at 10:27:11PM +0200, Frank Barknecht wrote:
Linux didn't stay an amateuer platform in other areas, why should
free software not be professionally used in the audio world as well?
There is no good reason why free software shouldn't be used, but not
everything required in
On Thu, Mar 15, 2007 at 02:00:46PM +0100, Joern Nettingsmeier wrote:
unless i'm very much mistaken, UHJ-encoded material can be played back
on any stereo rig without problems (some even report better stereo width
and precision).
Correct.
the only drawback i heard about is that
On Wed, Mar 14, 2007 at 11:16:46AM -0400, Lee Revell wrote:
With binary drivers kernel debugging requires the cooperation of the
vendor in the best case, and lots of guesswork and reverse engineering
in the worst case.
I'd say _driver_ debugging requires the cooperation of the vendor.
You can
On Wed, Mar 14, 2007 at 06:32:14PM +0100, Christian Schoenebeck wrote:
I think most of the people on this list know these kind of issues. And I
totally agree that this is an argument to avoid using binary drivers, but
it's definitely NOT a sufficient argument to completely reject a BDI.
I
On Wed, Mar 14, 2007 at 02:26:35PM -0400, Lee Revell wrote:
The interface does not change that fast.
Indeed it doesn't, and that is quite normal - after so many years
it should be quite clear to both kernel and driver developers what
constitutes a good interface. One more reason to define and
On Mon, Mar 12, 2007 at 12:53:28AM +0300, Andrew Gaydenko wrote:
Will anybody find a minute ot two to explain me how does the plugin
work - I mean a user POV rather technical realization details.
(Assuming you mean my plugin from the MCP package)
This is an emulation of an analog phase delay
New releases of JAAA and JAPA and the libraries they depend on
are now available at
http://www.kokkinizita.net/linuxaudio/downloads
jaaa-0.4.1: bugfixes.
japa-0.2.0: bugfixes, white and pink noise generators now built-in.
clalsadrv-1.2.1: bugfixes. This version should now work correctly
Hi Stefano,
I still don't know if I can get there...
I got really disappointed to see that there's no easy way to get to
Berlin via train from Turin, I'm considering bus, and I need someone
else to come with me too... :-(
You could try to get in contact with Daniele Torelli [EMAIL
On Sun, Dec 31, 2006 at 01:10:54PM +0100, Alberto Botti wrote:
Il giorno dom, 31/12/2006 alle 02.25 +0100, Christoph Eckert ha scritto:
Congrats for the new job! Hope you'll enjoy yourself in Italy. :) See
you in Berlin.
of even better in Italy - will there be a LAD-party?!?
Why
Hello all,
First of all, my best wishes for 2007 to all Linux Audio Developers !
2007 will be a special year for me. As some of you already know, I said goodbey
at Alcatel Space two months ago, and starting 8 Jan 2007 I'll be working at LAE
- Laboratorio di Acustica ed Elettroacustica -
On Mon, Dec 04, 2006 at 04:05:37PM -0500, Ivica Ico Bukvic wrote:
Because PCMCIA != internal soundcard (even if it is physically inserted into
the laptop's body). Namely, it is enclosed in its own metal casing, not to
mention PCMCIA enclosure both of which allow for a much better R/F shielding
Hello all,
Since I'm preparing to move abroad, I've transferred my website
to a host that's independent of my local ISP. The new site is
www.kokkinizita.net/linuxaudio
'Kokkini Zita' means 'red zeta' - the Greek letter, and all
the 'i' are pronounced as English 'ee', not 'y'.
It's also my
A first release of Yass is now avaiable at
users.skynet.be/solaris/linuxaudio.
Yass is a 'scrolling scope' jack client. It has been
hanging around in prototype form for some time. This
is a first beta release.
Main features:
- up to 32 channels,
- variable scrolling speed,
-
On Wed, Nov 22, 2006 at 08:14:11PM +0100, Dominic Sacré wrote:
Got one problem though: high CPU load (I'm not kidding). At small window
sizes everything's fine, but when it exceeds a certain size (a little
less than fullscreen on my machine), CPU load immediately goes up from
~5% to
On Wed, Nov 08, 2006 at 12:20:42AM +, James Courtier-Dutton wrote:
The current jackd skips a step in the processing of the poll events.
Looking at the code it seems already quite elaborate.
Basically what happens comes down to (ignoring error
checking and timeouts):
- The set of pollfd is
On Wed, Nov 08, 2006 at 02:24:30PM +0100, Fons Adriaensen wrote:
Is snd_pcm_poll_descriptors_revents() more than an
accessor ? If it is, the name is a quite misleading.
To answer my own question, it seems that it *is* more
than an accessor.
The docs leave one thing unclear. Does this call
On Wed, Nov 08, 2006 at 05:58:40PM +0100, Clemens Ladisch wrote:
In that case, how can one test if *all* pollfd for a given
pcm are ready ?
You cannot. The state of the file descriptors is not necessarily
related to the state of the PCM device (which is why this function
exists).
OK,
On Mon, Oct 30, 2006 at 01:30:53PM -0500, Lee Revell wrote:
On Mon, 2006-10-30 at 18:52 +0100, Fons Adriaensen wrote:
The real question is how to fit this into the existing architecture:
- hardware presents itself as 2 * 96 kHz
- user wants to see a device with 4 * 48 kHz
On Fri, Nov 03, 2006 at 04:39:18PM +, Simon Jenkins wrote:
On Fri, 2006-11-03 at 08:53 +0100, Fons Adriaensen wrote:
[...]
I'd say that the essential feature of JACK is not that it is a
callback based system, but that it presents and expects audio
data in fixed size blocks
On Fri, Nov 03, 2006 at 06:37:13PM +1100, Loki Davison wrote:
On 11/3/06, Jens M Andreasen [EMAIL PROTECTED] wrote:
On Fri, 2006-11-03 at 13:42 +1100, Loki Davison wrote:
mmm. I think they are missing the point about ALSA vs OSS api here. It
doesn't matter. The only one who should care
For the brave: an alpha release of Aliki (Room Impulse
Response Measurement) is now available on:
http://users.skynet.be/solaris/linuxaudio
along with a manual that should get you started.
This is basically the code used at the LAC2006
workshop, cleaned up a but.
As said, ALPHA, incomplete,
On Wed, Nov 01, 2006 at 09:48:37AM -0600, Ben Loftis wrote:
Can you please tell us more about Aliki?
I am using Denis Sbragion's DRC program to generate an impulse response
file, and his suite of graphing tools to generate various views of the
measurement.
What is the output of Aliki?
On Mon, Oct 30, 2006 at 11:41:36AM -0500, Lee Revell wrote:
On Mon, 2006-10-30 at 17:35 +0100, Alfons Adriaensen wrote:
What they want to do is trick hardware 2-ch recorders into accepting
the 4-ch signal. So what is in reality 4 channels at 48 kHz will be
presented as 2 channels at 96
On Mon, Oct 30, 2006 at 02:55:33PM -0500, Paul Davis wrote:
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
Hello all,
Core Sound http://www.core-sound.com/default.php willsoon
be offering a tetrahedral (Ambisonic) microphone at a very
reasonable price. They are also working on a combined preamp
+ AD converter unit for this mic. This will be able to multiplex
the 4 channels over a single SPDIF link, by
On Fri, Oct 20, 2006 at 03:51:09PM +0100, Dave Griffiths wrote:
The tricky bit is of course getting a flash that's totally synchronous
with the beep. Absolute synchrony is not achievable without dedicated
hardware, but we need to get an approximation that's within the few ms
range.
I
On Fri, Oct 20, 2006 at 10:44:48PM +0200, Tim Goetze wrote:
Back in the 80s, the humble Commodore 64 could be readily programmed
to fire an interrupt on vertical sync. Have 20 years of progress
really deprived us of this fine feature, or is it just missing from X?
Same on the humble BBC
On Wed, Oct 18, 2006 at 06:50:39PM +1000, Erik de Castro Lopo wrote:
...
Its very easy to get carried away with trying to reach some sort
of audio perfection. Things like upsampling in order to apply
compression is over-engineering.
I'd agree. This is a non-problem for a well-designed and
On Wed, Oct 18, 2006 at 12:14:45PM +0100, John Rigg wrote:
The fact remains that a lot of high end professional users consider many
of the free software plugins to be nearly unusable (see Ben Loftis'
earlier post in this thread). This isn't intended as a criticism of the
developers, just an
On Mon, Oct 16, 2006 at 07:48:35PM +0100, Dan Mills wrote:
The gain control signal has energy right the way out
to the band limit (and probably aliased around it),
never mind what happens when that hits the multiplier!
The question is: how much of this HF energy is there ?
There shouldn't be
On Tue, Oct 17, 2006 at 09:59:10AM -0400, Paul Davis wrote:
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
On Tue, Oct 17, 2006 at 06:50:04PM +0200, David Olofson wrote:
On Tuesday 17 October 2006 16:43, Fons Adriaensen wrote:
It could be. OTOH, most DACs today would upsample and filter before
the real conversion takes place, and could allow for this. But maybe
they don't, and just clip
On Thu, Oct 05, 2006 at 05:12:20PM +0100, Steve Harris wrote:
The SC* plugins do the same as TAP (calculate the gain every 4 samples),
but I interpolate the gain values between each computation. The
attch/deacay times were slow enough in my testing that it was OK to do
that.
It should be OK
On Thu, Aug 31, 2006 at 11:13:52AM -0400, Gene Heskett wrote:
Looking at the subject line prompts me to ask if you have it (your brain)
backed up on tape somewhere? Most of us have claimed that at one point or
another...
To help Paul find back his brain, we should all burn candles,
chant
On Sat, Aug 26, 2006 at 12:06:16AM +0200, David Olofson wrote:
What really annoys me about these is that they're usually written to
give the impression of a personal message from someone who would know
what you're doing and what kind of social network you have. Sometimes
you can't really
(Since this was rejected on LAA ('no reason given'), I'll answer
the question here.)
On Mon, Aug 14, 2006 at 09:20:36PM +0400, Andrew Gaydenko wrote:
What do Kokini Zita words mean? I have thought, Fons Adriaensen is the
author
of the listed apps :-)
It should be Kokkini (with 2 k's
On Thu, Jul 27, 2006 at 08:46:27PM +0200, Jay Vaughan wrote:
There are public-domain RTOSes available that are suitable for this
task. To those, you can add drivers for USB and FAT32. Without an
RTOS to give you hard real-time scheduling, you have no chance to
achieve the
On Thu, Jul 27, 2006 at 06:58:32AM +1000, Erik de Castro Lopo wrote:
Yes, that was my idea. So if the sndfile.hh has:
class Sndile
{
int method (/* params */) ;
}
int method (/* params */)
{
/* whatever */
}
do I need to add an inline
Hello all,
I just updated the small set of Ambisionics related LADSPA plugins
available at users.skynet.be/solaris/linuxaudio.
The three decoders now feature optional phase aligned shelf filters
and independent control for LF and HF gain of the velocity components
(from in-phase over max rE to
On Thu, Jul 13, 2006 at 09:56:58PM +0400, Andrew Gaydenko wrote:
I mean some minimal C++ class set like: WavFile, WavHeader, WavFrame with
few apparent methods (open/close, read/write frame(s)).
Libsndsfile is plain C, but will do what you want without any fuss.
You could write a WAV
On Sat, Jul 08, 2006 at 01:34:44PM +0100, James Courtier-Dutton wrote:
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.
Hello all,
I finally took the big step and handed in my resignation at
my employer, Alcatel Alenia Space.
After 3 years of CAD, 3 years of real-time kernels, and 11
years in space telecoms, I want to return to my first love
which is audio, acoustics, and music. My activities in LAD
have
On Sun, Jul 02, 2006 at 10:49:54AM +0200, Thorsten Wilms wrote:
To me your behaviour of accusing Dave of plain lying is not
acceptable. You seem to implicate dishonesty.
I someone states as a fact and without any qualification a
certain interpretation of a text, while knowing very well that
On Sat, Jul 01, 2006 at 08:22:16AM +0200, Jens M Andreasen wrote:
Linear attack sounds OK. Given the exponential way we perceive volume,
this *is* the desired function.
That's the rationale for having exponential volume controls. In the case
of a release profile, it's rather because many real
On Sat, Jul 01, 2006 at 04:09:42PM -0400, Dave Robillard wrote:
Whether or not you agree with the licensing practise, calling it open
source is as misleading as calling MS shared source open source.
Defend the license/exception if you want, but don't intentionally
mislead people about the
On Sun, Jun 25, 2006 at 06:57:47PM +0100, Steve Harris wrote:
I agree that describing it as volts is a bit odd, but it instantly makes
people like me feel at home. There's not reason why a digital modular neds
units for its modulation sources. It's just real numbers.
I never mentioned
On Sat, Jun 24, 2006 at 07:15:38PM +0400, Andrew Gaydenko wrote:
And the question is: what is common way to smooth the result? Some offtopic
apps has
something like 1/24, ..., 1/3 octave smoothing. What does it mean?
Lowpass (linear phase) filtering of the response. Or windowing the
IR which
On Mon, Jun 19, 2006 at 01:49:14PM -0400, Dave Robillard wrote:
On Mon, 2006-06-19 at 14:15 +0200, Alfons Adriaensen wrote:
All of them fake.
Fake like the countless bug reports I get about your filter plugins not
working because they take some silly arbitrary unit instead of Hz for
On Mon, Jun 19, 2006 at 10:34:05PM +0100, Steve Harris wrote:
FWIW, I think the not changing any code thing is a blind, someone,
somewhere has to change some code if you want new behaviour*. To me the
critical thing is not that, but that a display function or whatever only
solves half the
On Mon, Jun 19, 2006 at 11:25:52PM +0100, Steve Harris wrote:
On Mon, Jun 19, 2006 at 11:58:43PM +0200, Fons Adriaensen wrote:
What worries me is that LV2 is *not* going to solve the problem that
DR raised w.r.t. my Moog filter plugins.
This particualr one I'm not worried about, as it's
On Wed, Jun 07, 2006 at 08:49:38AM -0400, Paul Davis wrote:
nice to hear that they are faster. on the other hand, once again POSIX
screws us all over by not integrating everything into a single blocking
wait call. i've said it before, i'll say it again - this is one of the
few things that the
On Wed, Jun 07, 2006 at 05:42:26PM -0400, Lee Revell wrote:
But, from the original post it seems that pthread_cond_signal is not
realtime safe as it locks a mutex:
...
How can glibc guarantee that we are not put to sleep if there is
contention?
The mutex associated with a CV is held only
On Tue, May 30, 2006 at 06:07:15PM +0100, Steve Harris wrote:
On Tue, May 30, 2006 at 11:43:57AM -0400, Dave Robillard wrote:
char* type = lv2_port_get_type(someplug, 0);
if (!strcmp(type, LV2_DATATYPE_FLOAT))
/* ... */
free(type);
Makes sense to me. You could make the API
On Tue, May 30, 2006 at 03:10:38PM -0400, Dave Robillard wrote:
I'd consider any interface that just returns a constant and requires
a malloc() and a free() to do it plain broken. This data doesn't live
in kernel space, or does it ? You could just return a const char *.
It's not a
On Tue, May 30, 2006 at 05:48:35PM -0400, Dave Robillard wrote:
The function we're talking about pulls this info directly from the data
file (not eg from a loaded Port object which would have a const type
string). The library doesn't load all the stuff from the file into
memory (in which
On Mon, May 15, 2006 at 11:31:00PM +0200, Christoph Eckert wrote:
recording... :) .
+ oggenc + ftp + mail LAD ??? :-) :-) :-)
--
FA
Follie! Follie! Delirio vano e' questo!
Hello all,
The long announced new release of Aeolus is finally available.
Version 0.6.6 is almost a complete rewrite of the previous
official release, 0.3.1 (a lot happened in bewteen).
This should still be considered a beta release - no doubt
some nasty bugs will be uncovered when this version
On Tue, May 02, 2006 at 12:15:20PM -0400, Paul Davis wrote:
saying that the port range is exponential doesn't pin it down very much.
it still requires the host to make decisions about precisely what kind
of exponential curve to use for the range, and it may get it wrong.
It does pin it down
On Tue, May 02, 2006 at 05:21:44PM +0100, Steve Harris wrote:
this goes from 0Hz to fs/2Hz, and I want it to be logarithmic,
That's a contradiction.
--
FA
Follie! Follie! Delirio vano e' questo!
On Sat, Apr 22, 2006 at 02:26:57PM +0200, Thorsten Wilms wrote:
Distribution / finding plugins:
Stability:
Control/audio rate:
Port grouping:
Port Roles:
Referencing:
Hints:
Presets:
Help / Discription:
MIDI/OSC
GUI lib:
To which I'd add:
Polyphony/Multiple channels:
Plugin
On Sat, Apr 22, 2006 at 03:01:26PM +0200, Lars Luthman wrote:
...
support for polyphony (you can run several plugin instances as a
polyphony group with a single call to run_multiple() which lets you do
common calculations once).
That's not the point. Even in that case each plugin instance
On Wed, Apr 19, 2006 at 05:39:01PM +0400, Dmitry Baikov wrote:
Moscow
St.Petersurg
--
Helsinki
Stockholm
Berlin
Koeln
Brussels
London
Paris
Bourdeaux
Madrid
Barcelona,Figeras
Genova
Milan?
Rome
Venice
Vienna
Prague
--
Moscow
If you'd come to Antwerp (45 km from Brussels and
A maybe silly question: where on a typical system are the rdf
descriptions of ladspa plugins supposed to live ?
I can't find them !
--
FA
Follie! Follie! Delirio vano e' questo!
A second release of the Ambisonics plugins is now available.
* Added cube (8-speaker) decoder.
* Removed conflicting port hints.
http://users.skynet.be/solaris/linuxaudio
--
FA
Follie! Follie! Delirio vano e' questo!
On Wed, Apr 12, 2006 at 09:45:42PM +0200, Esben Stien wrote:
Actually, it varies between 206.765 and 206.766
That's about 20 nanoseconds difference, or less than
the delay of 10 meters of cable...
--
FA
Follie! Follie! Delirio vano e' questo!
On Sat, Apr 08, 2006 at 05:17:14PM +0100, James Courtier-Dutton wrote:
Hi,
Are there any linux tools out there that will sample the line in, and
display a detailed spectrum scope of the detected sound?
E.g.
http://www.pcavtech.com/soundcards/ct462048/SNR_LB_FS.gif
Eventually I wish to
On Sat, Apr 08, 2006 at 08:09:49PM +0100, James Courtier-Dutton wrote:
Neither of those applications work.
I have ardour working fine with stereo in/stereo out.
jaaa -J seems to talk to jack, but does not capture or play any sound.
jaaa -A fails to even open the alsa device.
Have these
On Thu, Apr 06, 2006 at 08:28:32PM +0200, Jan Weil wrote:
206.805
This is a Thinkpad T43 with an unpatched 2.6.16 with CONFIG_PREEMPT=y,
jackd 0.100.7, [EMAIL PROTECTED]
Now what does that value tell me? These are samples, I suppose, i. e.
4.308 msec?
Yep, it's samples.
Fons, how do
On Thu, Apr 06, 2006 at 02:57:28PM -0400, Lee Revell wrote:
On Thu, 2006-04-06 at 20:52 +0200, fons adriaensen wrote:
Have you been able to use -p 64 with 2.6.16 on the Thinkpad with
ACPI ? I had no problems with it on my R51 when using 2.6.8 (SuSE
9.2), but since 2.6.13 (SuSE 10.0) I have
On Sat, Mar 18, 2006 at 12:02:40PM -0300, Denis Alessandro Altoe Falqueto wrote:
My /etc/hosts was like this:
127.0.0.1 localhost.localdomain localhost
I changed it to:
127.0.0.1 bach bach
And it all worked. I found the solution in the hexter homepage, at the
On Mon, Mar 13, 2006 at 10:21:39PM +0100, [EMAIL PROTECTED] wrote:
On Sun, Mar 12, 2006 at 03:50:26PM -0500, Lee Revell wrote:
Why do you use big-endian on the wire, requiring a double swap for x86
- x86? Wouldn't LE make more sense, especially as PPC Macs become
unavailable?
well i
On Mon, Mar 13, 2006 at 05:25:04PM -0500, Paul Davis wrote:
On Mon, 2006-03-13 at 23:10 +0100, fons adriaensen wrote:
Is it true on the common platforms that using ntohl and htonl on
floats will always result in compatible data on the wire or in a
file ? In other words, are floats byte
On Mon, Mar 13, 2006 at 11:59:15PM +0100, stefan kersten wrote:
as paul stated, network byte order is defined to be
big-endian, so yes, you have to convert 32 bit floats (and
doubles, for that matter) on intel, because they are stored
lsb first. of course it would be perfectly valid for
On Sat, Mar 04, 2006 at 05:06:49PM -0500, Lee Revell wrote:
Where can I find a decent map of Germany? The whole country is blank on
Google Maps...
Try http://www.de.map24.com/
--
FA
On Mon, Feb 27, 2006 at 01:16:17AM +0200, Jussi Laako wrote:
On Sun, 2006-02-26 at 20:38 +0100, Albert Graef wrote:
Canvases give you much more than just rendering. They also manage the
graphical objects that you created and, if anything changes, rerendering
the changed parts happens
On Mon, Feb 20, 2006 at 11:01:10PM +0100, David Kastrup wrote:
Lee Revell [EMAIL PROTECTED] writes:
By this logic, locking my doors is immoral because it diminishes
people's freedom to roam around my house.
Those people have not paid for access to your house. Purchasers of
proprietary
On Mon, Feb 20, 2006 at 11:41:43PM +0100, David Kastrup wrote:
They have paid for a license to use it, and for nothing else.
Well, then they might have some expectation to be able to use it, no?
Without the ability to adapt the software to different devices or
applications, or fix errors
On Mon, Jan 30, 2006 at 08:05:42PM +, James Courtier-Dutton wrote:
It is due to the fact that the alsa programming in jackd has been
implemented wrongly. The poll revents are not handled correctly.
This results is OK operation when using hw:0,0, but likely to fail for
more exotic alsa
On Tue, Jan 24, 2006 at 09:02:23AM +1100, Erik de Castro Lopo wrote:
fons adriaensen wrote:
What I need in particular is some way to calibrate the time
axis - i.e. to say frame #N corresponds to t = 0, and some
other similar info, mostly sample indices.
There is no existing chunk type
Hi all,
is there a recommended way to write / read additional chunks in
WAV files, using libsndfile (assuming it's possible at all - I
didn't find any hints to this in the docs) ?
What I need in particular is some way to calibrate the time
axis - i.e. to say frame #N corresponds to t = 0, and
JACE is a Convolution Engine for JACK and ALSA, using FFT-based
partitioned convolution with uniform partition sizes.
I wrote it mainly as a 'proof of concept' for something more
complicated, to be announced at the next LAC. But it could be
useful as it is, hence this release.
Main features:
-
On Sat, Jan 07, 2006 at 04:45:21PM +0100, [EMAIL PROTECTED] wrote:
why dont you open a separate display connection for the plugin ?
then you can even move the gui updates to a different thread and there
you go...
look into gtkplug.c and gtksocket.c on how this works.
Does X allow multiple
On Sun, Jan 01, 2006 at 06:42:02PM +0100, Robert Jonsson wrote:
Indeed, there are several issues at work here...
In anycase, MusE has from 0.7.2pre2 a fix that enables synths with identical
names to be used.
Also, in the case with ZynAdd, another option is to use only one instance.
ZynAdd
On Sat, Dec 31, 2005 at 08:03:06AM -0500, Paul Davis wrote:
On Sat, 2005-12-31 at 00:04 +0100, fons adriaensen wrote:
1. If things have to be timed accurately, it seem logical to concentrate
this activity at one point. At least then the timing will be consistent,
you can impose priority
On Fri, Dec 30, 2005 at 11:54:56AM -0500, Paul Davis wrote:
you don't know the correct priority to use. i imagine an api along the
lines of:
jack_create_thread (pthread_t*, void* (thread_function)(void*),
void* arg, int relative_to_jack);
the last
On Fri, Dec 30, 2005 at 05:10:44PM -0500, Paul Davis wrote:
On Fri, 2005-12-30 at 22:27 +0100, Pedro Lopez-Cabanillas wrote:
On Friday 30 December 2005 17:37, Werner Schweer wrote:
The ALSA seq api is from ancient time were no realtime threads were
available in linux. Only a kernel
On Fri, Dec 23, 2005 at 01:37:31AM +, James Courtier-Dutton wrote:
I have a question for some audio professionals out there.
What is the smallest sensible gain control step in dB.
Is it 0.5dB ?
I am asking, because if one is using a digital gain control in a 24bit
fixed point DSP, once
On Thu, Dec 15, 2005 at 04:56:04PM +0100, Asbjørn Sæbø wrote:
Some colleauges of mine do need a tool for coding and decoding of
high-order Ambisonic for their research. They are aiming for seventh
order, played back over sixteen loudspeakers. They are now planning to
implement this, using
On Thu, Dec 15, 2005 at 09:04:28PM +0100, Georg Holzmann wrote:
Regardless of the JACK / LADSPA question, seventh order Ambisonics using
16 speakers is just ridiculous. Either it's horizontal only, and in that
case using 7th order is just a waste of resources and effort (3th order
will do all
On Sun, Nov 20, 2005 at 08:29:37AM -0500, Paul Davis wrote:
however, this is not necessarily the right approach to handling xruns.
its worth trying. a better start is to check if the xruns go away or
occur less frequently with a larger buffer size.
USB cards often work a lot better (allow
On Sun, Nov 06, 2005 at 10:59:57AM +, James Courtier-Dutton wrote:
My question is really what should I do when the gain_multiplier is 0.0
Do I:
a) Limit the range of the gain control to 0dB to -40 dB and have a
separate Mute control.
b) When the gain control has a gain_multiplier of
On Sun, Nov 06, 2005 at 12:21:30PM +, James Courtier-Dutton wrote:
The problem I have is what should I
display for the 0.0 gain_multiplier setting. I.e. When it effectively
mutes the sound output at it's minimal slider setting.
Off ???
--
FA
On Sat, Nov 05, 2005 at 09:23:05PM +0200, Juhana Sadeharju wrote:
Thanks for the tip on diversity reception. Yes, I got the
idea from astronomer's systems.
Normally 'diversity reception' means to combine the signals from
2 or more receivers to obtain a result that has a better S/N ratio
(or
On Tue, Nov 01, 2005 at 09:24:58PM +0200, Juhana Sadeharju wrote:
How about a software which can combine the outputs coming from
two receivers tuned to the same station?
...
Strange, I recently read a recent (2000+) paper. They praised
(like having a patent on them) two innovations: (1) an
On Sun, Oct 30, 2005 at 01:53:48PM +0100, Florian Schmidt wrote:
Oh i thought i read somewhere that when pthread_cond_wait it is not
guaranteed that anyone actually signalled. Will do some more reading.
It can return on unix signals, so you have to test for EINTR.
I don't think it will wake up
On Mon, Oct 31, 2005 at 01:44:45AM +0100, Florian Schmidt wrote:
Btw: i just discovered that pthread mutexes and condvars can have a
process shared flag which makes it possiblo to synchronize threads
across processes as it seems. Could be useful for jack, no?
pthread_condvar_setpshared()
On Sat, Oct 29, 2005 at 03:10:50PM +0300, Jussi Laako wrote:
On Wed, 2005-10-26 at 02:41 +0200, fons adriaensen wrote:
Filter 1: F = 50 Hz, A = 9
Filter 2: F = 2120 Hz, A = 1
and add the two outputs.
From quality point of view, at least I would recommend using IIR filters
On Sat, Oct 29, 2005 at 03:10:50PM +0300, Jussi Laako wrote:
From quality point of view, at least I would recommend using IIR filters
for this...
Please ignore my previous post - I misread 'FIR' where you wrote 'IIR',
and that explains it all...
--
FA
1 - 100 of 322 matches
Mail list logo
