On 01/06/2011 01:26 AM, Krzysztof Foltman wrote:
That's always a good idea anyway - especially now I'm preparing for an
official release of the new version (with good-looking GUIs and new
plugins, all by Markus Schmidt). Any extra testing may help making the
release more stable and useful.
On 12/22/2010 08:37 PM, i...@vt.edu wrote:
Hi all,
I've been battling a kind of a dsp-writer's-block as of late. Namely, I am
dealing with a project where (at least as of right now) I would like to
explore
human whisper and its percussive/rhythmic power.
spectral delay (freqtweak), if
On 12/20/2010 09:14 AM, torbenh wrote:
torben, i wonder: will alsa_[in|out] be perfectly bit-transparent when
the two interfaces it bridges are externally synced, or will its
tracking algorithm keep oscillating around the correct samplerate?
it will oscilate, but the oscilation will get
On 12/20/2010 07:29 PM, Paul Davis wrote:
2010/12/20 Jörn Nettingsmeier netti...@folkwang-hochschule.de:
there is a complication: loudness is no identical to maximum sample
value, but the relationship is good enough for government work, so to
speak.
no, it's not. there is absolutely
On 12/16/2010 08:49 PM, Lee Azzarello wrote:
On Thu, Dec 16, 2010 at 1:36 PM, Lee Azzarello l...@rockingtiger.com wrote:
On Thu, Dec 16, 2010 at 6:58 AM, torbenh torb...@gmx.de wrote:
On Thu, Dec 16, 2010 at 08:02:29AM +0100, Luis Garrido wrote:
On Thu, Dec 16, 2010 at 4:21 AM, Lee Azzarello
On 12/15/2010 11:14 AM, gene heskett wrote:
Ralf I suspect, if he were to use pgp, would be like me, and only trust
pgp-2.6.2a, the last one before they put Zimmerman in jail for a few years.
I have often said, and have been called the uber paranoid for it, that one
of the conditions of
On 11/30/2010 02:57 PM, Paul Davis wrote:
On Tue, Nov 30, 2010 at 4:59 AM, David Griffiths d...@pawfal.org wrote:
Hi all,
I don't know if this is a known issue or something fixed in recent
kernels, but I recently found a fix for a problem I was having with my
harddisk spinning down in live
On 11/11/2010 05:58 PM, Camilo Polymeris wrote:
I understand that a lot of you develop for free software and are passionate
at what you do. But how do you pay the bills? What do you do for a living?
Are you a student? Do you do software development just as a hobby, or do
you want to make a
On 11/04/2010 06:01 AM, Jens M Andreasen made my day with this hilarious
sig:
-- jedes mal wenn du eine quintparallele verwendest
tötet bach ein kätzchen.
http://www.youtube.com/watch?v=43RdmmNaGfQ
(each time you employ parallel fifths, bach kills a kitten)
but since you give that
On 11/04/2010 01:18 PM, Victor Lazzarini wrote:
Not to mention Jazz musicians, arrangers, Debussy, Ravel, Stravinsky...
Sounds like a teutonic idea of Musik (sorry Joern...), rather than
anything else.
of course it is. in strict counterpoint, parallel perfect consonances
are streng verboten,
On 10/31/2010 08:51 PM, Rory Filer wrote:
I've actually heard _some_ recognizable audio - I could tell it was
the song I had selected, but it sounded very distorted (sounded
horrible) and I'm certain it wasn't because of a weak FM transmitter
signal. I'm guessing it was a mismatch in the
On 10/31/2010 08:51 PM, Rory Filer wrote:
I've actually heard _some_ recognizable audio - I could tell it was
the song I had selected, but it sounded very distorted (sounded
horrible) and I'm certain it wasn't because of a weak FM transmitter
signal. I'm guessing it was a mismatch in the
On 10/19/2010 11:20 AM, Matt Henley wrote:
I have found the best way to build ingen is to grab the entire svn tree
( instructions at http://drobilla.net/blog/software/ingen/ )and do a top
level make and make install. Installing the programs piecemeal seemed
to cause problems.
that would be
On 09/16/2010 02:18 AM, Niels Mayer wrote:
This looks like a better mouse-knob interactor, e.g. use the mouse for
pointing/selecting, and then use this knob for simultaneously
controlling multiple parameters on the selected channel at once.
On 08/04/2010 08:11 AM, Fernando Lopez-Lezcano wrote:
Unrelated question, does ambdec do third order vertical? Will it?
(I'm going to have zenith and nadir speakers in the Listening Room so it
would be able to do 3rd order h and v).
me too! wouldn't even need a gui to create them, but it
On 08/03/2010 07:45 PM, Fernando Lopez-Lezcano wrote:
Hmmm, so, you send WXYZ to the four subs and whatever order you can
decode to the regular speakers (6, 8, etc). Sounds good.
it does :) except you'd usually throw away z, unless you wanted to try a
tetrahedral setup for the subs (for
On 07/28/2010 02:04 PM, Dominic Sacré wrote:
2010/7/28 Jörn Nettingsmeiernetti...@folkwang-hochschule.de:
On 07/27/2010 10:45 PM, f...@kokkinizita.net wrote:
Real detents require force feedback. A mouse doesn't
provide that. This GTK thing is completely broken.
i wonder how you'd do it on a
On 08/03/2010 09:44 PM, Fernando Lopez-Lezcano wrote:
I currently have 8 Mackie 824's at ear level, 4 624's in the ceiling and
4 624's below the floor (and I will add two more 624's, one zenith, one
nadir). So their frequency responses don't exactly match in the low
end.
in that case, you
On 08/03/2010 09:38 PM, f...@kokkinizita.net wrote:
A separte VLF decoder is one of the features I want in the next
generation Ambdec. The question I'm asking myself is how much
'configurability' this requires.
as far as i'm concerned, it would be cool to have some help creating a
hybrid
On 07/31/2010 11:14 PM, Bearcat M. Şandor wrote:
Thank you. I think i understand all that, but let me take this apart to
make sure. What you're saying is that having full range speakers only
effects the playback quality of the music not the ambisonics and
that ambisonics itself does not
On 07/31/2010 04:40 AM, Bearcat M. Şandor wrote:
Folks,
I'm a recovering audiophile.
:-]
I have read that speakers in an ambisonic set up should be full range.
I'd like to set up a ambisonic speaker system (8 channel to start), and
the prospect of 8 full range channels is daunting. Since
john,
On 07/29/2010 02:35 PM, JohnLM wrote:
On 2010.07.28. 22:06, f...@kokkinizita.net wrote:
Compared to conventional 5.1 pairwise panning the result will be
more even, without emphasising the speaker locations as would be
the case otherwise. In other words, the sound will be much less
seem
On 07/27/2010 10:45 PM, f...@kokkinizita.net wrote:
Real detents require force feedback. A mouse doesn't
provide that. This GTK thing is completely broken.
mmmh. lawo mixing desks, anyone? i just got to play with one a few days
ago, and you can ask the motorfaders to act up around 0dB, which
On 07/22/2010 11:44 PM, f...@kokkinizita.net wrote:
Extrapolating a bit, that is one of the reasons why an
unamplified singer in an opera theatre can have a dramatic
effect that is much stronger than someone yelling into a
microphone and being amplified to 130 dB SPL. By which I
don't want to
On 07/24/2010 10:45 AM, Ralf Mardorf wrote:
On Fri, 2010-07-23 at 10:02 -0700, Fernando Lopez-Lezcano wrote:
On Fri, 2010-07-23 at 08:51 +0200, Ralf Mardorf wrote:
On Fri, 2010-07-23 at 01:09 +0200, Jörn Nettingsmeier wrote:
one thing that often gets overlooked: people have learned to accept
On 07/22/2010 05:02 PM, f...@kokkinizita.net wrote:
Such a 'virtual stereo mic' is part of Tetraproc, and
there's also a Ladspa plugin doing this. The latter has
some problems in Ardour as it has 4 ins and 2 outs, and
Ardour get confused by this and will (IIRC) copy inputs
3 and 4 to the
On 07/22/2010 08:15 PM, Gene Heskett wrote:
1. How are the signals brought into phase such that electronically, all mic
ribbons or diaphragms seem to occupy the same space, just facing in
different directions?
well, you can't do that :)
two approaches:
if you only care for horizontal surround
On 07/22/2010 08:42 PM, Ralf Mardorf wrote:
As an ape (of course I'm an ape like every human is an ape) and troll (I
don't see myself as a troll) I suspect phasing too, that's why I
overstated argued with the next generation Cochlea-Implant, or needles
in the brain.
that is a bogus statement.
On 06/09/2010 03:19 PM, Brett McCoy wrote:
Both a plugin version and a standalone app would be awesome!
a plugin, a standalone version, and an emacs extension with lots of
chewy parentheses! puh-leeeze!
;-]
___
Linux-audio-dev mailing list
On 07/21/2010 08:56 PM, JohnLM wrote:
If I code program to handle attenuation of sounds depending on their
source (emitter) position in virtual 3D space, I guess then there's no
simple way to relate the effect to real world.
How this is usually handled?
you reduce the level with distance
On 07/17/2010 10:46 PM, f...@kokkinizita.net wrote:
What I don't understand is how the contacts got so dirty.
If a resistance of a few kOhm is enough to make it look
as a closed contact then it can't be handling large currents,
so there should not be any arcing. And the construction of
the
hi calf folks, hi everyone!
unless i'm getting confused, the calf plugin set used to contain a set
of hi- and lowpass filters that seem to have vanished with my last git
pull. which is a major pity, since a) the current git solves a number of
crashes and b) i've been using those hi- and lowpass
On 06/17/2010 04:41 PM, Simon Burton wrote:
Does anyone have a recommendation for a more comprehensive (non-free even)
piano sound palette ?
pianoteq comes to mind. haven't used it myself, but many people whose
ears i trust have been quite enthusiastic about it. it's closed and
pricey, but
On 06/17/2010 07:37 PM, Jeremy wrote:
Now, assuming a 60 Hz refresh rate (which is what I believe my monitor has),
and a buffer size of 1024 pixels, that would be a throughput of 61440
samples per second. More than the 48000 sampling rate. So it seems that
drawing every sample *is* a
On 06/16/2010 10:50 PM, f...@kokkinizita.net wrote:
On Wed, Jun 16, 2010 at 06:41:02PM +0100, Rui Nuno Capela wrote:
ok. fixed (qjackcl 0.3.6.29+)
Works nicely, thanks !
the uber-procrastinator is committing at relativistic speeds, as always.
rui, you're awesome!
On 05/23/2010 11:20 AM, Erik de Castro Lopo wrote:
f...@kokkinizita.net wrote:
On Sun, May 23, 2010 at 08:52:16AM +1000, Erik de Castro Lopo wrote:
I really don't think any modification is necessary.
At least one: the sample rate value in the header must
remain the same.
hi *!
i need to re-synchronize two recordings of the same event that for
technical reasons had to be done with unsynchronized clocks. i'm
assuming for the sake of sanity that both clocks were perfectly stable.
my approach is this: in ardour, align some unique feature (a click in
this case) at
On 05/22/2010 12:38 PM, Erik de Castro Lopo wrote:
Jörn Nettingsmeier wrote:
unfortunately, both zita-resampler and libsamplerate seem to represent
the sample rate ratio internally as a fraction of two integers,
I haven't looked at the code for zita-resampler but for libsamplerate,
what
On 05/22/2010 12:38 PM, Erik de Castro Lopo wrote:
Jörn Nettingsmeier wrote:
unfortunately, both zita-resampler and libsamplerate seem to represent
the sample rate ratio internally as a fraction of two integers,
I haven't looked at the code for zita-resampler but for libsamplerate,
what
hi *!
in the light of recent timer discussions (was it here or on
jack-devel?), lwn.net has interesting coverage about the time stamp
counter and its oddities in the recent weekly edition:
http://lwn.net/SubscriberLink/388188/62e8027425e224f6/
(free link, the rest of the issue will become
On 05/17/2010 03:22 PM, Louigi Verona wrote:
Hey guys!
In many DAWs it is not possible to use a vocoder, because it requires
two inputs - a modulator and a carrier. So in order for this to work you
either have to allow routing in the mixer (which none of the linux daws
are capable of, as far as
hi *!
the lac2010 presentation recordings are now available at
http://www.linuxproaudio.org/lac2010/ - kudos to faberman for
very-close-to-realtime post-production!
let me take the opportunity to thank all stream team people (many of
them members of the linux video community, who put in
On 04/26/2010 08:47 AM, Louigi Verona wrote:
Hey guys!
I was wondering about the following.
On Windows we have lots and lots of plugins and synthesizers and effect
racks. On Linux the selection is much less variable.
However, am I correct in understanding that the variety of the Windows
On 04/26/2010 09:08 PM, Louigi Verona wrote:
Jorn! Thanks, very informative answer.
What can you say about stuff like this:
1. vocoder
2. grnulizer
3. slicer (when a file is sliced into pieces)
4. beat matching
i hate to admit it, but i don't have the slightest idea about any of
those.
On 04/16/2010 08:06 AM, michael noble wrote:
hi folks,
I just saw an interesting line over at opensuse.org (
http://news.opensuse.org/2010/04/14/opensuse-11-3-milestone-5-the-community-strikes-back/)
regarding the installation of JACK2 as default in the upcoming opensuse 11.3
release. That
On 03/20/2010 04:08 AM, Jesse Chappell wrote:
In the meantime I really will try to update FT soon.
cheers!
freqtweak is way too cool to retire.
___
Linux-audio-dev mailing list
Linux-audio-dev@lists.linuxaudio.org
On 03/19/2010 12:10 AM, Erik de Castro Lopo wrote:
Reuben Martin wrote:
FTutils.hpp needs a #includestdint.h added to it. It's to do
with changes made in gcc 4.4 to do with headers.
That wasn't the problem I ran into. My problem was that sig++/sig++.h
wasn't found even though I have
On 03/08/2010 08:52 PM, David García Garzón wrote:
The CLAM project[1] is delighted to announce the long awaited 1.4.0 release of
the C++ framework for Audio and Music, code name '3D molluscs in the space'.
[1] http://clam-project.org
i'm getting a 403 on http://clam-project.org/download/.
On 02/24/2010 12:19 PM, alex stone wrote:
2. use a virtual stereo mike to pick out a stereo perspective. it allows
for more control, is a lot easier to produce, and it's probably the
safer option if you want your mix to be reliable on consumer gear...
2.) Virtual stereo mike? I don't
On 02/24/2010 03:29 PM, f...@kokkinizita.net wrote:
On Wed, Feb 24, 2010 at 05:01:20PM +0300, alex stone wrote:
Jorn, Fons,
I'm getting deeper into a setup now, and trialling a few different configs.
A further question.
Given that an amb sphere is equal on all sides, and the sweet spot
is
hi alex, fons!
On 02/23/2010 02:37 PM, alex stone wrote:
Jorn, Fons, i'm looking for a ladspa UHJ encoder, and can't seem to
find one. Any idea if such a beast exists? Or if there's a standalone
instance or ambdec preset i can use, and route in and out of?
as fons said, jconv does a nice job
On 02/21/2010 11:03 AM, alex stone wrote:
Once upon a time, jack_diplomat was alive and well. Sadly, all
attempts to find it have failed, as the hosting site seems to be,
extinct.
Does anyone have a copy of this app on their machine that they might
consider sharing?
never used it, but
On 02/20/2010 08:57 PM, Tim E. Real wrote:
Stretchers are playing a very important role in my designs now.
Even just using tempo maps requires them with audio tracks, if the
user wants to change a tempo AFTER recording an audio track.
frankly, i haven't heard any stretchers that weren't
On 02/20/2010 10:01 PM, Tim E. Real wrote:
Found a new one Zita-resampler. Can't wait to try it.
http://www.kokkinizita.net/linuxaudio
zita is a resampler. it can't do time-stretching independent of pitch.
___
Linux-audio-dev mailing list
On 02/19/2010 05:40 PM, Paul Davis wrote:
On Fri, Feb 19, 2010 at 11:30 AM, Simon Jenkins
sjenk...@blueyonder.co.uk wrote:
I'm reading CV input as invitation to modulate and, yes,
sometimes it makes no sense to modulate at the full audio rate, but
sometimes it does. I'm just not sure a
On 02/18/2010 10:54 AM, alex stone wrote:
As a power user who's modestly (just kidding) keen on saving time,
using great workflow, and avoiding as much of the drudgery of editing
work over and over again to get an end result as is possible, i've had
the privilege and pleasure of testing and
On 02/18/2010 12:18 PM, alex stone wrote:
I will clarify here that i'm talking about a user experience, before
the discussion gets into jousting with white papers.
that's what i was interested in, too. can you describe the advantages of
the non-mixer approach (which i haven't tried yet) to,
On 02/11/2010 05:28 PM, Ralf Mardorf wrote:
but I'm just a stupid user spreading FUD, so simply
ignore my experiences.
jeez, just shut up.
say, do you also feel an urge to reply to obituaries and commercial ads
in your paper with letters to the editor?
it's just plain stupid to discuss your
On 02/07/2010 01:06 AM, nescivi wrote:
On Friday 05 February 2010 18:37:26 Jörn Nettingsmeier wrote:
rashif, marije,
On 02/05/2010 03:45 PM, nescivi wrote:
On Monday 01 February 2010 18:34:46 Jörn Nettingsmeier wrote:
since the runlevel corresponds with the need for a ll kernel, i wonder
rashif, marije,
On 02/05/2010 03:45 PM, nescivi wrote:
On Monday 01 February 2010 18:34:46 Jörn Nettingsmeier wrote:
since the runlevel corresponds with the need for a ll kernel, i wonder:
is there any way to tell the kernel (via grub) to tell init to ignore
the initdefault in /etc/inittab
On 02/04/2010 05:12 PM, Adrian Knoth wrote:
Like all stand-alone jack tools, it's usability hell.
VocProc also connects to physical ins/out at startup.
Next: VocProc changes the volume.
So plenty of room for improvements. ;)
easy, man. it's the first public release, and it sez
hi everyone!
here's a question for sysadmin type low-latency adepts:
i have a general-purpose notebook that doubles as a lean and mean
recording machine. it's opensuse, which means it works nicely with most
bells and whistles, but it also has an awful lot of questionable stuff
running that
On 01/25/2010 11:59 PM, Gabriel M. Beddingfield wrote:
From 'man 3 random_placebo':
If rand(), random(), and random_r() aren't random
enough for you, random_placebo() is generates
more randomer numbers by algorthmically generating
more entropies on your system. The
On 01/27/2010 08:39 PM, alex stone wrote:
It's been a good day, and i've enjoyed the stability. I used jack1
before, because it gave me fewer challenges, xruns and occasional pops
and spits, than jack2, which i ended up having to ease out to
48000/512/3, to reach near the same performance.
On 01/27/2010 09:11 PM, Stéphane Letz wrote:
Le 27 janv. 2010 à 21:02, Jörn Nettingsmeier a écrit :
On 01/27/2010 08:39 PM, alex stone wrote:
It's been a good day, and i've enjoyed the stability. I used jack1
before, because it gave me fewer challenges, xruns and occasional pops
and spits
sorry about the previous bogus message, i fat-fingered the rewrap
button... :(
On 01/27/2010 09:11 PM, Stéphane Letz wrote:
Le 27 janv. 2010 à 21:02, Jörn Nettingsmeier a écrit :
i didn't think too much about jack2's apparent overhead, since it
has the benefit of scaling to smp, which
On 01/27/2010 09:28 PM, f...@kokkinizita.net wrote:
On Wed, Jan 27, 2010 at 09:02:39PM +0100, Jörn Nettingsmeier wrote:
but i'm
generally able to use much lower latencies down to 64 (or 128 in the
jack2 case), unless i use jconvolver, which forces me to go to at least
1024 so as not to max
hi aqualung people!
your player has been discussed as an interesting way to pave the path of
ambisonics towards world dominance, because it's cross-platform, hooks
up to jack and allows for easy integration of ladspa plugins (where we
could slip in an ambi decoder to make an easy-to-use ambi
f...@kokkinizita.net wrote:
On Mon, Dec 07, 2009 at 11:06:23PM +0100, Jens M Andreasen wrote:
On Mon, 2009-12-07 at 20:41 +0100, Karl Hammar wrote:
So, 24bit, 48/96kHz is the spec. to aim at?
If you happen to sit on a warehouse full of them, otherwise 192kHz is
priced the same these days.
On 12/04/2009 06:15 AM, Ivica Ico Bukvic wrote:
Dear fellow FOSS enthusiasts,
This is probably already old news according to Internet standards but as it
turns out we spent a good time this evening in class not knowing that we got
slashdotted. For those interested in belatedly joining the
On 12/01/2009 07:31 PM, Karl Hammar wrote:
The aim is to provide upto 8 audio i/o ports.
I don't think we should set an upper limit, instead it would be
interesting to see how many channels the system could support.
I would say we try to make (input and output)
one desent channel, e.g.
Steve Harris wrote:
On 5 Nov 2009, at 23:33, David Robillard wrote:
On Thu, 2009-11-05 at 22:21 +0100, f...@kokkinizita.net wrote:
On the LV2 website, Lars Luthman's UI extension is described as:
This extension is written for revision 2 of the LV2 specification
and is NOT compatible with
might be of interest to some LAD patrons:
Original Message
Subject: [Sursound] Call for Participation: Int. Symposium on
Ambisonics and Spherical Acoustics, 2010
Date: Wed, 4 Nov 2009 12:57:25 +0100
From: Markus Noisternig markus.noister...@ircam.fr
Reply-To: Surround Sound
hi *!
thanks to contributions from a number of ffado folks) (notably stefan
richter, kernel firewire developer, the irq priorities howto in the
ffado wiki at
http://subversion.ffado.org/wiki/IrqPriorities
should be in a usable shape.
it's split in two parts: the first one shows how to tweak
hi everyone!
i'm playing with my shiny new BCF2K, and i'm going to use it some
distance from my machine, so i'm going to try a midi link instead of USB.
what is the maximum length of midi chain that you have used without
problems? i read somewhere that no more than 15 meters are recommended,
David Robillard wrote:
On Sun, 2009-11-01 at 01:11 +0100, Jörn Nettingsmeier wrote:
as a standard, lv2 is ages behind ladspa,
Core LV2 is more powerful than LADSPA. In addition, there's other
features you can use. Saying this is ages behind LADSPA makes no sense
whatsoever.
dave
Arnold Krille wrote:
Hi,
On Sunday 01 November 2009 16:47:44 Jörn Nettingsmeier wrote:
i'm playing with my shiny new BCF2K, and i'm going to use it some
distance from my machine, so i'm going to try a midi link instead of USB.
what is the maximum length of midi chain that you have used
David Robillard wrote:
many, many things
dave, if you'd really read the my post in its entirety and not jumped at
the one provocative catchphrase, you might have noticed that we're
pretty much on the same side. heck, i've even begun to dig into larsl's
lv2 tutorial.
to re-iterate what i had
On 10/26/2009 07:33 AM, Drip Stone wrote:
Hi everybody,
I have a question regarding amplifying a PCM frame. For each frame, I
get a float number, and multiply this number by 2, and then output
frames to alsa one by one using function snd_pcm_writei. But the wav
file sounds the same to me,
On 10/26/2009 11:38 AM, Victor Lazzarini wrote:
Well if the audio was clipped, there would probably be some audible
distortion.
Perhaps if a code fragment was posted we could give a more informed
opinion.
i was thinking of operating on files, where clamping can be done with
perfect fidelity.
[paul, i assume this was meant for the list?]
Paul Davis wrote:
2009/10/22 Jörn Nettingsmeier netti...@folkwang-hochschule.de:
hi everyone!
this:
A feature of BFS is that it detects when an application tries to obtain a
realtime policy (SCHED_RR or SCHED_FIFO) and the caller does
http://lwn.net/SubscriberLink/357800/0fb8fb6f0975403c/
interesting coverage as usual, thanks to lwn for the free-link feature.
___
Linux-audio-dev mailing list
Linux-audio-dev@lists.linuxaudio.org
hi everyone!
is the rme hdspe raydat supported under linux?
if so, any raydat users here with success or horror stories?
if not, is there an alternative that has at least 3 adat i/o,
preferrably 4 and uses pci express?
i know of (and like) the rme 9652 cards, but i'd rather not buy pci
cards
Ivica Ico Bukvic wrote:
Here's a story that may give you all a good chuckle:
Last weekend Robin Gareus, our LAO guru has contacted me inquiring why there
was a ~1hr linuxaudio.org server downtime that took place that Sat. morning.
Luckily we now have an UPS that gives us almost 2 hours of
might be of interest:
http://lwn.net/SubscriberLink/354690/ae66a6a912e14fca/
(this is a free link to content which would normally be subscribers only
for the next week or so - thanks to lwn.net for this service!)
___
Linux-audio-dev mailing list
Frank Neumann wrote:
Hi,
[this is mostly addressed to Linux Audio friends from Germany's Southwest]
from November 15th-16th, 2009, the German SoftwareSupport Media Group is
hosting the OpenSourceExpo in Karlsruhe, Germany. I am not exactly sure
yet about its target audience, but it appears
jens, i assume you meant this to go to the list?
Jens M Andreasen wrote:
On Sun, 2009-09-27 at 14:53 +0200, Jörn Nettingsmeier wrote:
those who are following the development of the xiph CELT low latency
codec might find this presentation interesting:
From the overheads:
- expf() is even
Ralf Mardorf wrote:
Peter Nelson wrote:
For release velocity it's the same as for poly pressure, I don't know
any keyboard providing this. The only unusual function provided by a
keyboard I know is breath control. My Yamaha DX7 supports breath
control.
There are some; my CME UF8
those who are following the development of the xiph CELT low latency
codec might find this presentation interesting:
http://www.celt-codec.org/presentations/
torben has implemented CELT support in netjack1, and this video helped
me a lot in understanding how it actually works.
http://www.jpbouza.com.ar/ESP2/tutoriales/gnulinux/blenderardour/id/en
___
Linux-audio-dev mailing list
Linux-audio-dev@lists.linuxaudio.org
http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Nedko Arnaudov wrote:
Julien Claassen jul...@c-lab.de writes:
Hello Nedko!
Now I've come up against a wall, which I don't want to climb.
It's all to do with GUI-toolkits and their relatives. Especially
flowcanvas. I dearly hope and plea for the feature, to optionalise the
GUI-parts.
David Robillard wrote:
On Sat, 2009-08-15 at 01:04 +0200, Fons Adriaensen wrote:
On Fri, Aug 14, 2009 at 05:44:21PM -0400, David Robillard wrote:
Another outcome of the Graz meetings is that the port groups
that are currently named H#V# should really be named H#P#.
The H#V# ones do also
David Robillard wrote:
On Mon, 2009-08-10 at 11:02 -0400, David Robillard wrote:
http://lv2plug.in/ns/dev/port-groups.lv2/port-groups.html
P.S. Comments on this from anyone with experience in multi-channel stuff
would be appreciated if there's any problems.
It's being used to support
David Robillard wrote:
On Sun, 2009-08-16 at 18:27 +0200, Jörn Nettingsmeier wrote:
David Robillard wrote:
On Mon, 2009-08-10 at 11:02 -0400, David Robillard wrote:
http://lv2plug.in/ns/dev/port-groups.lv2/port-groups.html
P.S. Comments on this from anyone with experience in multi-channel
David Robillard wrote:
Is there any existing 'standard' for the order of channels for
higher-than-stereo multi-channel streams?
I'd assume this would be defined for interleaving or something, though
that's not what it's needed for: I think the port groups extension
should specify an order
David Robillard wrote:
On Fri, 2009-08-14 at 22:54 +0200, Jörn Nettingsmeier wrote:
most favoured normalization scheme is sn3d, i.e. plugins will have to
deal with inputs greater than 1.0f.
Hm. I never considered normalization... the reasoning and details
behind this seem to be pretty
David Robillard wrote:
Building this logic on top of a set of optional
extensions would seem to be a nightmare, and that is and always
has been my main objection to this approach.
FUD, nothing more. It would be no more of a nightmare than implementing
it on top of... well, anything else;
standard ianal disclaimer applies.
at the risk of starting another zillion-mail thread, here's how i
understand the gpl to work under german laws (which should be almost the
same in most other countries):
a) there is a difference between the license and the copyright. what
many people fail to
hi guys!
lwn.net has a very nice article on the progress of -rt in the latest .31
kernel:
http://lwn.net/SubscriberLink/345076/aab59b866d6f169d/
(this is otherwise subscribers-only coverage, brought to you by the
free link feature - if you have some bucks to spare, check out lwn and
consider
way over my head, but it would give me a warm fuzzy feeling if some
hotshot linux audio people were attending:
http://www.osadl.org/RTLWS-2009.rtlws-2009.0.html
best,
jörn
___
Linux-audio-dev mailing list
Linux-audio-dev@lists.linuxaudio.org
101 - 200 of 237 matches
Mail list logo
