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THANK YOU!
> http://news.harmony-central.com/Newp/2003/Final-Scratch-11-Traktor.html
>
> i poked around the website(s) but couldn't figure out if this means
> that TraktorFS runs on linux. someone with more time might be inclined
> to find out ...
Hi,
All you might want to know about Traktor FS, may be found at
http://www.nativeinstruments.de/index.php?id=traktorfs_us
Where they clearly state:
--8<-----------
Final Scratch on Linux and Mac OS X: Powered by TRAKTOR FS
-->8----------
BTW, I live next door to Native Instruments' office, and I heard and
saw that Traktor DJ stuff working on their release party in our
favourite low-tech cocktail bar, quite impressive, but those people
(~30) of NI actually forgot to pay their bill for about 4 weeks ;)
Alas, don't know what OS it was, but if something remains unclear, I
might jump in my pants and go ask them.
Martin
I would dearly love to see some Layla20 ALSA drivers for Linux. I have been
waiting for 3 years! I have no kernel hacking experience nor driver writing
experience.
Craig M. Moran
Booz|Allen|Hamilton
Advanced Technology Team
301-757-0616
Hallo,
in case nobody noticed it yet: The 06/2003 issue of the german
"Linux-Magazin" has a feature on our LAD-conference in Karlsruhe by
Stefan Klett.
Nice work, Stefan.
ciao
--
Frank Barknecht _ ______footils.org__
Hi,
Sorry if this is the wrong list. I have never found a list with people
who can help me with these kind of questions.
I am trying to figure out how a rectifier works in code. I have created
something like it using tanh() distribution ( which will turn a sine
wave into a sort-of-square wave). I can also achieve it with hard
clipping/saturation. In the same sense I'd also like to know how a
smooth operation works as well. I can not find any source code or
explanations to help me figure out the algorithm yet.
My other question is how are exponential signals generated? I want to
experiment with exponential FM. My only understanding of Exp. Fm is that
it sweeps half down and full up ( at 440hz, it would sweep 220hz down
and 880 hz up). I haven't much of an idea how this is achieved. How do I
convert a sine wave into an exponential signal for Exp FM?
all my interactions with sonic foundry were unpleasant, and i have no
idea what "rich media" means, but now we have another 800lb gorilla in
the tent, and another valuation of how much its worth to create a
program like Acid: US$18M.
http://news.harmony-central.com/News/2003/Sony-Sonic-Foundry.html
JACK 0.70.4 release
JACK is a low-latency audio server, written primarily for the GNU/Linux
operating system. It can connect a number of different applications to
an audio device, as well as allowing them to share audio between
themselves. Its clients can run in their own processes (ie. as normal
applications), or can they can run within the JACK server (ie. as a
"plugin").
JACK is different from other audio server efforts in that it has been
designed from the ground up to be suitable for professional audio work.
This means that it focuses on two key areas: synchronous execution of
all clients, and low latency operation.
CHANGES:
qjackconnect/jack_lsp bug fixed.
now can use either SysV or POSIX shm methods. Linxu 2.4.4 is required
for the POSIX method. Defaults to SysV, but can be switched with a
./configure option (--enable-posix-shm)
new --enable-optimize configure time flag, for compiler optimization.
jack_set_buffer_size_callback() de-deprecated.
added port metering API.
added option --timeout (OR -t) for client timeout. The default time is
as before: 500 msec.
jack_lsp now has usage (-h or --help) and version (--version) options.
various bug fixes and documentation updates.
CAVETS:
The shm code requires the tmpfs kernel module either loaded or compiled
in. A tmpfs filesystem must be mounted as well. This can be done by
adding:
tmpfs /dev/shm shm defaults 0 0
to /etc/fstab and running "mkdir /dev/shm; mount /dev/shm".
In versions of linux prior to 2.4.4, tmpfs was known as shmfs.
Client programs compile with the previous stable release (0.61.0) will
need to be recompiled to work with 0.70.4.
Client programs now need to also link against librt. That library
should be automatically included via pkg-config.
Taybin Rutkin
hello * !
frank neumann has sent me some photos he took during the lad meeting at
the zkm in march. they have been uploaded to the lad page and can be
viewed at
http://www.linuxdj.com/audio/lad/contrib/zkm_meeting_2003/photos/frank_n_ph…
have fun.
--
All Members shall refrain in their international relations from
the threat or use of force against the territorial integrity or
political independence of any state, or in any other manner
inconsistent with the Purposes of the United Nations.
-- Charter of the United Nations, Article 2.4
Jörn Nettingsmeier
Kurfürstenstr 49, 45138 Essen, Germany
http://spunk.dnsalias.org (my server)
http://www.linuxdj.com/audio/lad/ (Linux Audio Developers)
(Sorry for reposting to LAD, I've already sent this to alsa-user but I'm
unsure which list is more appropriate so I'm trying both)
Hello,
I have a machine with two DELTA 1010LT cards and they work very well
with ALSA (using ALSA binaries from Planet CCRMA).
I would like to write a single threaded application
that is able to write on all out channels (analog outs 1-8 of card 1 and 2 =
16 channels) simultaneously.
The card's manual says that in order to synchronize two cards one should
select the first card as master and set the internal clock generation
(I do this via envy24control) and then connecting the SPDIF-OUT of card 1
to the SPDIF-IN of card 2 and set the card 2's clock source to SPDIF-IN.
So far so good (from the hardware point of view)
I managed to write ALSA code that writes to a single card:
it opens the PCM hw:0 device selects 10 channel interleaved S32_LE mode
and then writes out some data. This works perfectly for a single card.
Now my question:
I would like to extend this app so that it writes to both cards simultaneously
without going out of sync.
I assume that the SPDIF writing and clocksouce selection provides this
from a hardware point of view, but I am unsure how to proceed from with
the application code since the ALSA docs do not mention such a scenario.
currently for a single card setup,
I do the usual card setup and then sit in a while loop that does
while(1) {
snd_pcm_writei(pcm_handle, audiobuf, MY_PERIOD_FRAMES);
}
Is the following way to extend the app to two cards correct ?
call the functions to setup card 1 (hw:0)
call the functions to setup card 2 (hw:1)
while(1) {
snd_pcm_writei(pcm_handle1, audiobuf1, MY_PERIOD_FRAMES);
snd_pcm_writei(pcm_handle2, audiobuf2, MY_PERIOD_FRAMES);
}
(where audiobuf1 is the buffer sent to card 1 while audiobuf2 is
the buffer sent to card 2)
Or is some special "sync start" procedure needed ?
I remember ALSA had/has provides some functionality in this regard
but I was unable to find documentation about it.
Thanks in advance for your help.
PS: does the DELTA cards only support the interleaved mode ?
The model seems a bit inefficient since in most apps the data
sources are non interleaved audio streams.
(eg a HDR app that outputs each audio track to a different audio channel)
cheers,
Benno
http://linuxsampler.sourceforge.net
-------------------------------------------------
This mail sent through http://www.gardena.net
Just a quick notification to you all that pawfal (the webste for arty
free software types) has been reborn as a wiki, and is open for all to
use and abuse. Feel free to add information of your own - we're
especially interested in people using linux and/or free software for
artistic purposes to put links to their work there, you can create your
own wiki page etc etc.
http://www.pawfal.org
For more information, if you are new to wiki's:
http://www.pawfal.org/index.php/PawfalFaq
cheers,
dave