I've just released BruteFIR 1.0
www.ludd.luth.se/~torger/brutefir.html
It is a high performance generic multi-channel audio convolver,
compatible with ALSA, OSS and JACK.
Apart from the OSS I/O module, there is not much new compared to the the
previous 0.99n release. Just some boring compile/portability error
fixes and stuff.
/Anders Torger
Hi all! (I posted this a while ago using a different account but it never went through so here it is again :-)
Two questions:
1) when using via82xx driver the sound using aplay sounds choppy, faster than usual, and has xruns (???) even when using a simple aplay command on a wavefile.
syslog suggests using something called dxs_support=1 or 4 but even if I put this into my /etc/modules.conf it still does nothing and syslog continues to suggest the same thing.
Does anyone know where am I supposed to put this statement in order for it to work? Please be as verbose as possible. Thanks!
The sound sounds fine when using "play" though and there are no xruns.
2) when trying to use jackd and via82xx as a regular user it always starts up does not report any errors but qjackctl complains that it could not connect to the jack and when starting it on command prompt it starts without any apparent errors but when trying to connect to it using for instance jack_metro it says "is jack running?" or something similar. Now, when I do the same as a root, it works like a charm. Any ideas why is this?
Any help on these issues would be greatly appreciated!
Best wishes,
Ico
Torben Hohn and I are pleased to release an initial version of libfst,
a small GPL'ed C library that provides support for using win32/x86 VST
plugins (FX and VST/i) within native Linux applications, with the
assistance of the Wine project's libwine.
==========================================================
We expect there will be several minor problems with this
initial release. Please help us fix them!
==========================================================
How is this different from VSTserver?
-------------------------------------
This work differs from Kjetil Mattheusen's VSTserver because the
plugin is loaded into the address space of the program using the
library, not into a server. As a result, it is a more appropriate
solution for audio applications that might support many VST plugins,
where the context switches required in the VSTserver case do not scale
well. The downside, of course, is that a misbehaving plugin will crash
the host application, which VSTserver avoids.
Why is it called FST?
---------------------
In case you wonder, "fst" stands for FreeST. Imagine saying "VST" with
an odd lisp.
Where do I get it?
------------------
The tarball is available from
http://linuxaudiosystems.com/fst/fst-1.5.tar.gz
How do I know it works?
-----------------------
Ardour and gAlan are already using this technology, and please read my
next announcement.
Dependencies
------------
The library is very small. You will need a recent version of Wine. The
configure step (which uses stuff built by Wine) takes much longer than
the compilation.
To build the library from source, you will need the Steinberg SDK
header files AEffect.h and aeffectx.h. These cannot be redistributed
by us or anyone else. The configure script will tell you where to get
them. The registration process with Steinberg points you to an FTP URL
that does not have the VST SDK on it. Use this instead:
ftp://ext2asio:sdk1ext@ftp.pinnaclesys.com/SDK
Please do NOT bypass the Steinberg registration process. It is
important to show them whatever level of interest the Linux world has
in VST.
--p
JACK RELEASE 0.98.0
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.
JACK is available at http://jackit.sf.net
***CHANGES***
Fixed bug when using non-dithered 16bit output.
Fixed crashing bug with JACK clients that use SSE.
three new functions in JACK API:
int jack_client_name_size(void);
int jack_port_name_size(void);
int jack_port_type_size(void);
These sizes are inclusive of the final NULL character.
Automatic server startup (more on this below).
Added OSS JACK driver.
New option -m,--no-mlock:
Do not attempt to lock memory, even if --realtime.
New option -p,--port-max n:
Set the maximum number of ports the JACK server can manage.
The default value is 128.
New option -T,--temporary:
jackd will exit when last client disconnects.
Configuration process reworked for better portability. This has
helped jack run on MacOSX and FreeBSD.
Added JACK thread initialization callback.
***AUTO START SERVER FUNCTIONALITY***
libjack will now try to automatically start jackd when jack_client_new()
is called if it isn't already running. Because this changes the
semantics of jack_client_new() and confuses certain apps, the new
semantics apply if and only if $JACK_START_SERVER is defined and
$JACK_NO_START_SERVER is not defined. This will change in future
releases.
libjack determines the proper arguments to pass to jackd by first
checking ~/.jackdrc, failing that /etc/jackd.conf, failing that
hardcoded strings that we've determined to be the most likely to work on
a variety of platforms.
The format of ~/.jackdrc and /etc/jackd.conf is as follows:
absolute path to the jackd or jackstart binary to be executed, followed
by the regular arguments all on one line. libjack will automatically
insert the --temporary argument so that any auto started jack server
will exit when the last client has disconnected.
If libjack is unable to start the server, jack_client_new() will fail
normally.
We hope that this new functionality will make using JACK easier and more
seamless with the JACK client.
Taybin
Should there be 3 seperate irc rooms then? One for each concurrent lecture?
Taybin
-----Original Message-----
From: Joern Nettingsmeier <nettings(a)folkwang-hochschule.de>
Sent: Apr 20, 2004 5:33 AM
To: Linux Audio Development Mailing List <linux-audio-dev(a)music.columbia.edu>,
Linux Audio User Mailing List <linux-audio-user(a)music.columbia.edu>
Cc: Frank Neumann <Frank.Neumann(a)st.com>, Goetz Dipper <goetz(a)zkm.de>,
"Dr. Matthias Nagorni" <mana(a)suse.de>
Subject: [linux-audio-user] laconf2 call for help w/ irc and streaming
hi everyone!
as you will certainly have heard, we are getting closer to the 2nd
linux audio conference at zkm (http://www.zkm.de/lad).
as you might also have heard, we will have live streams of all talks
and lectures in ogg format for those who can't make it there in person.
last year, we had an irc "feedback channel", meaning i would sit in
the lecture room with a notebook, answer questions from people who
were listening to the stream and relay their questions to the
audience if necessary. i want to do the same this year.
problem is: there are three lecture rooms now.
so i'm looking for a couple of people who'd be willing to help out.
specifically, i need people who
* are at the conference :) preferably the whole time
* bring their own laptop with working sound output and headphones
* have some experience with networking
* have or can operate a webcam on their machine and are willing to
install xawtv
* like to chat
your tasks will be:
* to attend talks
* to have an irc window open and answer and relay questions
* to set up a webcam that will provide the listeners with stills so
that they'll know which slide is up atm etc.
* to monitor the outgoing stream and to ping the relays from time to
time.
the more helpers, the less work for all of us. last year, it has
been great fun, although chatting, monitoring (with some 5 seconds
lag) and listening at the same time can make your head spin in funny
ways.
please cc: me privately on any replies. thanks.
regards,
jörn
--
Jörn Nettingsmeier
Lortzingstr. 11, 45128 Essen, Germany
http://spunk.dnsalias.org (my server)
http://www.linuxaudiodev.org (Linux Audio Developers)
Torben Hohn and I are pleased to announce the initial release of
jack_fst, a small JACK client designed to run VST FX and VST/i's with
connections to the rest of the JACK world, and, for VST/i's the ALSA
sequencer.
Tarball is available at:
http://linuxaudiosystems.com/fst/jack_fst-1.2.tar.gz
You will need the recently announced FST, a recent version of Wine,
GTK+2 and JACK (from CVS at this time).
Here is a list of VST plugins (actually) that are known to run
successfully with jack_fst:
Ambience.dll
AnechoicRoomSimulator.dll
BarsStripes.dll
Big Tick Hexaline.dll
BLOCKFISH.dll
Boss DS-1.dll
Boss SD-1.dll
Classic EQ.dll
Crystal.dll
Cyanide2.dll
DebaserDemo.dll
DeeLay_sm.dll
Delay Lama.dll
deloizer095.dll
DFX Transverb.dll
dominion v1.2.dll
Drumatic_22.dll
endorphin.dll
E-Phonic XPressor.dll
FLOORFISH.dll
Frohmage.dll
GoldenGate.dll
H2O.dll
JS Vibrato V1.0.dll
LoopaZoid.dll
mabento.dll
MadShifta.dll
mda DX10.dll
mda ePiano.dll
mda Piano.dll
MjMultibandCompressor.dll
MjRotoDelay.dll
ParisEQ.dll
relofter.dll
resolator_1.03_demo.dll
RetroDelay.dll
SIR.dll
sloper.dll
SoloString_v10.dll
SPITFISH.dll
Stretch & Squash.dll
SupaPhaser.dll
Syntar.dll
Tape Delay.dll
THD.dll
themodulator2.dll
Trancemitter.dll
Paul Davis:
>
> >Can you elaborate in terms of which version of wine you have used
> >successfully? (ie: wine = x.y.z or wine >= x.y.z?)
>
> i am using wine 20040309. i think torben has a slightly earlier
> version than this which has worked for him to the same extent.
>
This is the version I'm using with the vstserver now as well. It
seems to work better than the december 2003 version (finally!).
(Vsterver won't compile with it yet though.)
--
Paul Davis:
>
> Torben Hohn and I are pleased to release an initial version of libfst,
> a small GPL'ed C library that provides support for using win32/x86 VST
> plugins (FX and VST/i) within native Linux applications, with the
> assistance of the Wine project's libwine.
>
> ==========================================================
> We expect there will be several minor problems with this
> initial release. Please help us fix them!
> ==========================================================
>
> How is this different from VSTserver?
> -------------------------------------
>
> This work differs from Kjetil Mattheusen's VSTserver because the
> plugin is loaded into the address space of the program using the
> library, not into a server. As a result, it is a more appropriate
> solution for audio applications that might support many VST plugins,
> where the context switches required in the VSTserver case do not scale
> well. The downside, of course, is that a misbehaving plugin will crash
> the host application, which VSTserver avoids.
>
Another small point, I might add, is that the API against vstserver
is far far simpler, at least for adding native plugin-gfx support. But
regarding the performance and simplicity for the user, this solution is of
course better.
--
Hi all,
2 quick questions:
1) A while ago I heard that the JACK plug has been included into the ALSA's default array of plugs. Also today I found this on the Web (for the asoundrc):
pcm.jackplug {
type plug
slave { pcm "jack" }
}
pcm.jack {
type jack
playback_ports {
0 alsa_pcm:playback_1
1 alsa_pcm:playback_2
}
capture_ports {
0 alsa_pcm:capture_1
1 alsa_pcm:capture_2
}
}
Does this mean that now one can channel ALSA-only aware apps directly to JACK and if so, are there any penalties of doing it this way as oposed to using JACK-aware apps (i.e. Sample-sync?)?
2) Is there also a plug in ALSA that allows for non-interleaved cards (i.e. hdsp) to be directly talked to (i.e. using aplay) since by default aplay complains how the soundcard is not interleaved and hence it fails.
Apologies if these questions have been answered before. I did a bit of looking through the archives but was unable to find anything of relevance.
Many thanks for your help!
Best wishes,
Ico
hi everyone!
as you will certainly have heard, we are getting closer to the 2nd
linux audio conference at zkm (http://www.zkm.de/lad).
as you might also have heard, we will have live streams of all talks
and lectures in ogg format for those who can't make it there in person.
last year, we had an irc "feedback channel", meaning i would sit in
the lecture room with a notebook, answer questions from people who
were listening to the stream and relay their questions to the
audience if necessary. i want to do the same this year.
problem is: there are three lecture rooms now.
so i'm looking for a couple of people who'd be willing to help out.
specifically, i need people who
* are at the conference :) preferably the whole time
* bring their own laptop with working sound output and headphones
* have some experience with networking
* have or can operate a webcam on their machine and are willing to
install xawtv
* like to chat
your tasks will be:
* to attend talks
* to have an irc window open and answer and relay questions
* to set up a webcam that will provide the listeners with stills so
that they'll know which slide is up atm etc.
* to monitor the outgoing stream and to ping the relays from time to
time.
the more helpers, the less work for all of us. last year, it has
been great fun, although chatting, monitoring (with some 5 seconds
lag) and listening at the same time can make your head spin in funny
ways.
please cc: me privately on any replies. thanks.
regards,
jörn
--
Jörn Nettingsmeier
Lortzingstr. 11, 45128 Essen, Germany
http://spunk.dnsalias.org (my server)
http://www.linuxaudiodev.org (Linux Audio Developers)