> What about using midi-thru boxes? they don't add latency to
> the midi signal,
> they just 'amplify' and 'split'.
If you run 14 MIDI devices through one MIDI Port using
MIDI Thru Boxes you get a worse timing compared
to using a multiple Output Interfaces like
the Steinberg Midex 8 oder the Midiman Midisport 8x8.
But I don't know if there is a 8x8 Interface with Linux drivers.
All of them implement a feature to reduce latency and I think
that's a problem for the driver developers because the companies
don't want to show the specifications for that in public.
Regards,
Joachim
PS: I added the Linux-Audio-User group to the cc because this
is linux specific question.
> ----- Original Message -----
> From: "Tjeerd Sietsma" <tsietsma(a)hotmail.com>
> To: <music-dsp(a)shoko.calarts.edu>
> Sent: Tuesday, October 22, 2002 9:03
> Subject: [music-dsp] linux and midi
>
>
> >
> >
> > I need to connect 14 midi devices (each using 1
> programmable midi channel)
> > to a PC runnig linux. Midi thru isn't an option, since the
> latency will be
> > too high. Does anyone has hardware recommendations?
> >
>
> dupswapdrop -- the music-dsp mailing list and website:
> subscription info,
> FAQ, source code archive, list archive, book reviews, dsp links
> http://shoko.calarts.edu/musicdsp/
>
I just searched the archives on this and the suggestions there
didn't help.
I'm trying to convert some mp3 files into wav files. First I
tried having Audacity export a wav file. Then I tried loading the
mp3 into Broadcast 2000 and having it render it to a wav. Finally
I tried the suggestion in the archives of mpg123 -w filename.wav
filename.mp3.
In each case, when I tried to burn a CD from the resulting wav
file, cdrecord reports "inappropriate audio coding in
[filename]." I'm using Eroaster 2.0.10 as a front end to
cdrecord.
Audacity and Broadcast 2000 both load all these wav files and
play them just fine.
Anybody know what's going on here?
Thanks.
Howard Sanner
flagstad(a)mindspring.com
Apparently my subscription to the list got dropped. I guess
that's why things have seemed so quiet lately in Linux audio
land.
I just searched the archives on this and the suggestions there
didn't help.
I'm trying to convert some mp3 files into wav files. First I
tried having Audacity export a wav file. Then I tried loading the
mp3 into Broadcast 2000 and having it render it to a wav. Finally
I tried the suggestion in the archives of mpg123 -w filename.wav
filename.mp3.
In each case, when I tried to burn a CD from the resulting wav
file, cdrecord reports "inappropriate audio coding in
[filename]." I'm using Eroaster 2.0.10 as a front end to
cdrecord.
Audacity and Broadcast 2000 both load all these wav files and
play them just fine.
Anybody know what's going on here?
Thanks.
Howard Sanner
flagstad(a)mindspring.com
Hi all,
If I use mpg123 -o oss it plays fine, but I can only use my first
soundcard. If I use mpg123 -o alsa09 it plays the same volume, but the
sound is distorted. mpg321 has the same problem. Programs like aplay do
not have the problem, so it's not for all alsa software.
what do mpg123 and mpg321 have in common? and what makes them distort when
using the alsa output instead of oss-emulation?
the system is a AthlonXP2000+ with Asus on-board cmipci and a Hoontech
ymfpci card. Both cards have the same problem with mpg123 and mpg321 and
both not with aplay.
anybody an idea?
thanks,
Olivier
For thos who are not following the alsa-devel list and want to know
about this card.
--
Patrick Shirkey - Boost Hardware Ltd.
For the discerning hardware connoisseur
Http://www.boosthardware.comHttp://www.djcj.org - The Linux Audio Users guide
========================================
"Um...symbol_get and symbol_put... They're
kindof like does anyone remember like get_symbol
and put_symbol I think we used to have..."
- Rusty Russell in his talk on the module subsystem
New in FreqTweak 0.4.4:
* Now deactivates processing while switching FFT sizes. This
should prevent JACK from kicking you out here.
* Added ability to reconnect to JACK with a different basename,
or just reconnect period if we are kicked out (I/O tab).
* Control-Alt left mouse draws arbitrary lines between
initial click and current cursor. Also smoothed regular drawing.
* More configure checks to ensure compiling against the float version
of FFTW.
http://freqtweak.sourceforge.net
jlc
Hello all,
I'm releasing a bit of an old version of SpiralSynth Modular, while work
continues on the 0.2.0 version on sourceforge.
It's got some rather nice features, and I should have released it months
ago - but it had a crash bug I only found yesterday...
anyway:
Main app changes:
Back to one window GUI with seperate movable areas
Fixed Plugin window events/drawing bug
Echo/Delay crash bug fix
Less C code in SpiralSynthModular.C (Andy Preston)
Ability to load ssm files from command line (Dan Bethell)
gcc3.1 fixes + dynamic linking bugfix (Takashi Iwai)
configure + make fixes (Takashi Iwai)
Plugin changes:
LFOPlugin (Andy Preston)
MeterPlugin (Andy Preston)
StereoMixerPlugin fix (Andy Preston)
Scaling fix and name sorting for the LADSPAPlugin (Mike Rawes)
Sequencer bugfix for dragging note events
Matrix now 64X24 + tweaked GUI
Matrix can be controlled by an external clock pulse
Matrix copy/paste and transpose functionality (Andy Preston)
JackPlugin fixed for recent versions of jack (Takashi Iwai)
http://www.pawfal.org/Software/SSM/dload/SpiralSynthModular-0.1.1.tar.gz
and if you want the cutting edge it's here:
http://sourceforge.net/projects/spiralmodular
: noize farm
I'd also like to announce the noize farm, an open collection of ssm patches
that are shared in the spirit of the GPL:
http://groups.yahoo.com/group/noize_farm/
Feel free to download, hack, modify, upload and generally have fun with
anything you find there. It's currently hosted at yahoo, which means you
have to sign up with them to get access to the files - but we think it's
worth the hassle for now.
I don't see it as being restricted to just ssm patches either, if you
like the idea then you can share anything (linux) audio related - as
long as the files are small, and open for all to play with.
All the best,
Dave
: www.pawfal.org :
hi all,
got a quick q about usb midi in alsa...
i've already built my alsa-driver (0.9.rc3) for snd-usb-audio, to run my
quattro....
but i've just been given an Evolution UC-16 on trial, which is usb midi.
so normally, i'd rebuild my alsa-driver ./configure
--with-cards=usb-audio --with-cards=usb-midi but it will only ever build
just one of the drivers if i do that (usually the last one in the
list)...
is there a way i can build usb-midi *and* usb-audio?
i'll probably be doing quite a bit of testing with usb-midi, but will
mostly be running usb-audio for my own use.
or is it possible to build the alsa-driver for usb-midi and manually
copy it into place?
thanx in advance
m~
--
iriXx
www.iriXx.org
copyleft: creativity, technology and freedom?
info(a)copyleftmedia.org.uk
www.copyleftmedia.org.uk
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*** stopping make sense ***
Greetings,
I just bought a new system. It has integrated sound on the Mother
Board. There is a jack marked "spk" (obviously the speaker jack). But
the manual refers to that jack as "line-out" I was always under the
impression (just an assumption) that "line-out" and the speaker jack
put out completly different voltage levels (and we know how dangerous
it could be to plug things into the wrong voltage levels).
Are the standard speaker jacks indeed "line level"?
There's several possiblities that I can think of:
1. The standard "speaker out" jack complies to the "line level" standard
2. Since it's a costom made computer, the jack is incorrectly marked
and its actually a line out jack. (after all, the VGA port is marked
as "COM2")
3. Since the manual covers several versions of their Mother
board, it's actually a different interface that it's referring to (I've
run into this with the manual before).
I'm actually hoping to find out that the speaker out jack complies to
the line level standard. This way I can use the speaker out for line
level devices.
Let me know what you know,
Thanks,
Rocco
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hi,
i've just got a m-audio quattro (very nice :) and have alsa running
happily... but i'm having trouble starting the alsamixer.
i've copied in Patrick Shirkey's scripts for .asoundrc and qinit, but
when i run qinit it segfaults :(... alsamixer says 'no mixer elems
found' and when i run gnome-alsamixer i just get a blank screen with
'USB Audio Quattro' at the top. i'm getting sound fine out of
applications such as xmms though... although SpiralLoops crashes and/or
produces digital glitch noise when i import a sound....
any ideas?
thanks in advance
m
--
iriXx
www.iriXx.org
copyleft: creativity, technology and freedom?
info(a)copyleftmedia.org.uk
www.copyleftmedia.org.uk
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*** stopping make sense ***
