I have recently bought the ice1712-based DSP2000 C-PORT from Hoontech and
I'm really satisfied with it. I only have a little problem: MIDI!
The card has 1 MIDI in and 2 MIDI out, and the 2 outs are working fine.
However, I can't get anything from MIDI in :-(
Anyone has that card and can confirm this problem or I have to check my
hardware? Maybe ALSA get confused? With aconnect I see 2 devices (0 and 32)
both as inputs and as outputs...
I've tried on the staudio.com forum and they suggest to test the card under
windows, but (ehm..) I don't have it...
| Emiliano Grilli |
| emillo(a)libero.it |
| Linux user #209089 |
| http://www.emillo.net |
I have jack installed and working now. So, now I'm trying to get
AlsaModularSynth compiled and running. Has anyone else accomplished this
in debian stable (woody)?
I installed libqt3-mt (along w/everything it depended upon), fftw-dev,
fftw2, sfftw-dev, sfftw2 and ladspa-sdk to get ladspa.h. I've tried to
edit the top of make_ams to agree with the location of the qt3 libs, bin
and include directories on my system as best as I could figure ... but
I'm not sure I got it right:
it seems to compile OK when I do:
make -f make_ams
though there are a lot of warnings about comparing signed to unsigned
and other warnings that flew by. Are these benign?
Anyway. I end up with an executable ams. When I run ./ams I get this:
./ams: error while loading shared libraries: libjack.so.0: cannot open
shared object file: No such file or directory
I have that in /usr/local/lib/libjack.so.0 so creating this link appears
box1:/usr/lib# ln -s /usr/local/lib/libjack.so.0 libjack.so.0
Is that the correct sollution?
ams: cannot connect to X server
I assume this is because X is running as me (eric) and not as root. I'll
try getting out of X and starting it as root as soon as I finish writing
this message. But, is there a way to let root connect to the X server
that's running as eric?
I was sort of ok (well, not really, but I'm giving it a try) with the
idea of running jackd and its clients as root ... but, running X as
root, too? that makes me more nervous, somehow... maybe I shouldn't be
nervous about that ... I'm not sure.
anyway. i'll give it a shot with a root X session.
I'm thinking that maybe I should patch and recompile the kernel and
recompile jack to enable capabilities so I don't have to run as root.
Do those of you running jackd as root also run your X session as root?
Thanks for listening,
PS If all of this has been covered before and my questions are annoying
you, feel free to send me to the archives. I try to follow jackit-devel,
but most of it seemed to go over my head before I actually started to
try it. So, re-reading the past couple months would probably do me some
good now that I have the real thing in front of me. -edrz
First the vitals:
The computer system is a 400 MHz K6-2 with 512 MB memory, and a Hercules
Fortissimo II (Cirrus Logic Soundfusion CS4624 processor) sound card.
The OS is Linux-2.4.20 (built up from Slackware-8.0, with kernel patches
and package upgrades). I have the following ALSA components installed:
alsa-driver-0.9.0rc7, alsa-oss-0.9.0rc1, alsa-utils-0.9.0rc7,
The kernel is mostly monolithic, with the following options (abridged to
include only those I know are relevant; if folks want the complete
listing, I can easily provide it):
ALSA is the only component that I have modularized (only because I don't
have the impression it can easily be just built-in, actually), and the
following modules are loaded:
Module Size Used by
snd-mixer-oss 11120 1 (autoclean)
snd-cs46xx 58096 1 (autoclean)
snd-pcm 50752 0 (autoclean) [snd-cs46xx]
snd-timer 9392 0 (autoclean) [snd-pcm]
snd-ac97-codec 26912 0 (autoclean) [snd-cs46xx]
snd-rawmidi 11808 0 (autoclean) [snd-cs46xx]
snd-seq-device 3824 0 (autoclean) [snd-rawmidi]
snd 27280 0 (autoclean) [snd-mixer-oss snd-cs46xx
snd-pcm snd-timer snd-ac97-codec
I've been using Linux with OSS and a Media-Vision Pro-Audio-Studio-16
soundcard for years, and have basically had very little trouble with
that combination. Upgrading to Linux-2.4.x caused the sound output to
be rather noisy, so I recently took advantage of the situation to buy a
new soundcard, and upgrade to using ALSA (which I've been wanting to do
so I can investigate certain rather exciting applications such as Ardour
and the new Rosegarden).
The applications I've been using for years with OSS, (notably Dap,
Mixviews, Aumix, and a few others) all appear to mostly be working fine,
presumably through the ALSA OSS emulation. Aumixer now shows some
lables (PhoneIn and Video) which don't correspond to any inputs I have
on my soundcard, while some others (IGain and Line1) don't appear to
have any use, given the way I'm currently using the card (signal from a
hardware mixer going to the Line input, and signal from the main, front
line output going back to the mixer).
I can play back sound files without any problem. In fact I've also tried
a few new (to me) applications, such as Audacity, ProTux, and (though
I'm having problems with it I won't cover here) Ardour, and they too
play back sounds quite nicely (the sound is "grainy", though, perhaps I
need to investigate the Low-Latency issues more, or increase the size of
some buffer somewhere).
I can send sound to the computer, and listen to that sound coming back
from the computer (adjusting the level with the "line" level in Aumix,
for example), but if I try to record that sound (in any of the
applications I've tried, including all those listed above), with "record
enable" selected in Aumix, I get no audio recorded (silent sound file).
Also, I find it interesting to note that I can record-enable only one
input source at a time (contrary to my old Media-Vision card). I'm
assuming that's a function of the hardware, though and can't be fixed in
Finally, we get to my questions:
- Can someone point me in the right direction(s) to find a solution to
this, so I can record audio (preferably from numerous sources) into
my computer? I'll be perfectly happy with pointers to documentation
(though I expect to get at least some to documentation I've already
read, because I think I've repeatedly gone through all the
documentation I already found myself), and suggestions to upgrade
- I've tried to start Jackd at system boot, with the following command
added to my rc.local file:
/local/bin/jackd -d alsa -d cs46xx -p 512 &
Jackd fails to start at that point (I'm afraid I haven't noted the
exact error message, but I can make a point of doing so next time I
reboot the system), but it does start when run manually from a root
shell. Does anyone have any idea why jackd might not start at boot
time, (perhaps the ALSA modules aren't loaded yet at that time)? If
I understand my boot sequence properly, kernel modules are setup
(via "depmod -a") before rc.local is run, so the modules may not be
loaded, but they should by that point be loadable (they load
automatically by the kernel in regular use).
- Is it possible to build ALSA into the kernel?
- on a slightly unrelated point, my new soundcard has a built-in
synthesizer with at least some features I'd like to explore. Though
I know this won't replace any of my hardware synths (or some of the
software synths I've begun to play with), I'd like to play around
with some of its built-in sounds, and probably use it (at least
until I get something better) as my "preset playback" device,
controlled from an external MIDI controller. Does anyone have
suggestions for how I might go about that? (again, simple pointers
to existing documentation would be more than appreciated).
In case people are wondering about my specific intended application,
I'd like to use a Roland Octapad to trigger percussion sounds from
the soundcard's built-in synth.
Thanks in advance for any pointers people can provide....
Sylvain Robitaille syl(a)alcor.concordia.ca
Major in Electroacoustic Studies Concordia University
Faculty of Fine Arts / Music Department Montreal, Quebec, Canada
Dear audio users:
Previously I asked about USB capture, eg., to laptop, but I still have
1. Is there a "standard" USB interface for audio? For example, will newer
devices, like "Audiophile", "MobilePre", and "Transit" from Maudio work
using some kind of standard interface as the "Duo" and "Quadro"?
I've seen references to "ASIO" or somesuch. Does that mean they
will all function with standard drivers?
2. What do the [alsa|au|x]mixer controls do?
In particular, if I adjust input gain on (analog) line input will that be
done before or after the A/D? Note, I may need to adjust line level if I
am driving from different sources, like phonograph, cassette tape, or
Similary, for mic in, is there a separate control for this, eg., before
the mic preamp?
I could not find this information, neither in docs from Maudio nor in alsa
docs. Please point me to the appropriate place if this is written down
somewhere. What I am looking for is a "block diagram" showing where
the amps and level controls really are.
3. If mixers do allow level adjustments above, which USB devices have mixers?
People have claimed that Edirol UA-1A doesn't have a mixer.
How about Edirol UA-3D?
What about Maudio Audiophile, MobilePre, Transit?
Or, perhaps some devices have mixers, but USB control code is not written?
4. Is it usual to have stereo mic inputs on sound cards?
Maudio "Revolution" (and most soundblasters?): *mono* input.
Maudio "Transit": claims stereo input
Maudio MobilePre, Duo, etc. All naturally have stereo input.
5. Finally, I noticed that many of the Maudio USB products, like Audiophile,
Quatro and Duo, need external power. The funny thing is that they
need 9 VAC. AC voltage! What's up with that? I was expecting
some standard DC voltage, eg., 9 or 12V, suitable for mobile use.
Is there a simple way to power these from DC voltage, eg., batteries? I am
willing to open one up (and void the warantee) if necessary. Perhaps there
is a psu inside and I can just bypass that and give some standard voltage
Len at Core Sound made a portable Duo. Perhaps he can tell us
how he did it...
Thanks for all the help. I'm really enjoying this list...
I am trying to use the hardware wavetable midi synth on my YMF 740C sound "card" onboard my motherboard with ALSA and MusE. I have been searching through the various sources of documentation trying to solve the mystery but I am stumped.
# lsmod lists snd-mp401-uart, snd-rawmidi, and snd-opl3-lib among the sound related modules. Can someone give me a definitive answer as to what these modules are? I am pretty sure that snd-mp401-uart is the wavetable midi synth, and am guessing that snd-opl3-lib are the patches for the legacy FM synth that is also on the card, but I don't have any idea what the snd-rawmidi module is for. And why is there no snd-opl3 module itself? Is the legacy FM synth not supported under ALSA?
I checked /proc/modules, so these three modules are loaded, but I can't figure out how to access them generally or through MusE.
Checking /proc/asound/devices does not list any midi devices, (see below):
[root@localhost root]# cat /proc/asound/devices
0: [0- 0]: ctl
27: [0- 3]: digital audio capture
18: [0- 2]: digital audio playback
17: [0- 1]: digital audio playback
16: [0- 0]: digital audio playback
24: [0- 0]: digital audio capture
1: : sequencer
33: : timer
As far as MusE goes, I have gone through the configuration section fairly thoroughly but without finding a way to access mp401-uart. The closest thing that I found was under the midiport configuration table->other raw devices, which brought up a context dialog box which had a choice of raw midi, pipe, or something else which I can't seem to remember at the moment. I tried entering /dev/midiC0D0 as the path for the raw midi device but that didn't configure a valid midi output for the device.
I want the option to use my hardware synth, as well as the software synths, with MusE and generally under ALSA/Linux. Does anyone see something I missed?
ams-1.5.12 is a major update for AlsaModularSynth. There is now a
"Parameter View" dialog where you can group any parameter from any module
and create your own GUI for your patches. Values for the parameters in
"Parameter View" can be stored as presets and restored via MIDI program change.
There is a new tutorial on modular synthesis (commented patches) and
documentation has been updated as well.
Have fun !
Dr. Matthias Nagorni
SuSE Linux AG
Deutschherrnstr. 15-19 phone: +49 911 74053375
D - 90429 Nuernberg fax : +49 911 74053483
I need advice on getting a very good PCI soundcard.
I hope to use it for recording my own multi-track
sounds using Linux Audacity with Red Hat 9.
Guitars and bass mostly. Maybe a DX100 keyboard
My system has an Athlon CPU,
Asus A7N8X motherboard with a gig of DDR
I'll be useing a Eurorack UB802 mixer for input.
Could someone tell me a good choice for
a sound card? I expect to spend about $250.
( I being optimistic! )
Any advice would be most welcome!
> I know you are very busy and I hate to bother you but I have been
> trying for long time to get my soundcard (Santa Cruz aka Cirrus Logic CS4297A rev 4 as
> listed in AlsaMixer) to record it's output into various audio apps. For
> example it has a cancun drum wave table and I want to record the midi output to wave.
> I've been experimenting with the mixer: enabling the capture, pcm out etc. to no avail.
> I know this can be done in windows with the soundcard's GUI. If you
> have the time I could use a few suggestions. Thank you.
> Richard O'Donnell
I have forwarded this to the lau list as there should be someone there
who can help more.
Are you having trouble with capturing the audio or getting the sound of
I'm not very well versed on midi issues but I suggest trying muse, seq24
or rosegarden. They are very capable apps for working with midi.
The basic problem is making sure the onboard wavetable is being
accessed. If you are sure this is working the above apps will be able to
capture the sound relatively easily.
Patrick Shirkey - Boost Hardware Ltd.
Http://www.boosthardware.comHttp://www.djcj.org - The Linux Audio Users guide
Being on stage with the band in front of crowds shouting, "Get off! No!
We want normal music!", I think that was more like acting than anything
I've ever done.
Goldie, 8 Nov, 2002
I know my way around my linux box (kernel compiles et all), and I'm inches
away from dumping windows completly. Still I've got problems with sound
recording on linux.
I need help trying to record from Windows Media ASF streams on the net. I
currently use MPlayer to play those streams, and it plays just fine.
I tried to record from it. I'm using KDE so I fired up KMix and found that
MPlayer plays through the PCM device (that's what KMix calls it). But I
can't set PCM as a recording device, so I can't record from it. And since
MPlayer doesn't play through the KDE sound daemon, can't use that stuff
So how should I go about this ?
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