Hi,
running debian sid with a 2.4.18 kernel and alsa 0.9.5, I think.
Folks on this list and alsa-user helped me get alsa working with my
onboard sound, via vt8233, about a year ago. It works ok, but I've
got 1 problem and one question.
PROBLEM: every once in a while -- not often -- sound will hang and
bring the whole system down; this happens with xmms, alsaplayer, and
zinf, so I don't think it's strictly the playback software's fault.
usually this isn't a catastrophe, since this is a workstation, not a
heavy-duty server, but I'd like it not to happen. Anyone see this
behaviour before, or know how I should try to diagnose it?
QUESTION: My sound card has three analog outs (green, blue, pink). I
gues sthese are for a 5.1 surround sound system or something, but I'd
like to try to have one stereo out go to my cheapo speakers, and one
go to my headphones. Unfortunately I'm unable to get a signal out of
anything other than the pink out. The others just give static and
wierd stuff, even with all the outputs turned up in alsamixer. Anyone
know if it's possible to set this up the way I want? Even general
hints abbout how 5.1 systems work would probably be really helpful.
thanks loads,
matt
Richard,
For me it looks like your kernel has
ALSA included into source tree already.
It is only a guess -- I dont have
a lny linux in hands right now
but you can check
it with something like:
find /usr/src/linux/ | xargs grep -sqi alsa && echo yes I have ALSA in kernel
NB: i haven't try this but should work
horsh
==================8<===========================
Hallo,
Richard hat gesagt: // Richard wrote:
> This only 2 examples.
>
> gcc-3.3.1-6
> kernel-2.4.22-1.2097.nptl
> kernel-source to match
>
> gcc -D__KERNEL__ -DMODULE=1 -I/home/ric/alsa-driver-0.9.7c/include -I/lib/modules/2.4.22-1.2097.nptl/build/include -O2 -mpreferred-stack-boundary=2 -march=athlon -DLINUX -Wall -Wstrict-prototypes -fomit-frame-pointer -Wno-trigraphs -O2 -fno-strict-aliasing -fno-common -pipe -DALSA_BUILD -DEXPORT_SYMTAB -c hwdep.c
> In file included from /home/ric/alsa-driver-0.9.7c/include/sound/driver.h:42,
> from hwdep.c:22:
> /home/ric/alsa-driver-0.9.7c/include/adriver.h:134: error: redefinition of `PDE'/lib/modules/2.4.22-1.2097.nptl/build/include/linux/proc_fs.h:17: error: `PDE' previously defined here
> make[1]: *** [hwdep.o] Error 1
> make[1]: Leaving directory `/home/ric/alsa-driver-0.9.7c/acore'
>
...
> At some point i been told that the drivers needed be fixed to compile in
> fedora.so. well if you can make out something out of this.. ill appreciate
Well, this is a bit strange. I'm guessing a bit now:
The offending lines in your example result from adriver.h defining
things for some kernel versions, including kernel 2.4.22.
For example the error regarding PDE:
/home/ric/alsa-driver-0.9.7c/include/adriver.h:134: error:
redefinition of `PDE
'/lib/modules/2.4.22-1.2097.nptl/build/include/linux/proc_fs.h:17:
error: `PDE' previously defined here
This means, that somewhere in "include/linux/proc_fs.h" the name PDE
is already defined. In adriver.h ALSA tries to define it on its own
like this:
#if LINUX_VERSION_CODE < KERNEL_VERSION(2, 5, 4)
#include <linux/fs.h>
static inline struct proc_dir_entry *PDE(const struct inode *inode)
{
...
#endif
The strange thing is, that in my rather vanilla version of kernel
2.4.22 there is no such thing as PDE in include/linux/*:
$ grep PDE
/usr/src/linux/include/linux/proc_fs*
$ (nothing here)
So I suspect, but I can only guess here, that your kernel source has a
patch that introduces this PDE thing (I have no idea what it could be
used for).
I don't know if ALSA could fix that at all. It might be, that if
Fedora ships a custom Linux kernel source, they should also provide
adapted ALSA-sources.
ciao
--
Frank Barknecht _ ______footils.org__
Sorry about this extremely naive question, but since this is my first time
trying to get linux to be a DAW I stumble onto some problems of course. And
my know-how is somewhat limited unfortunately.
My staudio c-port (envy 24 based 8 channel card analogue + 2 channel digital
from www.staudio.com with built in midi-interface) works as far as audio goes
now, but I can't seem to get anything out of the midiports even though it
seems to be properly recognized. Alsa Patch bay gives me a: Rawmidi: 0 -
Hoontech SoundTrack Audio DSP24 MPU-401 entry at least (actually 2 - one for
each midi port), but no matter which I choose in rosegarden or any other app,
I get no signal in the displays. How do I find out if it's working, and how
do I troubleshoot. If this is well documented somewhere and comprehensible
for semidummies, I'd be happy to receive a link!
Best regards
Ketil
I believe the ALSA status page isn't as up-to-date as it should be. I would actually rely more on annecdote than it.
Taybin
-----Original Message-----
From: Klaus Kosten <Klaus.Kosten(a)gmx.de>
Sent: Oct 19, 2003 7:27 AM
To: A list for linux audio users <linux-audio-user(a)music.columbia.edu>
Subject: Re: [linux-audio-user] How to check if I have midi connection?
Robert Jonsson wrote:
>
> Hi Klaus,
>
> If you need the midi ports there is a chance that you are correct.
>
> The audio ports though should work splendidly, support for envy24 based cards
> is generally very good in Linux, and quite sure this includes the DSP2000
> (Some related reading
> http://myweb.cableone.net/eviltwin69/ALSA_JACK_ARDOUR.html)
>
> > For me, there is no other choice than switching back to Win ?98 and
> > Cubase VST and forgetting completely about sound under Linux, at least
> > for now.
>
> As I said, if you really need the midi-ports you might be right. On the other
> hand, midi-ports are cheap, most mainboards come with joystick-ports
> nowdays... (perhaps not the same quality though).
> In general it's more of a question of what apps you intend to be using. VST is
> probably a more stable and more feature rich environment.
> Linux-audio is still in what "suits" like to call, early adopter stage,
> meaning that it's mainly applicable to people that really want to use it and
> are prepared to jump through a few hoops to do it.
> But we are going to change that, right? ;)
>
> /Robert
Hi Robert,
you are completely right, and I agree with you in principle. But the
DSP24 as the base of the DSP2000C has been marked as "supported" in the
sound card matrix since 2 years or so, even the ST Audio homepage says
it?s supported, and now suddenly the ALSA guys tell us it?s even
"untested". That?s what I call bad manners, at least. I never would have
bought the DSP2000C with this status.
FWIW, I?m a Linux user since 1994, as a Sysadmin I have moved the
network in our university institute from Novell to Linux, for everything
I do with a computer I use Linux (except music), so I am well aware of
the problems and am willing to be patient, but this is definitely not
the way I like it.
Regards
Klaus
--
Dr. Klaus Kosten
Am Ginsterberg 13
D-52477 Alsdorf
hi * !
music.columbia.edu have upgraded their mailman setup.
douglas repetto tells me there may have been a few lost messages, so if
yours has failed to show up by now, please repost.
problem reports to lad(a)music.columbia.edu.
btw, as a reminder: both lad and lau are now members-only. you *must*
subscribe before you can post. moreover, messages containing html will
now be discarded quietly without notice.
best,
jörn
Greetings:
I've been working with Tim Thompson's KeyKit, trying to get it to send
its output to a softsynth. I can set a variable for KeyKit that will
select the ALSA rawmidi device with this syntax :
export ALSA_RAWMIDI_DEVICE=hw:1,0
where hw:1,0 refers to my card1 (the ALSA virmidi module) and its first
device (0). Now when I start KK (and set the Port Enabler to ALSA) I can
route its output to whatever connections are available via aconnect (or
the kaconnect GUI).
I just wanted to say "Thanks!" to the ALSA developers for the virmidi
module. I've used it with MIDI sequencers running under emulation
systems such as DOSemu and Xsteem, I've used it for the TK707 virtual
drum machine, and now I'm using it with KeyKit. The combination of
TiMidity-as-softsynth (with the Fluid soundfont) and the virmidi module
is a very powerful and flexible system for me, and I just wanted to pass
on a note of appreciation for the ALSA team's work.
Okay, that's all. We now return to our regularly scheduled programming...
Best regards,
== dp
Does anyone know how to get RoseGarden4 and Ardour 0.9beta5 to play nicely? I
thought that's why they added the JACK master/transport master function in
Beta5 and I'd really like to have both of them synced.
What I'm trying to do is not unlike using SMPTE have Rosegarden act as a slave
and Rosegarden to be the master.
Here's what I do:
1) Fire up Jack using QjackCTL
2) Load Ardour
3) Load Rosegarden
I create a sequence in Rosegarden. Then nothing happens. :)
I've gotten this far, but I can't get Ardour in Sync with Rosegarden when I
press play in Ardour, rosegarden isn't "chasing" or locking onto sync. I'm
trying to capture my sampler which is inputted to my Delta card and have the
MIDI/Audio in sync. and no, I haven't been using MTC either..
I've noticed there is a Start|Pause feature in the qjackctl front end for
JackD but It doesn't seem to have any effect.
Is this configuring possible, or am I mistaken and just misinterpreting what
the terminologies mean? If it's in fact just ardour and/or rosegarden has
anyone been successful in syncing a MIDI sequencer and multi-track audio
editor (MUSE/Audacity perhaps?)
Regards,
Chris
sometimes I can be such a damn cone head. I've sent this twice, letting it
be in html form which I think will get rejected I hope y'all don't get
three copies of it.
--
Hello all - here's something I'm having a bit of trouble with, I thought
y'all might know (hell it might've been in that huge mp3/ogg thread and I
just missed it)
I need to be able to rip hi res mp3s (320kbs) for submission to certain
liscensing agencies (they insist) - but I haven't been able to figure out
how to do this on my linux box. I know lame is the main backend - but I had
some kind of trouble a couple weeks back either locating it or finding
detailed info about it or something (I'm gonna go back and do this homework
again), and then of course there's the matter of a front end, which I don't
necessarily care about :) (but I remember not being able to figure out the
proper command line commands or something)
I had much success doing 128kbs mp3s - grip does it, audacity will just
change a wav to an mp3, that's awesome, but they don't seem to deal with
higher resolution than 128 - something made me think it's just not happening
yet in linux and that I need to just rip them on my windows machine...was I
right about this?
I hope I didn't just miss someone talking about this on that long thread :)
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