Hello Xenia!
the only idea I'd have, is to re-enable the pcspkr module and in alsamixer 0
or whatever you use to setup your audio ardware -, mute it.
Otherwise the real pc-speaker as such is no ALSA device, as far as I'm
aware. The pc-speaker module is, I believe, intended to use it as a
l
Hello Klaus!
I've got a delta 1010 LT, which is also based on the ice1712. I onlyuse
alsamixer, but there I haven't had a problem in years. I can see all my adc,
dac, hw-in and more. by pressing f5 I get both full capture and playback
controls. So if this is really an ALSA fault, then it must
You're very welcome!
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
=== AND MY PERSONAL PAGES AT: ===
http://www.juliencoder.de
---
Hello Conrad!
Did you install anything new recently? Oder did you update something or
enable/not disable auto-update?
You have both pulseAudio and ESD sound servers. I'm not really sure which
one of them is used with desktops these days, but I think, that they block
normal ALSA access as a
Hello Paul!
Thanks for the detailed reply. It all looks very helpfu, except one minor
thing, which I hope is wrong. :-) But frst the good news: We tried some more
and restarted and now there is a soundcard, which evidently works, so I
suppose it's udev trouble. Well in my experience udev and
Hello everyone!
A friend of mine can't get sound on Linux Debian Wheezy/Sid. Here are the
hardware data:
Card: HDA Intel pch (6 series chipset family, High Definition Audio controller
Rev. 05)
Chip: Intel cougarPoint HDMI
He runs the standard Kernel with the Distro ALSA version 1.0.23.
D
Hello!
Hm, that's rather a complex setup. I only have two ideas: 1. Shouldn't you
directly supply which hw device you want to use, so instead of "hw" something
like "hw:1" or "hw:1,0"?
2. Is it out of the question to use JACK? JACK can automatically mix streams
and most audio apps support
Hello Jose!
I do have the Delta 1010LT. But from what I know, and this is quite certain,
M-Audio always uses the same basic chip (ICE1712 or ICE1724) - in their cards.
That is very well supported, has been so for years. So 98% very good support!
Good luck!
Kind regards
Julien
Hello!
go for the MAduio Delta card. I'm not sure about the Audiophile, but I think
they have the same chips inside. The Delta44 has 4ins/4outs. It has spdiff and
MIDI as well. In alsamixer I can set the clock to 32kHz, 44.1kHz, 48kHz,
64kHz, 88.2kHz and 96kHz. The card support 24bit.
The
Hello Immanuel!
Thanks for this, it really does help and sort out the problem.
Kindly yours
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
=== AND
Hello1
In a system, where cards are just found by the system, is there a way, to
actually configure a persistent numbering of cards? My friend is running
Debian Lenny and cards are discovered by, what ever finds them. But now his
card numbers were changed this morning, which caused some confu
Tom!
Usually a sink, is something where data "disappears". So the ALSA sink,
would be the ALSA output, since the audio data fom your computer is emptied
into it - so to speak - to get to your speakers. I sometimes heard of such a
concept as drain as well. the opposite would be the source. So
Hello Tom!
I don't know your video editor and haven't come across such a message
anywehre. But, it sounds like your ALSA output is busy. Do you have JACK
running? Or do you have another application using/blockng ALSA output?
Kindly yours
Julien
Music was my first love and
Hello Yan!
This problem was mentioned before. I don't think, that I've read a solution
yet, but I didn't follow it too closely. Perhaps in the meantime you could use
mplayer/ It is also a commandline tool, support a lot of formats and drivers.
I use it.
Another possibility might be, to cha
Hi Lenson!
No problem. As long as you fixed your problem. I thik that's why this list
exists. To help. It's always learning. I can tell from own experiences. :-) I
guess so can others.
Warmly yours
Julien
Music was my first love and it will be my last (John Miles)
===
OK, but so you were abel to completely establish, that the card receives MIDI,
can probably send it and it can doso under Linux. It shouldn't be an issue, my
card is running here since 2002, first under a 2.4 kernel and now under ever
changing linux kernels (2.6) with their built-in ALSA. I only
Hey!
Do you know, why this is happening? Can you see any reason, why your
keyboard shouldn't send MIDI-events? Or do you suspect something's broken
there? Do you have a simple old MIDI keyboard, or does it have a USB or other
connector?
Kindly yours
Julien
Music was my
Hi!
I don't exactly have your kernel, but very close.
Did you try:
aconnect -li
and
aconnect -lo
These should also list your ALSA sequencer MIDI ports.
To see, if your soundcard gets MIDI-input:
aseqdump -p 20:0
Or was your card "10".
If you see something there, then at least you'
Hello Tghomas!
I might not be abele to answer you completely. But here's what I can say:
1. VirMIDI is NOT connect to any hardware. It is just used to a) simulate MIDI
in a computer without any MIDI, if you run an application which requires it.
b) route MIDI data internally.
I don't know, i
Hey Grant!
Yes there is a difference. plughw automatically does some conversions for
you and some looking after you. I'm not sure what exactly this comprised. But
I _THINK_! it was samplerate conversion?, channel-counting and opening the
device, so it won't conflict. If there was more, I don'
Hi!
If it works fine with JACK,peraps that's your way to go. I don't know anyu
plugin either, where you can tune period and frames. But I think audacity
supports JACK as well.
Sorry, for not being anymore of help...
Kindest regards
Julien
Music was my first love and
Hi!
A few questions: Do you get any sound at all? Did you have another OS, than
Linux, installed and tested the sound there?
If you get no sound at all: It's the ich-driver again, I would suspect or
still some issue with routing the sound to the right output. If sound is just
too low and a
Hi!
I'm not sure, if alsa does it, still. But you can do it with jackd (Jack
Audio Connection Kit). It's a low latecny audio server and a lot of Linux
Audio software support it. You can find packages in your distro.
Then you simply do:
jack_connect system:capture_1 system:playback_1
jack_co
One idea: Is your jack a combined RCA/stereo jack? If so, windows might have
some code to switch between the two. But ALSA doesn't properly recognise, when
no jack is in.
Regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT
Hello Peng!
Well the part of .asoundrc I posted should work for your card. I have a
delta 1010LT, but regarding JACK it's not so complicated.
the only thing you need to do, to make this configuration work is installing
JACK. The JACK Audio Connection Kit. It should be shipped with all Linux
Hi!
I never used virtualbox or whatever. I guess it is seem kind of virtual
machine like vmware?
I can help you with mplayer a bit though. It's running on my system and it
works just fine. Nowadays I like to have an easy life though and I use JACK.
But in the old days I just did this:
mp
I have the jack plugin working here. I'm running debian Lenny with a
self-built kernel (ALSA).
Did you test the jack plugin with the same samplerate as your soundcard.
So if your card's 44100 start JACK like this:
jackd -d alsa -d hw:0 -r 44100 [your other favourites]
and then use a CD quali
Hi!
This should already use samplerate conversion. If libsamplerate is used... I
wouldn't guess so.
I remember there was a special plugin for samplerate conversion or at least
in the old docs, they showed how to convert samplerates explicitly...
But jack itself should be able to convert
Hi!
I'm not sure about your asoundrc, but something like that should do.
BUT: Why not use JACK instead? You have single inputs there for every
channel. There are simple recording tools or a simple way to stream something.
Icecast itself can work with Jack. Or you can use one of the other
Hi!
I always did something like this:
~/.asoundrc or /etc/asound.conf [snippet]
pcm.!default
{
type plug
slave
{
pcm "hw:1"
}
}
That's if your card is the second. You could use type hw, probably, but I
never really experimented much in the last years.
Kindest regards
Hi!
I don't know, which one my friend tried first. I think it was the macbook
speakers. We'll go through the rest.
Kindest regards
julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net
the
Hi!
Thanks. I'll have a go at the loopbacks and we'll try the headphones output.
Btw.: I like viola... :-)
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net
the Linux
Hello!
I know there have been many of them. But a friend of mine has a macbook and
trouble getting audio out of his card.
Here's what we got:
Audio device: Intel Coorperation 82801h (ich8 family HD audio controller)
(rev 03)
Subsystem: Apple computer inc. device 00a1
Loaded module: snd_hda_i
Hi!
I guess you work with orca? Ever suspected festival? I had some trouble with
it. But I can't tell you what I did then, because now, I only use JACK. Maybe
that's an alternative?
Configure your asound.conf so, that the "!default" PCM-device is just passed
to JACK and then start JACK. JA
Make it easy. Just turn up the volume of your speakers for these occasions. Or
just switch them off, when you don't need them.
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforg
Hi!
Here's an idea. Does your headset use ALSA as well? I've never worked with
one, so I don't know.
If so, why not use your .asoundrc or asound.conf to create a device, that
talks to both your speakers and your headset? Of course it will always use
both resources then, but it's still wort
Thanks Clemens!
If I'd like to use the "PCM slave timer", how do I access that? Or were the
instructions you gave me at the end of your mail already for that prupose?
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PR
Hello everyone!
I know, tat this is more a question for the devel list, but one more
mailinglist and I'll dy burried in mails. :-(
Does the ALSA timter interface take its timing info from the soundcard
(sample clock) or from the system's timer (RTC)?
If it is the system: Is there a way t
To my knowledge this is possible. Try something like:
arecord -f cd -D PCM_DEVICE -t wav FILES
This is the way it should work. For the -f option of course use your own
format. Try:
arecord --help
to see a list of synonyms for some standard formats.
HTH.
Kindest regards
Julie
John!
with which kernel/ALSA did you test your Delta card?
Another short idea, though perhaps not too helpful taken by itself: Your
problem with the new ALSA seems to be a depencence problem. It looks like your
ALSA needs some other module loaded first, to have some symbols. I don't know:
Hello!
Do you want to reduce/increase the samplerate live? If so alsa just has
plug
devices. Those devices just convert the audio sent to it into one desired
samplingrate, that of the soundcard. It can work like this:
aplay -Dplughw:0,0 file.wav
or
aplay -Dplug:default file.wav
You can c
Hi!
I'd guess there are still two possibilities:
1. You could create a .asoundrc in your home-dir or the equivalent in /etc.
2. You could use the aplay -D syntax:
alsa1 hw:1
Or something similar.
HTH.
Kindest regards
Julien
Music was my first love and it will be my l
Hi Armin!
If your kernel is custom-built anyway, you might get a change. I believe
there was an alsa-option for this and under "Misc devices" or something like
it, far down the list if you use "gmake config" there is another option to
turn on PCspeaker.
Not 100% sure, I never had an HDA in
Hi!
To your second ps (which should be a pps :-) ): I also noticed when doing a
software suspend, that a lot of things are strange, but repeatable.
I don't exactly know: I always compiled my own kernels, my own alsa. But
when I had standard onboard soundcards, I could open alsamixer and see
Hi!
Just ideas:
1. Do you have jack running? (Term # ps-ax | grep jackd)
2. try: term # alsamixer
and then: term # alsamixer -c 1
Perhaps your soundcard is not card 0, but card 1.
3. What your soundcards samplerate setting? What your soundcard anyway?
Perhaps multichannel?
All I can co
Hi!
Your mail arived perfectly. But there doesn't seem to be anyone - at the
moment - who knows about your problem. Unfortunitely, it can take some time
nowadays.
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJ
Hi!
I suppose the best way to start might be to create some pcm-device in your
.asoundrc or asound.conf.
There you know exactly how many channels you have.
Only other thing _I_ can imagine might help is: startup alsamixer, see how
many playback channels you have, look for the setting of
Hi Romeo!
I don't have neither. But I have an M-Audio Delta 1010LT, which I suppose is
RELATIVELY close to the revolution 5.1. If they also use the ICE-chip, then
they are nice. I also remember, that we had a lot of traffic about the
revolution cards here. Since I didn't notice those mails fo
Hi!
I don't know if it fits your server setup, but I think icecast does
something like it. It can split recordings each hour, or whatever you need. It
supports jackd, if you take Carl (or Karl) version. It's the same thing they
used during the linux audio conference.
HTH. For more info and b
Hi!
WHICH is your soundcard?
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
=== AND MY PERSONAL PAGES AT: ===
http://www.julie
Hi!
I think /dev/dsp (or /dev/dsp1) (on my system it's just a link) can open all
10 channels at once. Far as I know delta 1010LT only offers one OSS /dev/dspN
device. Maybe you can try something via asoundrc magic or with the aoss
library, but I never had to use the later one.
Kindest regard
Hi!
Probably you could try the samplerate conversion plugin. I don't know the
exact name, but you can have a look at the alsa page. Look for the wiki main
page and then look for the plugins.
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
=
Hi!
Not really insight suggestions: But if you can negotiate with PCI try the
M-Audio delta series. I've got a delta 1010lt and it works great. There are
smaller models, the delta44 and delta66.
They have XLR, chinch and digital I/O and MIDI in/out.
My card cost about 300 EUR (about 6 years
Hi Matt!
This looks a bit similar to the problem I had. But I'm running 1.0.14. I had
to first update to 1.0.14a and then I had to apply a patch. My problems in
between were different. But it seems to be the same general trouble.
Try looking up something like:
/usr/lib/alsa-lib/libasound_modu
Hi!
I wanna buy a firewire soundcard (or USB if else comes up), that has spdif.
All I really demand is SPDIF, external card and good quality.
Can anyone suggest something that matches this description?
Kindest regards
Julien
Music was my first love and it will be my last (
Wrong format of your audio file? Do you know what your application expects?
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
=== AND MY PE
Hi!
These are just ideas, I really don't know for sure. But maybe it helps...
Try wrapping your devices in plug-devices like this:
pcm.plug_player {
type plug
slave { pcm "audio_player"
}
Or try to remove the ctl-devices. With aplay and other good alsa-programs
they shouldn't be necessary
Sorry, I know it's bad, practice, replying to oneself. But I solved it.
There seem to have been a lot of bugs around 1.0.14 in the alsa-lib. I had
to fix the ./configure script, or better configure.in.
so if anyone's interested, that's the solution.
Kindest regards
Julien
M
Hi everyone!
OK, I solved my "plugindir/libasound_module_pcm_jack.so: no such file or
directory" problem. It was a bug in alsa-lib 1.0.14, which was obviously fixed
in 1.0.14a.
But now I try to use my jack plugin and get a smple format problem.
I start jackd like this:
jackd -R -d alsa -d h
Hi!
If I remove or change my .asoundrc to exclude the jack pcm plgin everything
works well. The problem lies with the extra plugins only. I tried to use
plughw, but that didn't work at all. It said the same only about
"libasound_module_pcm_plughw.so' and that really does not exist. Was it
rem
Hi everyone!
Sorry for ross-posting. But I need help urgently!
so here's my problem. I'm running alsa 1.0.14 (driver/lib/plugins/utils).
But with every app I tried, I get the following error:
ALSA lib pcm.c:2105:(snd_pcm_open_conf) Cannot open shared library
plugindir/libasound_module_pcm_jac
Hi everyone!
I've got a small problem Just instaled alsa-plugins-1.0.14, which fits in
version to the rest of my alsa-stuff. I compiled it without a problem and
installed it. But now I always get:
ALSA lib pcm.c:2105:(snd_pcm_open_conf) Cannot open shared library
plugindir/libasound_module_pcm
Hi!
As I remember:
hw directly accesses the card, which does plughw. But: If you have say a
four-channel card configured at 48kHz, the player will try to open four
channels at 48kHz.
Where plughw, does some automatic conversion. i don't know, if it also looks
up the number of channels, and o
Hi!
I may be mistaken, but I believe that the plug or hwplug alsa-plugin does
samplerate conversion, that's alsa-internal. I don't know about real hardware
sr-conversion.
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB
Hi Takashi!
Sorry it took me so long, but I had to recompile the kernel. and I still
have the low volume problem. Here's what you asked me for.
Running kernel: linux-2.6.21 (mm2 patches)
alsalib/utils/tools: 1.0.13
Following is my asound.state, amixer -c 1 scontents and amixer -c 1
conten
Hi!
This problemk seems not too dificult to me. Try the following:
cd /etc
cp asound.state asound.state.mysave
Then restart your alsa completely. It will have not sound-settings at all,
but that's to be expected. Use your favourite mixer to set all the controls
right again.
You problem come
Hi!
I suppose it's not expected to work exactly like that. I don't know much
about it, have never done much with alsa-programming. But I think it waits for
sound-input from your soundcard not another asa-program. So try to plug some
sound-source into your soundcard's input-jack and see what ha
Hi!
I haven't looked at your listing of controls, but the following might help:
a) if you can see use envy24control, which is - I believe - in the alsa-tools
or alsa-utils package.
b) If you can't see or don't want a graphic mixer, take alsamixer and try f4
or f5 which changes "views". - that m
Hi Takashi!
But still I used (tried to use) those kernels (and alsa-versions) myself,
and volume was really much too low! And I am aware of at least three, who have
the same problem.
So is the volume range (up to 12DB) the old range? How can I set maximum
gain in my alsamixer (which of the e
Hi!
The problem is unfortunitely known. In alsa 1.0.12, or so, the removed
support for the ak4xxx (ipga analog volume). This means, you might be abel to
record, but volume is too low.
There was a thread on alsa-devel about that. So best solution for you is:
stay with kernel 2.6.17 or try to
Hi Steve!
I use fluidsynth with my delta 1010lt. I had to create several pcm-devices
for the task. Each pcm-device held two outputs. I had to wrap some other
(whplug I believe) device around everyone of them. So I had my original out1,
out2,... and my wrapped po1, po2,... devices.
But the ni
Hi!
It should still capture under 2.6.21, but much quieter. The delta1010LT has
some analogue ampsd (IPGA analogue volume), which the alsa-crew decided to not
support any longer. There was some discussion about good quality and extra
gain reducing it... Don't ask me why they really let it disa
Jerry!
I'd treally try the pcm-device thing, because mplayer seemed to be picky
about formats of the audio-files and supported soundcard formats. This seems
to be a special problem with alsa. As you experienced oss is no dificulty as
well as jack.
Kindest regards
Julien
Music
Hi Jerry!
How many channels can your soundcard play? What your default samplingrate? I
had similar problems with mplayer, with my multichannel card. I had to create
a .asoundrc and create a 2-channel pcm device, which I gave to mplayer like
that:
mplayer -ao alsa,o1 $FILE
Best thing if you c
Hello!
I think this is a question for Jaroslav or perhaps Takashi. I have a problem
with my alsa-driver for the delta1010lt card. My IPGA-analog-controls don't
get included. It worked with alsa 1.0.11rc2 and failed with 2.6.20 kernel
(Don't know, which alsa is included there). Also failed to w
Hi Stan!
I use my older kernel at the moment it.s 2.6.16.2 and it works. I always use
the alsa, that comes with the kernel. I build them manually.
Of course, when I installed the 2.6.20 kernel, I installed the correspondig
alsa-* source packages. But I left them installed and returned to my o
Hi Stan!
No I can't use the envy24control, for I'm blind. But I tried just adding the
IPGA analog volume to asound.state. It failed, telling me, that there was some
unexpected control.
I tried copying an earlier asound.state. Inserting the IPGAs in a newly
generated asound.state, at the old
Greetings!
I still have a problem with my 2.6.20 kernel and my delta 1010lt.
Alsamixer doesn't show my ipga analog volume, but that it didn't do for a
longer time now. But now ipga analog doesn't even appear in my asound.state.
RESULT: Especially analog input volume is TOO LOW! Is there a fix
Hello!
I've got a problem with my delta 1010 LT with kernel 2.6.20
(alsa-1.0.14rc1).
The IPGA analog volume has gone. All inputs have too low volume! It doesn't
even appear in /etc/asound.state. Everything else seems to be present as
before.
I need some good advise here, please! - Btw.: Ca
Hello!
I've still got the same problem. ALSA 1.0.14rc1, built-in of Linux kernel
2.6.20 doesn't recognise my IPGA-analog volume. It doesn't even show up in
/etc/asound.state. But even before with linux-2.6.16.2 it didn't show
IPGA-analog volume in the mixer. So I had to set it by hand, or the
Hi!
I've got a VIA 8235 running alinux kernel 2.6.16.13-4, with
alsa Version 1.0.11rc3
It's giving me trouble recording. Either it is much to loud or much to low,
but nothing in between.
Below is my /etc/asound.state.
Please can anyone help me? I tried to mute/unmute everything in a franzy
Hello!
I've still got the same problem. ALSA 1.0.14rc1, built-in of Linux kernel
2.6.20 doesn't recognise my IPGA-analog volume. It doesn't even show up in
/etc/asound.state. But even before with linux-2.6.16.2 it didn't show
IPGA-analog volume in the mixer. So I had to set it by hand, or the
Hello!
I've still got the same problem. ALSA 1.0.14rc1, built-in of Linux kernel
2.6.20 doesn't recognise my IPGA-analog volume. It doesn't even show up in
/etc/asound.state. But even before with linux-2.6.16.2 it didn't show
IPGA-analog volume in the mixer. So I had to set it by hand, or the
Hi!
I just installed the Linux-kerlen version 2.6.20 with built-in alsa
1.0.14rc1. But now I have the problem, that alsa doesn't knwo my ipga analog
volume anymore. This happened once before with 0.9.4 something. I also
installed the correct alsalib/utils/tools.
Even the asound.state doesn't
Hi!
I just recently had the following problem. I used jack-only for some time
and then tried alsa again, but what ever software I tried (aplay, ecasound,
mplayer) with my good old .asoundrc, it told me something like:
alsalib some error with pcm_conf
I used my .asoundrc up to kernel 2.6.15 wi
Hi!
This seems to be a problem, not unheard of. I think it depends on your
hda-intel version and where it belongs to, if it works or not. They are still
working on it, as I heard. I had that problem myself. Perhaps try finding out
what you can about your special hda (version with lspci), where it
Hi!
I have a soundblaster live 1024. And when I reboot the system and have a
look in alsamixer the master and pcm are muted. What can I do against this?
Info: System is suse 9.3 original.
Thanks for any help!
Kindest regards
Julien
Music was my first love and it will be m
HI!
alsa 1.0.10 doesn't work with 2.6.15.x. Does this mean I HAVE to use
alsa-lib/utils/tools 1.0.10rc3?
Kindest regards and thanks
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net - the Li
Hi!
I don't have it installed yet and can't unpack on this computer. I'd just
like to download the correct lib/utils...
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net - t
Hi!
I've just installed SuSE 10.0 on a notebook with a hda-intel soundcard. But
I get no sound. I tried to run the alsa-conf script, but it didn't change
anything. The real problem I get is when I call alsamixer or alsactl store.
$ alsamixer:
alsamixer: function snd_mixer_load failed: Invalid
Hi!
When I try the new kernels (later than 2.6.11) I get my jack-problems, which
still aren't resolved.
And I can't use envy24control, for I'm blind. But alsamixer worked before. I
remember that Takashi fixed it once (in 0.9.6 or something) and from then on
no problem.
By the way, I used corr
Hi everyone!
I still do have the same problem. My alsamixer is buggy and I can't fix that
problem myself. Also I still can't set the ipga-analog volumes, but I couldn't
do that since 2.6.10 kernels. Here I gathered all info I thought necessary
(and perhaps more than that :-))
uname -a:
Linux bac
Hi!
I've discovered a bad bug with my alsa and my delta 1010lt. The input volume
was much to low for a long time, but I thought it was the cable, because a pin
was a bit disfigured. Now I found out it was alsa.
Alsamixer doesn't display the ipga-analog volume controls and it crashes
with a frie
Hi Clemens!
The I/Os are supposed to be availabe for the PC. At least under Windows,
that's what it's all for.
For the rest, I mail back.
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb
Hi!
I've got a USB audio/midi device (Roland U8).I connected it and alsa
recognised the midi-port immediately. But no audio. NOt listed anywhere
(proc/asound).
The device is kind of a harddisk recorder - without a disk. It has its own
control-elements and onboard-FX. I don't need to rely on the
Hi!
I've just connect a Roland U8 usb device to my computer. Midi worked fine
from the beginning, but no audio devices were listed. The U8 is a kind of
hard-disk recorder without any disk. It has effects, control-elements and
offers some analog/digital i/os.
Does anyone have an idea about this?
Hi!
Did you try to use a simple dshare or something and then use bindings.
It could look like that:
pcm.my_40 {
type dshare
slave "hw:0,0"
bindngs {
0 0
1 1
2 0
3 1
}
ipc_key 1738
}
I'm not sure about the bindings section. I don't know which channel to bind
to wh
Hi!
The other card is not running, because it can't be loaded. Reading the
alsa-docs it probably should... It is still sutrned on in BIOS.
The lsmod looks good. I haven't got it here and can't get it now. But: All
alsa-modules, which should be there, are loaded. As I remember (correctly...
hope
Hi Folks!
I've got trouble with my alsa0.9.6 and my sblive. Alsamixer gives me a
warning,
but aplay seems to act normally. I got no sound though. Here are some infos
(lspci, /proc/interrupts alsamixer warning and uname -a):
00:00.0 Host bridge: nVidia Corporation nForce2 AGP (different version?)
Hi everyone!
I have soundcard trouble with a notebook. Here are the facts:
The soundcard is onboard (the chipset seems to be Intel i845GE). The linux
running is a mandrake 9.2 (fiveStar), not touched since the original
installation. On boot-up I get the following:
"no soundcards found".
Does
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