Sound modules missing/broken [solved]

SOLVED:
I stupidly put this module into modprobe.conf with an index=0 parameter. I should have picked up on that quicker from the dmesg output.
Hi all, I'm posting this in the Kernel section because it appears to be an issue with loading modules and not ALSA itself.
Yesterday I installed from scratch with the new 2009.08 cd and got everything working just fine except sound. My USB headphones work great, but I'd much prefer to use my speakers during the daytime. The trouble is, I can't get the correct kernel modules to load.
twilkins ~ $ lspci | grep Audio
00:0f.1 Audio device: nVidia Corporation MCP55 High Definition Audio (rev a2)
This onboard audio is supposed to (and has in the past) work with the snd-hda-intel module. However, I have not been able to get it loaded:
twilkins ~ $ lsmod | grep snd
snd_seq_dummy 3540 0
snd_seq_oss 36224 0
snd_seq_midi_event 8592 1 snd_seq_oss
snd_seq 64640 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
snd_seq_device 8324 3 snd_seq_dummy,snd_seq_oss,snd_seq
snd_pcm_oss 47328 0
snd_mixer_oss 20976 1 snd_pcm_oss
snd_hwdep 9976 0
snd_pcm 90872 1 snd_pcm_oss
snd_timer 25344 2 snd_seq,snd_pcm
snd 76744 8 snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_hwdep,snd_pcm,snd_timer
soundcore 8576 1 snd
snd_page_alloc 10784 1 snd_pcm
In fact, it isn't even in the holders directory:
twilkins ~ $ ls -l /sys/module/snd/holders/
total 0
lrwxrwxrwx 1 root root 0 2009-08-11 11:47 snd_hwdep -> ../../snd_hwdep
lrwxrwxrwx 1 root root 0 2009-08-11 11:47 snd_mixer_oss -> ../../snd_mixer_oss
lrwxrwxrwx 1 root root 0 2009-08-11 11:47 snd_pcm -> ../../snd_pcm
lrwxrwxrwx 1 root root 0 2009-08-11 11:47 snd_pcm_oss -> ../../snd_pcm_oss
lrwxrwxrwx 1 root root 0 2009-08-11 11:47 snd_seq -> ../../snd_seq
lrwxrwxrwx 1 root root 0 2009-08-11 11:47 snd_seq_device -> ../../snd_seq_device
lrwxrwxrwx 1 root root 0 2009-08-11 11:47 snd_seq_oss -> ../../snd_seq_oss
lrwxrwxrwx 1 root root 0 2009-08-11 11:47 snd_timer -> ../../snd_timer
Trying to load it manually will not work either:
root ~ # modprobe snd-hda-intel
FATAL: Error inserting snd_hda_intel (/lib/modules/2.6.30-ARCH/kernel/sound/pci/hda/snd-hda-intel.ko): Unknown symbol in module, or unknown parameter (see dmesg)
root ~ # dmesg | tail
[snipped useless info]
snd_hda_codec: Unknown parameter `index'
Alsaconf correctly recognizes the card and to use snd-hda-intel. When it completes it says it is ready to go, but it is lying and nothing has changed.
Can anyone help me with this?
Last edited by Mais (2009-08-11 18:57:02)

how did you re-install the kernel? I've never done that, and am too experiencing sound issues.
--nixIT

Similar Messages

[Solved] Strange error with sound modules

If i modprobe any sound modules, i get strange errors which tell me to lookup dmesg.
The error output looks like
WARNING: Error inserting snd_mixer_oss (/lib/modules/2.6.13-ARCH/kernel/sound/core/snd-pcm.ko): unknown symbol in module, or unknown parameter (see dmesg)
FATAL: Error inserting snd_pcm_oss (/lib/modules/2.6.13-ARCH/kernel/sound/core/oss/snd-pcm-oss.ko): Unknown symbol in modules or unknown parameter (see dmesg)
The same for my soundcard drivers etc, simply any alsa module.
My modprobe.conf looks as follows.
install snd-pcm modprobe -i snd-pcm; modprobe snd-pcm-oss; true
install snd-seq modprobe -i snd-seq; modprobe snd-seq-oss; true
options snd device_mode=666
alias snd-card-0 snd-ens1397
alias sound-slot-0 snd-ens1371
Well, this should work fine (as i know). Looking up dmesg i get the following:
snd_pcm_oss: Unknown symbol snd_info_unregister
snd_pcm_oss: Unknown symbol snd_info_get_line
snd_pcm_oss: Unknown symbol snd_pcm_format_width
I've got this problem since i updated to 2.6.13-ARCH.
Since this problem causes hotplug to take 10 minutes to start, it's annoying. I'd appreciate any hint what i could try.
Thank you,
STiAT

That did solve my problem.
Thank you alot!
STiAT

[SOLVED] No sound modules after kernel update

Hello,
after updating my kernel, I have no sound modules. Alsaconf can't detect card. My lspci sound card:
00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06)
My lsmod:
Module Size Used by
joydev 9895 0
usbhid 35352 0
hid 82435 1 usbhid
nvidia 12069499 40
broadcom 7270 0
tg3 133684 0
ehci_hcd 40794 0
intel_ips 11245 0
psmouse 55501 0
libphy 16985 2 broadcom,tg3
i7core_edac 16231 0
serio_raw 4294 0
i2c_i801 8187 0
pcspkr 1819 0
evdev 9530 10
usbcore 144144 3 usbhid,ehci_hcd
edac_core 37552 3 i7core_edac
i2c_core 20460 2 nvidia,i2c_i801
sg 25429 0
iTCO_wdt 11789 0
button 4470 0
iTCO_vendor_support 1961 1 iTCO_wdt
processor 24224 0
fuse 66858 2
ext4 375413 3
mbcache 5881 1 ext4
jbd2 70667 1 ext4
crc16 1359 1 ext4
raid0 7958 2
md_mod 100392 3 raid0
sr_mod 14951 0
cdrom 36553 1 sr_mod
sd_mod 28275 8
pata_acpi 3376 0
ata_piix 22077 6
libata 166724 2 pata_acpi,ata_piix
scsi_mod 132794 4 sg,sr_mod,sd_mod,libata
And my kernel:
Linux nekroman 3.1.2-1-ARCH #1 SMP PREEMPT Tue Nov 22 09:17:56 CET 2011 x86_64 Intel(R) Core(TM) i5 CPU 760 @ 2.80GHz GenuineIntel GNU/Linux
Thanks for help.
//edit:
Reinstalled kernel and it works!
Last edited by Nekroman (2011-11-25 15:08:32)

how did you re-install the kernel? I've never done that, and am too experiencing sound issues.
--nixIT

How to reload sound modules?

I'm using alsa + pulse sound system. I also have Openbox as a stand-alone WM and lightDM, but the same thing occurs on my debian testing + gnome. I've been trying to solve this issue for several days but with no success.
Below is the output from alsa-info.sh script
http://www.alsa-project.org/db/?f=9f32f … f7cdddcc40
Installed packages:
morfik:~$ pacman -Qs alsa
local/alsa-lib 1.0.27-1
An alternative implementation of Linux sound support
local/alsa-plugins 1.0.27-1
Extra alsa plugins
local/alsa-utils 1.0.27-4
An alternative implementation of Linux sound support
local/pulseaudio-alsa 2-2
ALSA Configuration for PulseAudio
morfik:~$ pacman -Qs pulse
local/libao 1.1.0-3
Cross-platform audio output library and plugins
local/libcanberra-pulse 0.30-3
PulseAudio plugin for libcanberra
local/libpulse 3.0-3
A featureful, general-purpose sound server (client library)
local/paprefs 0.9.10-2
A simple GTK-based configuration dialog for PulseAudio
local/pavucontrol 2.0-1
A GTK volume control tool for PulseAudio
local/pulseaudio 3.0-3
A featureful, general-purpose sound server
local/pulseaudio-alsa 2-2
ALSA Configuration for PulseAudio
Everything works fine until hibernation. After my system goes up, one speeker (of 4) is missing, I mean that 3 others play sound well, but the rear-left is silent. I don't know what is wrong, but on debian I can deal with it by reloading sound modules:
$ pulseaudio -k
# alsa force-reload
$ pulseaudio -D
But unfortunately there is no such a thing like "alsa" in Arch. When I type alsa force-reload on my debian, I get this:
root:/# alsa force-reload
Unloading ALSA sound driver modules: snd-ca0106 snd-seq-midi snd-seq-midi-event snd-rawmidi snd-ac97-codec snd-pcm snd-page-alloc snd-seq snd-seq-device snd-timer.
Loading ALSA sound driver modules: snd-ca0106 snd-seq-midi snd-seq-midi-event snd-rawmidi snd-ac97-codec snd-pcm snd-page-alloc snd-seq snd-seq-device snd-timer.
I tried to change a little pulse settings in /etc/pusle/daemon.conf but this does nothing. Now the file looks like this:
daemonize = yes
allow-exit = yes
use-pid-file = yes
exit-idle-time=0
resample-method=speex-float-0
default-sample-rate = 44100
alternate-sample-rate = 48000
default-sample-channels = 4
default-channel-map = front-left,front-right,rear-left,rear-right
I can't use hibernation because after my system comes to life, I have to reboot it in order to make sound work properly.
I don't know what info I should provide, so if you want to know something more about configuration of my pc, feel free and ask, I'll post the info as soon as possible.

Raynman -- thanks for the hint. I looked for the alsa script on debian, and it exists, so I moved it to arch.
This is the script:
#!/bin/sh
# alsa-base control script
# Description: Used to load and unload ALSA modules and
# restore and store mixer levels. There is no
# longer any need to run this script on bootup
# or shutdown. It is now moved to /usr/sbin.
set -e
# Exit if alsa-base package is not installed
[ -f /etc/modprobe.d/alsa-base.conf ] || exit 0
MYNAME=/usr/sbin/alsa
PATH=/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin
# Default values of variables in /etc/default/alsa
force_unload_modules_before_suspend=""
[ -f /etc/default/alsa ] && . /etc/default/alsa
# $* MESSAGE
warn() { echo "${MYNAME}: Warning: $* " >&2 ; }
# Attempt to create /var/run/alsa if it is absent.
# Return true if /var/run/alsa exists after this attempt,
# otherwise false.
check_run_dir()
[ -d /var/run/alsa ] && return 0
# We have no business creating /var/run if it doesn't exist
if ! [ -d /var/run ] ; then
warn "Could not create /var/run/alsa/ because /var/run/ is not present."
return 1
fi
if ! mkdir --mode=755 /var/run/alsa ; then
warn "Failed to create /var/run/alsa/."
return 1
fi
[ -d /var/run/alsa ] && return 0
return 1
echo_procs_using_sound()
for i in /proc/[0-9]*/fd/* ; do
var="$(readlink $i)"
if test x"$var" != x"${var#/dev/snd/pcm}" ; then
IFS=/; set -- $i; unset IFS; echo $3
fi
done
# $* [PID]...
echo_with_command_names()
[ "$1" ] || return 0
echo $( \
ps --no-headers -o "%p %c" "$@" \
| sed -e 's/$[0-9][0-9]*$ $.*$/\1(\2)/' \
kill_procs_using_sound()
procs_using_sound="$(echo_procs_using_sound)"
if [ "$procs_using_sound" ] ; then
echo -n "Terminating processes:"
for attempt in 1 2 3 4 ; do
echo -n " ${procs_using_sound}"
kill $procs_using_sound || :
sleep 1
procs_using_sound="$(echo_procs_using_sound)"
[ "$procs_using_sound" ] || break
done
# Either no more procs using sound or attempts ran out
if [ "$procs_using_sound" ] ; then
echo -n " (with SIGKILL:) ${procs_using_sound}"
kill -9 $procs_using_sound || :
sleep 1
fi
procs_using_sound="$(echo_procs_using_sound)"
if [ "$procs_using_sound" ] ; then
echo " (failed: processes still using sound devices: $(echo_with_command_names $procs_using_sound))."
return 1
fi
echo "."
fi
return 0
# $* MODULE-NAME [MODULE-NAME]... | "all"
unload_modules()
procs_using_sound="$(echo_procs_using_sound)"
if [ "$procs_using_sound" ] ; then
warn "Processes using sound devices: $(echo_with_command_names $procs_using_sound)."
fi
if check_run_dir ; then
:> /var/run/alsa/modules-removed
else
warn "Not keeping list of removed modules because /var/run/alsa is absent.
It will not be possible automatically to reload these modules."
fi
echo -n "Unloading ALSA sound driver modules:"
[ -d /proc/asound ] || { echo " (none loaded)." ; return 0 ; }
echo_snd_modules_loaded()
lsmod \
| sed -n -e 's/^$snd[-_][^[:space:]]*$[[:space:]].*/\1/p' \
| sed -e 's/_/-/g'
for FSMBS in $* ; do
MODULES_TO_REMOVE=""
SND_MODULES_LOADED="$(echo_snd_modules_loaded)"
case "$FSMBS" in
all)
MODULES_TO_REMOVE="$SND_MODULES_LOADED"
snd_*|snd-*)
FSMBS="$(echo "$FSMBS" | sed -e 's/_/-/g')"
for M in $SND_MODULES_LOADED ; do
if [ "$FSMBS" = "$M" ] ; then
MODULES_TO_REMOVE="$FSMBS"
break
fi
done
esac
[ "$MODULES_TO_REMOVE" ] || continue
if [ -d /var/run/alsa ] ; then
echo "$MODULES_TO_REMOVE" >> /var/run/alsa/modules-removed
fi
for M in $MODULES_TO_REMOVE ; do
echo -n " ${M}"
modprobe -r "$M" >/dev/null 2>&1 || :
done
done
if [ -f /var/run/alsa/modules-removed ] ; then
MODULES_STILL_LOADED="$(echo_snd_modules_loaded | grep -F -f /var/run/alsa/modules-removed)"
MODULES_STILL_LOADED="$(echo $MODULES_STILL_LOADED)"
else
MODULES_STILL_LOADED=""
fi
if [ "$MODULES_STILL_LOADED" ] ; then
echo " (failed: modules still loaded: ${MODULES_STILL_LOADED})."
return 1
else
echo "."
return 0
fi
# $* MODULE-NAME [MODULE-NAME]... | "all"
force_unload_modules()
kill_procs_using_sound || :
unload_modules "$@" || return 1
return 0
load_unloaded_modules()
LUM_RETURNSTATUS=0
MODULES_TO_LOAD=""
[ -d /var/run/alsa ] || warn "Directory /var/run/alsa is absent."
echo -n "Loading ALSA sound driver modules:"
[ -f /var/run/alsa/modules-removed ] && MODULES_TO_LOAD="$(echo $(cat /var/run/alsa/modules-removed))"
[ "$MODULES_TO_LOAD" ] || { echo " (none to reload)." ; return $LUM_RETURNSTATUS ; }
echo -n " $MODULES_TO_LOAD"
for MDL in $MODULES_TO_LOAD ; do
modprobe $MDL || LUM_RETURNSTATUS=1
done
case "$LUM_RETURNSTATUS" in
0) echo "." ;;
*) echo " (failed)." ;;
esac
return $LUM_RETURNSTATUS
case "$1" in
unload)
unload_modules all || exit $?
reload)
EXITSTATUS=0
unload_modules all || EXITSTATUS=1
load_unloaded_modules || EXITSTATUS=1
exit $EXITSTATUS
force-unload)
force_unload_modules all || exit $?
force-reload)
EXITSTATUS=0
force_unload_modules all || EXITSTATUS=1
load_unloaded_modules || EXITSTATUS=1
exit $EXITSTATUS
suspend)
case "$force_unload_modules_before_suspend" in
""|false) : ;;
all|true) force_unload_modules all || exit $? ;;
*) force_unload_modules $force_unload_modules_before_suspend || exit $? ;;
esac
resume)
case "$force_unload_modules_before_suspend" in
""|false) : ;;
*) load_unloaded_modules || exit $? ;;
esac
echo "Usage: $MYNAME {unload|reload|force-unload|force-reload|suspend|resume}" >&2
exit 3
esac
It requires two other files in /etc/modprobe.d/ . I also moved them from debian:
alsa-base-blacklist.conf
# Uncomment these entries in order to blacklist unwanted modem drivers
# blacklist snd-atiixp-modem
# blacklist snd-intel8x0m
# blacklist snd-via82xx-modem
# Comment this entry in order to load snd-pcsp driver
blacklist snd-pcsp
alsa-base.conf
# autoloader aliases
install sound-slot-0 /sbin/modprobe snd-card-0
install sound-slot-1 /sbin/modprobe snd-card-1
install sound-slot-2 /sbin/modprobe snd-card-2
install sound-slot-3 /sbin/modprobe snd-card-3
install sound-slot-4 /sbin/modprobe snd-card-4
install sound-slot-5 /sbin/modprobe snd-card-5
install sound-slot-6 /sbin/modprobe snd-card-6
install sound-slot-7 /sbin/modprobe snd-card-7
# Cause optional modules to be loaded above generic modules
install snd /sbin/modprobe --ignore-install snd && { /sbin/modprobe --quiet snd-ioctl32 ; /sbin/modprobe --quiet snd-seq ; : ; }
install snd-rawmidi /sbin/modprobe --ignore-install snd-rawmidi && { /sbin/modprobe --quiet snd-seq-midi ; : ; }
install snd-emu10k1 /sbin/modprobe --ignore-install snd-emu10k1 && { /sbin/modprobe --quiet snd-emu10k1-synth ; : ; }
# Keep snd-pcsp from beeing loaded as first soundcard
options snd-pcsp index=-2
# Keep snd-usb-audio from beeing loaded as first soundcard
options snd-usb-audio index=-2
# Prevent abnormal drivers from grabbing index 0
options bt87x index=-2
options cx88_alsa index=-2
options snd-atiixp-modem index=-2
options snd-intel8x0m index=-2
options snd-via82xx-modem index=-2
options snd-ca0106 index=0
options snd-hda-intel index=1
After hibernation, the script provides sound as expected. I also notticed that I can reload only snd-ca0106 and the sound comes back.
Now I have to make a .service file that could make my sound work. But I need more info how to do so.

Where are my sound module preset names in the track list?

I've been using Logic since version 2.5 and, since I have several MIDI sound modules whose sounds I'm perfectly happy with, I am now using those sound modules with Logic 8.
But having just upgraded to 8 from 7, I've discovered something I think is missing, which is a bit of an annoyance.
In all previous versions of Logic, if you set up a instrument in the Environment for a MIDI sound module and then added all the preset names when you set up the bank/program change commands, when you select a preset from the track list in the Arrange window, the name in the track list changed from the name of the instrument to the name of the preset.
For example, the name would change from 'Fantom XR' (the name I'd called the instrument in the Environment) to 'PNO SoTrue' (the name of the preset) in the track list when I selected that preset via Logic.
However, in Logic 8, when I select a preset, the name on the track list now stays as 'Fantom XR' (or whatever the name of the instrument is) and doesn't change to that of the preset.
Given that I use six MIDI modulss, keeping track of the sounds by having the trak list name change to that of the presets was really useful for keeping track of what was what.
Have I just missed something with how to do it, or is a permanent change? I really hope not.

No, it's still there, but the interface has changed.
Right-click on a track and select "Configure Track Header"
In the box that comes up, enable the checkbox under "Names" and next to it (the second flip menu), select "Software Instrument Setting Name".
Now the program names in your multi-instruments are displayed in the manner you have been accustomed to.
As ever, save this stuff in your autoload to always have you own configs available.
You have some options there in that the two flip menus determine the first and second displayed track name, so you can set up how you like - it's more flexible than earlier versions of Logic.

Adding a sound module using a second midi interface

In the arrange window, clicking and holding on an instrument icon produces a pull down menu which includes "MIDI instr" with a submenu displaying the three external midi instruments that are connected via USB and a 4X4 midi interface. We also use a 4th. midi sound module connected via a second USB cable directly to the computer. PROBLEM: The 4th. module does not appear in the submenu along with the other three instruments. It does appear as a selectable "Port" in the track information box on the far left side of the arrange window and when selected it does recieve midi data from Logic. Unfortunately, selecting it there also changes the port that one of the other three instruments is assigned to. I believe if I can get it to show in the submenu list with the other three it will solve our prolem.
G4 powerbook Mac OS X (10.4.5)
Mac OS X (10.4.5)
Mac OS X (10.4.5)

In case Jos's assistance doesn't resolve it, maybe this will: Using the example you gave
for example: "c:\my file system"--->"first_sub_directory"--->second_sub_directory--->"HelloWorld.java"
and assuming that you aren't using a "package ..." statement in HelloWorld.java, and also assuming that the HelloWorld.class file is in second_sub_directory , then you need to mount second_sub_directory in the Explorer. You always mount the parent of the class - or if a package is specified, the parent of the package.

Arch64 sometimes "forgets" sound card in ALSA [solved]

Hi all,
in order to some other issues with my firstly installed SB Live soundcard (disturbing sounds when nothing is played back) I played around with two other soundcards. One was a Soundblaster USB MP3+ and the other one a C-Media card based on the CM8738 chip.
Since I am using ALSA I was using the wiki guide to set up the cards:
http://wiki.archlinux.org/index.php/Alsa
Loading and storing the mixer level works normally fine but sometimes after a reboot arch seems to forget the default soundcard used at this time. The ALSA mixer within gnome simply shows my ATI R600 HDMI as "soundcard".
Similar to this is the output of aplay -l
**** Liste der Hardware-Geräte (PLAYBACK) ****
Karte 0: HDMI [HDA ATI HDMI], Gerät 3: ATI HDMI [ATI HDMI]
Sub-Geräte: 1/1
Sub-Gerät #0: subdevice #0
If I try verify that udev autodetected all my sound cards properly according to the wiki, then everything looks find to me:
[hb@hb-arch64 ~]$ ls -l /sys/module/snd/holders
insgesamt 0
lrwxrwxrwx 1 root root 0 24. Aug 21:31 snd_cmipci -> ../../snd_cmipci
lrwxrwxrwx 1 root root 0 24. Aug 21:31 snd_hda_codec -> ../../snd_hda_codec
lrwxrwxrwx 1 root root 0 24. Aug 21:31 snd_hda_intel -> ../../snd_hda_intel
lrwxrwxrwx 1 root root 0 24. Aug 21:31 snd_hwdep -> ../../snd_hwdep
lrwxrwxrwx 1 root root 0 24. Aug 21:31 snd_mixer_oss -> ../../snd_mixer_oss
lrwxrwxrwx 1 root root 0 24. Aug 21:31 snd_mpu401_uart -> ../../snd_mpu401_uart
lrwxrwxrwx 1 root root 0 24. Aug 21:31 snd_opl3_lib -> ../../snd_opl3_lib
lrwxrwxrwx 1 root root 0 24. Aug 21:31 snd_pcm -> ../../snd_pcm
lrwxrwxrwx 1 root root 0 24. Aug 21:31 snd_pcm_oss -> ../../snd_pcm_oss
lrwxrwxrwx 1 root root 0 24. Aug 21:31 snd_rawmidi -> ../../snd_rawmidi
lrwxrwxrwx 1 root root 0 24. Aug 21:31 snd_seq -> ../../snd_seq
lrwxrwxrwx 1 root root 0 24. Aug 21:31 snd_seq_device -> ../../snd_seq_device
lrwxrwxrwx 1 root root 0 24. Aug 21:31 snd_seq_oss -> ../../snd_seq_oss
lrwxrwxrwx 1 root root 0 24. Aug 21:31 snd_timer -> ../../snd_timer
snd_hda_codec is the driver used for my ATI HDA audio codec integrated on the mainboard. snd_cmipci is the driver for the C-media card.
But somehow the module isn't even correctly loaded although I added it to the modules inside the rc.conf (soundcard driver before oss module).
In dmesg I see no problems indicated and lspci lists my card fine. The strange thing is that I experienced this for all three soundcards I used in the last two days which are in order
1) Soundblaster Live 5.1 (emu10k1 chip)
2) Soundblaster USB MP3+
3) CMedia card
I modified the rc.conf and modprobe.conf after changing the "default" card each time.
Has anyone of you experienced similar problems ?
TIA,
D$
Last edited by Darksoul71 (2010-08-25 21:09:37)

Hm, I solved this now myself although I am unshure what the problem was.
According to the output of ls -l /sys/module/snd/holders all my sound cards were properly detected and the corresponding modules were load correctly.
But if this was true then why didn't the soundcards didn't show up in the alsa or gnome mixer as selectable devices ?
1st I removed every reference to a soundcard driver in /etc/modprobe.d/modprobe.conf
2nd I changed my rc.conf like this:
MODULES=(snd_cmipci radeon snd-pcm-oss !snd_hda_codec !snd_hda_intel)
I explicitely blacklisted my onboard soundcard and the sound module of the ATI HDMI output. Et voilà:
My CMedia card shows up correctly in the mixer.
If I have some time at weekend I will crosscheck back with my Soundblaster USB MP3+.
Last edited by Darksoul71 (2010-08-25 21:14:40)

Can the ipod touch be used as a midi sound module?

I was wondering if the ipod touch can send midi data via the usb connection to a midi controler and act as a sound module using an application such as mini synth? Any answers would be appriciated.

The iPod Touch (and the iPhone) is not designed to be used as a storage device which is why it is missing the option you've seen on regular iPods.

CONNECTION TO THE MODULE IS BROKEN

i am using FP-1600 with DO-AC-240 after two or three write operation to the FP-DO-AC240 module field point explorer give the error "CONNECTION TO THE MODULE IS BROKEN".error message comes with FP-RLY-422 module also.why?

Vinod,
Noticing that you are using both FP-DO-AC240's and FP-RLY-422's, I am assuming that you are attempting to switch AC voltage through the FP-RLY-422 module. I am further assuming that you are controlling an AC motor or solenoid or similiarly high inductive load. With high inductive loads, turning the power off results in a large spike of voltage, generated by the inductor (such as the solenoid) to be driven back along the circuit. These spikes, can have a number of effects on electronics. They can cause the the network module to temporarily lose communications with the I/O module, or in some cases can be so severe as to cause damage to a module.
If you were to measure the control lines with an oscilliscope, you are likely to see spikes that can be mo
re than 1000 Volts. We do have circuitry in the modules to help minimize the spikes (these spikes are known as back-EMF). Occasionally, however, the spikes are so large that our built-in protection is not enough. I recommend using an AC snubbing device such as a Metal Oxide Varistor (MOV) to reduce the back-EMF being seen in your circuit. Reducing the back-EMF will solve the temporary communications loss you are seeing.
Regards,
Aaron

How do I use logic as a mulit timbre sound module???

Whats up people,
Im trying to trigger all the sounds in logic via a hardware controller (mpc) separating everything on separate channels. I got it to read on just one channel at a time, but multi?? Forget about it.... I read other posts in regards to this same question and have run thru the suggestion switched midi setting to auto demix etc...
Doesn't respond if i go to another channel on my hardware and select the new instrument to the same channel in logic it still only allows playback from one channel at a time?? Huh is my theory even possible folks!?? I would think it is.
Im trying to comprehend the instructions in the manual for the environment setup, to see if i need to make any changes their. I thrown in a channel splitter, a seq input, a transformer but i have no idea if im doing it right?? I must be slow cos to me that enviroment chapter reads like riddles! (LAYMAN TERMS SOMEONE PLEASE!!)
If anyone can assist in this id appreciate it. Any and all suggestions are welcome.
Good lookin out
peace
D.

Greetings people,
The latest.............okay where do i start. Well i actually followed everybody suggestion on this thread and actually as a result now understand the enviroment a little better. But i have still been unable to get my multi playback channels. I managed to get access to all 16 channels etc.
Right now i have my physical input going thru the monitor to the channel splitter, each channel is cabled to my audio instrument(which im assuming is the multi inst i created as to represent the mpc) channels are set to all and then i have the out of my audio instrument going to the seq input. I hear my sounds and have actually triggered the internal sounds from the mpc pads no problem (did that b4 i posted)
but now say i select midi channel 1 on my mpc, which in turn is instrument 1, double click it to get the audio window select ultrabeat, go back to the mpc and program some drums using ultrabeats sounds. Now remember all im doing is recording midi signals into my mpc which is triggering logics sounds right. okay so now, im saying i want to go to the next channel/audio instrument/ and now use sculpture and put a nice fat bass on it. so i select the patch i want hit play on my mpc hoping that it plays back the drums, but NO! now i have selected a new track it pulls over the programming i did for ultrabeat into the sculpture patch.....follow me??
I tested logic to just record one at a time in the arrange, and it does but it records real erratic. btw my mpc is strung up thru my midisport usb which i did select in the enviroment window..check!
I tried Mac's method of dragging the audio instrument into the arrange and selecting the track with a corresponding sub channel etc but got allot of midi feedback!!??? I know that going to settings - record-auto demix etc is supposed to do the trick, well it aint for me?? Please what am i doing wrong??
Ultimately i want to be able to program my mpc using logics sounds via the midi channels simultaneously, and then dump it all down into logic once i am happy with the outcome. Am i asking too much for logic to be a sound source and recorder at the same time??
I dont mean to sound naive or ignorant but i have just started using software for the first time. i use to go the hardware route using digital recorders allday but it was a simpler routing method since my sound sources were seperate(keyboards, sound modules etc). obviously now im tryint to intergrate my old style with the new, but im getting very fustrated and creativley lathalgic from this setback. Many have suggested tossing the mpc and using logic for drums, but sorry i just cant do it i love that boom bap to much.
Anway sorry for the rambling its late and ive just spent the last 5 hours beating my head upside the g5. If anyone out their hasd a similar setup to me whether successful or not, get at me lets trade off techniques and ideas.
Thanks again for taking the time to read the thread and offering your help, please keep it coming.
peace
D.

Using External Sound Modules (Synths) in Logic Express?

Hi all,
I don't have Logic Express yet. I want to know if what I want to do is possible before I purchase. It may sound trivial, but I'm very new to the whole DAW environment. Apparently Garageband can't support my external sound modules so I'm hoping Logic Express will.
To help, please see my hardware midi setup diagram here:
http://www.greddin.com/images/My-Midi-Setup.jpg
Will I be able to add tracks that will play the external sound devices like the Wavestation and JD-990? Can you change the sounds through Logic? To what extent can I use my external sound modules?
Thanks all,
-Greg

Hi,
1st: Yes, Logic Express has "external MIDI" tracks that function similar to the "software instrument" tracks but trigger external instruments.
2nd: But: To setup these tracks, especially with more than one device, needs some experience with MIDI, the channels, patch selection via MIDI etc. The real downside is you have to find out which commands you need to "tell" your devices to change sounds and these things. Since there is no standard command for this, you'll have to look it up in your devices manual and use the MSB/LSB Values- which you have to enter manually. However, once you found it out, you can save these things as templates and use them in the future. So yes, it is very possible, but expect some learning curve.
3rd: You can use your sound modules to their full extent. Logic features a mighty MIDI environment setup once you learned how to use it.
Fox

Hooking up sound modules through MIDI Controllers

I have an Axiom 25 MIDI Controller
and an Alesis QSR sound module.
how can i hook the module up to the axiom then get the sounds out through garageband. I did this once years ago and i just can't remember for the life of me. I have instrument cables, midi cables. I guess i'm not plugging them in in the correct order. This sounds really noobish of me, i know, but i could desperately use some help here. Thanks!

Hi. The only way to get the sounds from the QSR into GarageBand is to record the audio out from the module. You will need to go from MIDI out (using the 5pin MIDI connection not USB) on the Axiom to MIDI in on the QSR. You need to make sure that the QSR is listening to the MIDI channel the Axiom is sending on. You will then need to go from the Main L/R Audio outs on the Alesis into 'something' that GB can listen to. If you have an Audio interface then all good. Set that up as your Input Source in GB's preferences, create a Real Instrument Track, set up the correct input source/channels in GB's info pane and your good to go. If you do not have an Audio interface and are going straight into your computers built in input, then you will need a cable that is suitable to connect the outs from the QSR to the single 3.5mm-1/8" jack that your computer has.
All the best

External sound module

Is it possible to connect an external sound module to iMac/Garageband so as not to use the quicktime software sounds?
iMac Mac OS X (10.4.5)

I don't understand that question! GB has midi input, but that doesn't help if you want to record the audio output of the sound module.
How would you do that? I don't know if MidiO is functional in the record mode - then you could just record the sound from the module while it is triggered by the midi out. If not, I'd record it in the background to an audio editor while playing back the GB file and later import that audio.
But as I said, that's theoretical reasoning, since I haven't used MidiO. MyMac should know more.

Simple question: using mainstage as sound module AND run multi tracks live

i want to play live, using my mac book pro, mainstage as a sound module while running several (audio) outputs/tracks at the same time. is this possible and how is it done? i don't want to change to ableton live! thx a bunch everyone!

If doing vocals and latency is a problem, you can also experiment with taking your live (dry) signal direct (if you interface allows this) into your mix, which will be zero latency. Live signal also goes through MainStage to apply effects such as reverb and delay, and you feed the wet only signal to a separate "vocal fx" mix channel.
Of course, you can't apply plug-in based eq and compression to the realtime signal (or your back to the latency problem), you would have to rely on hardware/mixer for that. I have experimented with this method with good results...
Others have reported not being bothered by the amount of vocal latency, but I wonder...

Not recording from USB keyboard using sound module

I have a USB keyboard and Poizone sound module but nothing gets recorded into AA3 when I record to a track. Track is armed to record, soundcard is selected (analog in) but nothing. What am I overlooking? Thanks.

To record the output of another application in Audition, you'll need to configure your default Windows recording port to be "What You Hear" or something similar, if that is an available option. Right click the speaker icon in your Windows system tray, and choose to adjust the Recording Properties. Here, depending on your OS, you are given a list of different input ports on your audio device: Microphone, Line Input, CD, and often, What You Hear. This just routes the output port back as an input so other programs can record this stream.
Not all audio devices offer this, so it may not show up. In that case, you'd probably need to route your output to a mixer, send a feed back to the Line Input port on your audio device, and listen via headphones through the mixer. Note that you'll need to disable software-based monitoring to prevent feedback and that you'll definitely want to turn off system sounds in case someone sends you an e-mail or an IM while you're recording.

Sound modules missing/broken [solved]

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