Advanced Linux Sound Architecture

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The Advanced Linux Sound Architecture (ALSA) is a Linux kernel component intended to replace the original Open Sound System (OSSv3) for providing device drivers for sound cards. Besides the sound device drivers, ALSA also bundles a user space library for application developers who want to use driver features with a higher level API than direct interaction with the kernel drivers.

Note: For an alternative sound environment, see the Open Sound System page.

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ALSA is included in the default Arch kernel as a set of modules, so installing it is not necessary.

udev will automatically probe your hardware at boot, loading the corresponding kernel module for your audio card. Therefore, your sound should already be working, but upstream sources mute all channels by default.

Users must be in the audio group to play sound with ALSA. To add a user to the audio group, use the following command:

# gpasswd -a yourusername audio
Note: This will not take effect until the user logs back in.

User-space utilities

Install the alsa-utils package which contains the alsamixer user-space tool, which allows for configuration of the sound device from the console or terminal. alsa-utils and the other packages mentioned in this section are available in the official repositories.

Install the alsa-oss package if you want OSS applications to work with dmix (software mixing).

Note: Since udev >= 171, the OSS emulation modules (snd_seq_oss, snd_pcm_oss, snd_mixer_oss) are not automatically loaded by default.

Install the alsa-plugins package if you want high quality resampling, upmixing/downmixing and other advanced features.

Unmuting the channels

The current version of ALSA installs with all channels muted by default. You will need to unmute the channels manually.

It is easiest to use alsamixer ncurses UI to accomplish this (alternatively, use amixer from the commandline):

$ alsamixer

The label MM below a channel indicates that the channel is muted, and 00 indicates that it is open.

Scroll to the Master and PCM channels with the Template:Keypress and Template:Keypress keys and unmute them by pressing the Template:Keypress key. Use the Template:Keypress key to increase the volume and obtain a value of 0 dB gain. The gain can be found in the upper left next to the Item: field. Higher values of gain will produce distorted sound.

To get full 5.1 or 7.1 surround sound you likely need to unmute other channels such as Front, Surround, Center, LFE (subwoofer) and Side (these are the names of the channels with Intel HD Audio, they may vary with different hardware). Please take note that this will not automatically upmix stereo sources (like most music). In order to accomplish that, see #Upmixing/Downmixing.

Leave alsamixer by pressing Template:Keypress.

  • Some cards need to have digital output muted/turned off in order to hear analog sound. For the Soundblaster Audigy LS mute the IEC958 channel.

Tango-inaccurate.pngThe factual accuracy of this article or section is disputed.Tango-inaccurate.png

Reason: please use the first argument of the template to provide a brief explanation. (Discuss in Talk:Advanced Linux Sound Architecture#)
  • Some machines, (like the Thinkpad T61), have a Speaker channel which must be unmuted and adjusted as well.
  • Some machines, (like Dell E6400) may also require the Front and Headphone channels to be unmuted and adjusted.

Next, test to see if sound works:

$ speaker-test -c 2

Change -c to fit your speaker setup. Use -c 8 for 7.1, for instance:

$ speaker-test -c 8

If it does not work, proceed to #Configuration and then #Troubleshooting to solve your issues.

If it works, you just need to save your mixer settings.

# alsactl store

This will create the file /var/lib/alsa/asound.state, saving the alsamixer settings.

Then, add the alsa daemon to your DAEMONS array in /etc/rc.conf to automatically restore the mixer settings at boot. See rc.conf#Daemons for detailed instructions.

Note: The alsa daemon merely restores your volume mixer levels on boot up by reading /var/lib/alsa/asound.state. It is separate from the alsa audio library (and kernel level API).


Tango-edit-clear.pngThis article or section needs language, wiki syntax or style improvements. See Help:Style for reference.Tango-edit-clear.png

Reason: please use the first argument of the template to provide a brief explanation. (Discuss in Talk:Advanced Linux Sound Architecture#)

If you cannot hear anything, double check your mixer settings, being sure to unmute PCM, MASTER (and some machines such as the IBM Thinkpad have an additional 'SPEAKER' channel) and try the alsaconf utility as root:

# alsaconf

All necessary modules should be detected and loaded automatically by udev. No special configuration is required unless an ISA card is being used.

Warning: Do not use alsaconf if you have a PCI or ISAPNP sound card, as the entries alsaconf adds to the modprobe.conf file might break udev's autodetection.

Set the default sound card

In Kernel Space

Telephony-capable modems and modern graphics cards with HDMI output can conflict with the sound card for the default sound card slot. (pcspkr is another, non-ALSA PC speaker module. It will not conflict with ALSA sound cards.) To prevent this, discover your sound card model name with lspci(8) and your ALSA driver module names with ls(1):

$ ls -l /sys/module/snd/holders
total 0
lrwxrwxrwx 1 root root 0 Apr 25 00:56 snd_hda_codec -> ../../snd_hda_codec
lrwxrwxrwx 1 root root 0 Apr 25 00:56 snd_hda_codec_realtek -> ../../snd_hda_codec_realtek
lrwxrwxrwx 1 root root 0 Apr 25 00:56 snd_hda_intel -> ../../snd_hda_intel
lrwxrwxrwx 1 root root 0 Apr 25 00:56 snd_hwdep -> ../../snd_hwdep
lrwxrwxrwx 1 root root 0 Apr 25 00:56 snd_pcm -> ../../snd_pcm
lrwxrwxrwx 1 root root 0 Apr 25 00:56 snd_timer -> ../../snd_timer

Or better (source[1]):

$ cat /proc/asound/modules

Then add the names of your sound card modules to (source[2]):

options snd slots=snd-intel8x0

These entries ensure that the Intel 82801DB-ICH4 sound card will become card 0 and the PC speaker will become card 1.

Note: You will need to unload all your sound modules and reload them for the changes to take effect. It might be easier to reboot. Your choice.
Note: This method will NOT work if the conflicting devices use the same module. If you would prefer to not configure sound on a per user basis as in the section below, you can directly modify /usr/share/alsa/alsa.conf. For example, replace all "card 0" with "card 1" .

In User Space

This method does not require root permissions, is on a per-user basis, and takes effect as soon as any software has been restarted (like your media player, for example).

Located in /usr/share/alsa/alsa.conf is a list of defaults that alsa uses out of the box. These can be overridden in a ~/.asoundrc file for each user, or using the same syntax, ~/etc/asound.conf for every user. The following disables dmix, so multiple applications cannot use the same pcm device.

pcm.!default {
type hw
card 1
device 0
ctl.!default {
type hw
card 1
device 0 #this may cause alsamixer to crash with "cannot open mixer: Invalid argument"

The following leaves dmix intact

#overrides the defaults set in /usr/share/alsa/alsa.conf
defaults.ctl.card 1
defaults.pcm.card 1
defaults.pcm.device 0

The 'pcm' options affect which card and device will be used for audio playback. However it is the 'ctl' option affects which card is used by control utilities like alsamixer, amixer, and the like.

To find out which numbers correspond to what audio device, use aplay -l.

$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: CONEXANT Analog [CONEXANT Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 1: Conexant Digital [Conexant Digital]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: JamLab [JamLab], device 0: USB Audio [USB Audio]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 2: Audio [Altec Lansing XT1 - USB Audio], device 0: USB Audio [USB Audio]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

Making sure the sound modules are loaded

You can assume that udev will autodetect your sound properly. You can check this with the command

$ lsmod|grep '^snd' | column -t
snd_hda_codec_hdmi     22378   4
snd_hda_codec_realtek  294191  1
snd_hda_intel          21738   1
snd_hda_codec          73739   3  snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel
snd_hwdep              6134    1  snd_hda_codec
snd_pcm                71032   3  snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec
snd_timer              18992   1  snd_pcm
snd                    55132   9  snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer
snd_page_alloc         7017    2  snd_hda_intel,snd_pcm

If the output looks similar, your sound drivers have been successfully autodetected.

Note: Since udev>=171, the OSS emulation modules (snd_seq_oss, snd_pcm_oss, snd_mixer_oss) are not loaded by default: load them with modprobe and/or add them in the MODULES array in /etc/rc.conf if they are needed.

You might also want to check the directory /dev/snd for the right device files:

$ ls -l /dev/snd
total 0
crw-rw----  1 root audio 116,  0 Apr  8 14:17 controlC0
crw-rw----  1 root audio 116, 32 Apr  8 14:17 controlC1
crw-rw----  1 root audio 116, 24 Apr  8 14:17 pcmC0D0c
crw-rw----  1 root audio 116, 16 Apr  8 14:17 pcmC0D0p
crw-rw----  1 root audio 116, 25 Apr  8 14:17 pcmC0D1c
crw-rw----  1 root audio 116, 56 Apr  8 14:17 pcmC1D0c
crw-rw----  1 root audio 116, 48 Apr  8 14:17 pcmC1D0p
crw-rw----  1 root audio 116,  1 Apr  8 14:17 seq
crw-rw----  1 root audio 116, 33 Apr  8 14:17 timer
Note: If requesting help on IRC or the forums, please post the output of the above commands.

If you have at least the devices controlC0 and pcmC0D0p or similar, then your sound modules have been detected and loaded properly.

If this is not the case, your sound modules have not been detected properly. To solve this, you can try loading the modules manually:

  • Locate the module for your soundcard: ALSA Soundcard Matrix The module will be prefixed with 'snd-' (for example: 'snd-via82xx').
  • Load modules:
 # modprobe snd-NAME-OF-MODULE
  • Check for the device files in /dev/snd (see above) and/or try if alsamixer or amixer have reasonable output.
  • Add snd-NAME-OF-MODULE and snd-pcm-oss to the list of MODULES in /etc/rc.conf to ensure they are loaded next time (make sure snd-NAME-OF-MODULE is before snd-pcm-oss).

Restore ALSA Mixer settings at startup

  • Run alsactl -f /var/lib/alsa/asound.state store once to create /var/lib/alsa/asound.state.
# alsactl -f /var/lib/alsa/asound.state store
  • Edit /etc/rc.conf and add "alsa" to the list of daemons to start on boot-up. This will store the mixer settings on every shutdown and restore them when you boot.
  • If the mixer settings are not loaded on boot-up, add the following line to /etc/rc.local:
# alsactl -f /var/lib/alsa/asound.state restore
  • These methods still may not work, or you may prefer to have audio settings for individual users. In this case, run alsactl store -f ~/.asoundrc as a normal user. This will save and restore volume settings on a per user basis. To automate this process, add the respective commands to ~/.bash_login and ~/.bash_logout, or the correct locations for the shell of your choice.

For zsh, use ~/.zlogin and ~/.zlogout.

Getting SPDIF Output

(from gralves from the Gentoo forums)

  • In GNOME Volume Control, under the Options tab, change the IEC958 to PCM. This option can be enabled in the preferences.
  • If you do not have GNOME Volume Control installed,
    • Edit /var/lib/alsa/asound.state. This file is where alsasound stores your mixer settings.
    • Find a line that says: 'IEC958 Playback Switch'. Near it you will find a line saying value:false. Change it to value:true.
    • Now find this line: 'IEC958 Playback AC97-SPSA'. Change its value to 0.
    • Restart ALSA.

Alternative way to enable SPDIF output automatically on login (tested on SoundBlaster Audigy):

  • add following lines to /etc/rc.local:
 # Use COAX-digital output
 amixer set 'IEC958 Optical' 100 unmute
 amixer set 'Audigy Analog/Digital Output Jack' on

You can see the name of your card's digital output with:

 $ amixer scontrols

System-Wide Equalizer

Using AlsaEqual (provides UI)

Install alsaequal from the AUR.

Note: If you have a x86_64-system and are using a 32bit-flashplugin the sound in flash will not work. Either you have to disable alsaequal or build alsaequal for 32bit.

After installing the package, insert the following into your ALSA configuration file (~/.asoundrc or /etc/asound.conf):

ctl.equal {
 type equal;

pcm.plugequal {
  type equal;
  # Modify the line below if you do not
  # want to use sound card 0.
  #slave.pcm "plughw:0,0";
  #by default we want to play from more sources at time:
  slave.pcm "plug:dmix";
#pcm.equal {
  # Or if you want the equalizer to be your
  # default soundcard comment the following
  # line and uncomment the above line.
pcm.!default {
  type plug;
  slave.pcm plugequal;

Then Reload your alsa settings (as root).

# /etc/rc.d/alsa restart

not sure if this is really needed - more important is to shut down all applications using ALSA...

And you are ready to change your equalizer using command

$ alsamixer -D equal

Note that configuration file is different for each user (until not specified else) it is saved in $HOME/.alsaequal.bin. so if you want to use AlsaEqual with mpd or another software running under different user, you can configure it using

# su mpd -c 'alsamixer -D equal'

or eg. you can make symlink to your .alsaequal.bin in his home...

Using mbeq

Note: This method requires the use of a ladspa plugin which might use quite a bit of CPU when sound plays. In addition, this was made with stereophonic sound (e.g. headphones) in mind.
  • You will need, in addition to the aforementioned userspace utilities, alsa-plugins.
# pacman -S alsa-plugins
  • Get the ladspa and swh-plugins packages too if you do not already have them.
# pacman -S ladspa swh-plugins
  • If you have not already created either an ~/.asoundrc or a /etc/asound.conf file, then create either one
$ vim ~/.asoundrc
  • Insert the following into your ALSA configuration file (~/.asoundrc or /etc/asound.conf):
pcm.eq {
  type ladspa

  # The output from the EQ can either go direct to a hardware device
  # (if you have a hardware mixer, e.g. SBLive/Audigy) or it can go
  # to the software mixer shown here.
  #slave.pcm "plughw:0,0"
  slave.pcm "plug:dmix"

  # Sometimes you may need to specify the path to the plugins,
  # especially if you have just installed them.  Once you have logged
  # out/restarted this should not be necessary, but if you get errors
  # about being unable to find plugins, try uncommenting this.
  #path "/usr/lib/ladspa"

  plugins [
      label mbeq
      id 1197
      input {
        #this setting is here by example, edit to your own taste
        #bands: 50hz, 100hz, 156hz, 220hz, 311hz, 440hz, 622hz, 880hz, 1250hz, 1750hz, 25000hz,
        #50000hz, 10000hz, 20000hz
        controls [ -5 -5 -5 -5 -5 -10 -20 -15 -10 -10 -10 -10 -10 -3 -2 ]

 # Redirect the default device to go via the EQ - you may want to do
 # this last, once you're sure everything is working.  Otherwise all
 # your audio programs will break/crash if something has gone wrong.

 pcm.!default {
  type plug
  slave.pcm "eq"

 # Redirect the OSS emulation through the EQ too (when programs are running through "aoss")

 pcm.dsp0 {
  type plug
  slave.pcm "eq"
  • Reload your alsa settings (as root).
# /etc/rc.d/alsa restart
  • You should be good to go (if not, ask in the forum).

High quality resampling

When software mixing is enabled, ALSA is forced to resample everything to the same frequency (48000 by default when supported). dmix uses a poor resampling algorithm which produces noticeable sound quality loss.

Install alsa-plugins and libsamplerate:

# pacman -S alsa-plugins libsamplerate

Change the default rate converter to libsamplerate:

defaults.pcm.rate_converter "samplerate_best"


defaults.pcm.rate_converter "samplerate_best"

samplerate_best offers the best sound quality, but you need a decent CPU to be able to use it as it requires a lot of CPU cycles for real-time resampling. There are other algorithms available (samplerate, etc.) but they may not provide much of an improvement over the default resampler.

Warning: On some systems, enabling samplerate_best may cause an issue where you get no sound from flashplayer.



In order for stereo sources like music to be able to saturate a 5.1 or 7.1 sound system, you need to use upmixing. In darker days this used to be tricky and error prone but nowadays plugins exist to easily take care of this task. Thus, install alsa-plugins:

# pacman -S alsa-plugins

Then add the following to your alsa configuration file of choice (either /etc/asound.conf or ~/.asoundrc):

pcm.upmix71 {
    type upmix
    slave.pcm "surround71"
    delay 15
    channels 8

You can easily change this example for 7.1 upmixing to 5.1 or 4.0.

This adds a new pcm that you can use for upmixing. If you want all sound sources to go through this pcm, add it as a default below the previous definition like so:

pcm.!default "plug:upmix71"

The plugin automatically allows multiple sources to play through it without problems so setting is as a default is actually a safe choice.

If you experience skipping or distorted sound, consider increasing the buffer_size (to 32768, for example) or use a high quality resampler.


If you want to downmix sources to stereo because you, for instance, want to watch a movie with 5.1 sound on a stereo system, you need to use the vdownmix plugin that alsa provides in the same package.

Again, in your configuration file, add this:

pcm.!surround51 {
    type vdownmix
    slave.pcm "default"
pcm.!surround40 {
    type vdownmix
    slave.pcm "default"


Software mixing (dmix)

Note: For ALSA 1.0.9rc2 and higher you do not need to setup dmix. Dmix is enabled as default for soundcards which do not support hardware mixing.

If that does not work however it is a matter of simply creating a .asoundrc file in your home folder with the following contents.

pcm.dsp {
    type plug
    slave.pcm "dmix"

This should enable software mixing and allows more than one application to make use of the soundcard.

See #Troubleshooting for common problems and solutions.

Hardware mixing

Tango-edit-clear.pngThis article or section needs language, wiki syntax or style improvements. See Help:Style for reference.Tango-edit-clear.png

Reason: please use the first argument of the template to provide a brief explanation. (Discuss in Talk:Advanced Linux Sound Architecture#)
  • Use a soundcard that supports hardware mixing: old SoundBlaster Live! series (emu10k1 chip) and new Audigy series (emu10k2 chip) cards are known to work.
  • The SoundBlaster Live! 5.1 is fairly easy to find and is cheap to buy.
Note: new cheap Audigy SE, Audigy 2 NX, SoundBlaster Live! 24bit and SoundBlaster Live! 7.1 do not support hardware mixing (they use other chips).
  • 99% of motherboards with integrated sound do not support hardware mixing.
Note: An exception is the onboard VIA8237 chip which supports 4-stream hardware mixing. However it does only 3 for some motherboards (the 4th makes no sound), or is just broken. Even if it works, the quality is not good compared to other solutions.
  • For Arch64 and the 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 02), and to get sound from Enemy Territory, it works this by setting these:
echo "et.x86 0 0 direct" > /proc/asound/card0/pcm0p/oss
echo "et.x86 0 0 disable" > /proc/asound/card0/pcm0c/oss


Sound Skipping While Using Dynamic Frequency Scaling

Some combinations of ALSA drivers and chipsets may cause audio from all sources to skip when used in combination with a dynamic frequency scaling governor such as ondemand or conservative. Currently, the solution is to switch back to the performance governor.

Refer to the CPU Frequency Scaling for more information.

Problems with Availability to Only One User at a Time

You might find that only one user can use the dmixer at a time. This is probably ok for most, but for those who run mpd as a separate user this poses a problem. When mpd is playing a normal user cannot play sounds though the dmixer. While its quite possible to just run mpd under a user's login account, another solution has been found. Adding the line ipc_key_add_uid 0 to the pcm.dmixer block disables this locking. The following is a snippet of the asound.conf, the rest is the same as above.

pcm.dmixer {
 type dmix
 ipc_key 1024
 ipc_key_add_uid 0
 ipc_perm 0660
slave {

Random Lack of Sound on Startup

If you randomly have no sound on startup, it may be because your system has multiple sound cards, and their order may sometimes change on startup. If this is the case, then change this section of /etc/asound.conf:

ctl.dmixer {
    type hw
    card FOO

Replace FOO with the desired audio device, as reported in the /proc/asound/cards file. An example of the file is shown below.

 0 [U0x46d0x9a1    ]: USB-Audio - USB Device 0x46d:0x9a1
                      USB Device 0x46d:0x9a1 at usb-0000:00:12.2-2, high speed
 1 [SB             ]: HDA-Intel - HDA ATI SB
                      HDA ATI SB at 0xf9ff4000 irq 16

Device 0 is the microphone built into a webcam, while device 1 is the integrated sound card. If you have copied the /etc/asound.conf from above as is, alsa will attempt to initialize the microphone as an audio output device, but will fail and you will have no sound. Rather than setting FOO to the number, you set it to the name next to the number, like so:

ctl.dmixer {
    type hw
    card SB

If you use dmix, you will want to have a configuration such as this one:

ctl.!default {
    type hw
    card Intel #adapt this to your card
    #device 0 #not required, defaults to 0 (see the output of aplay -L)

pcm.dmixer {
    type dmix
    ipc_key 1024 #required and must be unique
    slave.pcm {
        type hw
        card Intel #same as above
        #device 0 #same as above

pcm.!default {
    type plug
    slave.pcm dmixer

To make most programs use Alsa, make sure /etc/libao.conf only contains the following.


If you use mpd and the configuration tips above do not work for you, try reading this instead.

Specific Program Problems

For other programs who insist on their own audio setup, eg, XMMS or Mplayer, you would need to set their specific options.

For mplayer, open up ~/.mplayer/config (or /etc/mplayer/mplayer.conf for global setting) and add the following line:


For XMMS/Beep Media Player, go into their options and make sure the sound driver is set to Alsa, not oss.

To do this in XMMS:

  • Open XMMS
    • Options -> preferences.
    • Choose the Alsa output plugin.

For applications which do not provide a ALSA output, you can use aoss from the alsa-oss package. To use aoss, when you run the program, prefix it with aoss, eg:

aoss realplay

pcm.!default{ ... } doesnt work for me anymore. but this does:

 pcm.default pcm.dmixer

Model Settings

Although Alsa detects your soundcard through the BIOS at times Alsa may not be able to recognize your model type. The soundcard chip can be found in alsamixer (e.g. ALC662) and the model can be set in /etc/modprobe.d/modprobe.conf or /etc/modprobe.d/sound.conf. For example:

options snd-hda-intel model=MODEL

There are other model settings too. For most cases Alsa defaults will do. If you want to look at more specific settings for your soundcard take a look at the Alsa Soundcard List find your model, then Details, then look at the "Setting up modprobe..." section. Enter these values in /etc/modprobe.d/modprobe.conf. For example, for an Intel AC97 audio:

# ALSA portion
alias char-major-116 snd
alias snd-card-0 snd-intel8x0
# module options should go here

# OSS/Free portion
alias char-major-14 soundcore
alias sound-slot-0 snd-card-0

# card #1
alias sound-service-0-0 snd-mixer-oss
alias sound-service-0-1 snd-seq-oss
alias sound-service-0-3 snd-pcm-oss
alias sound-service-0-8 snd-seq-oss
alias sound-service-0-12 snd-pcm-oss

Conflicting PC Speaker

If you're sure nothing is muted, that your drivers are installed correctly, and that your volume is right, but you still do not hear anything, then try adding the following line to /etc/modprobe.d/modprobe.conf:

options snd-NAME-OF-MODULE ac97_quirk=0

The above fix has been observed to work with via82xx

options snd-NAME-OF-MODULE ac97_quirk=1

The above fix has been reported to work with snd_intel8x0

No Microphone Input

In alsamixer, make sure that all the volume levels are up under recording, and that CAPTURE is toggled active on the microphone (e.g. Mic, Internal Mic) and/or on Capture (in alsamixer, select these items and press space). Try making positive Mic Boost and raising Capture and Digital levels higher; this make make static or distortion, but then you can adjust them back down once you're hearing something when you record

As the pulseaudio wrapper is shown as "default" in alsamixer, your may have to press F6 to select your actual soundcard first. You may also need to enable and increase the volume of Line-in in the Playback section.

To test the microphone, run these commands (see arecord's man page for further information):

 arecord -d 5 test-mic.wav
 aplay test-mic.wav

If all fails, you may want to eliminate hardware failure by testing the microphone with a different device.

For at least some computers, muting a microphone (MM) simply means its input doesn't go immediately to the speakers. It still receives input.

Many Dell laptops need "-dmic" to be appended to the model name in /etc/modprobe.d/modprobe.conf:

 options snd-hda-intel model=dell-m6-dmic

Some programs use try to use OSS as the main input software. Add the following lines to /etc/modprobe.d/modprobe.conf to prevent OSS modules from being loaded:

Note: The OSS modules are no longer autoloaded anyway.
blacklist snd_pcm_oss
blacklist snd_mixer_oss
blacklist snd_seq_oss

See also:

Setting the default Microphone/Capture Device

Some applications (Pidgin, Adobe Flash) do not provide an option to change the capture device. It becomes an issue if your microphone is on a separate device (i.e. USB webcam or microphone) than your internal sound card. To change only the default capture device, leaving the default playback device as is, you can modify your ~/.asoundrc file to include the following:

    type hw
    card U0x46d0x81d

    type asym
        type plug
        slave.pcm "dmix"
        type plug
        slave.pcm "usb"

Replace "U0x46d0x81d" with your capture device's card name in ALSA. You can use 'arecord -L' to list all the capture devices detected by ALSA.

Internal Microphone not working

First make sure all the volume levels are up under recording in alsamixer. In my case adding the following option to /etc/sound.conf and reloading the snd-* module produced a new volume setting called Capture which was capturing for the internal mic. For eg, for snd-hda-intel add

 options snd-hda-intel enable_msi=1

Then reload the module (as below), up the recording volume of Capture and then test.

# rmmod snd-hda-intel && modprobe snd-hda-intel

No Sound with Onboard Intel Sound Card

There may be an issue with two conflicting modules loaded, namely snd_intel8x0 and snd_intel8x0m. In this case, blacklist snd_intel8x0m:

blacklist snd_intel8x0m

Muting the "External Amplifier" in alsamixer or amixer may also help. See the ALSA wiki.

No Headphone Sound with Onboard Intel Sound Card

With Intel Corporation 82801 I (ICH9 Family) HD Audio Controller on laptop, you may need to add this line to modprobe or sound.conf:

options snd-hda-intel model=$model

Where $model is any one of the following (in order of possibility to work, but not merit):

  • dell-vostro
  • olpc-xo-1_5
  • laptop
  • dell-m6

Note: It may be necessary to put this "options" line below (after) any "alias" lines about your card.

You can see all the available models in the kernel documentation. For example here, but check that it is the correct version of that document for your kernel version.

Note that there is a high chance none of the input devices (all internal and external mics) will work if you choose to do this, so it is either your headphones or your mic. Please report to ALSA if you are affected by this bug.

And also, if you have problems getting beeps to work (pcspkr):

options snd-hda-intel model=$model enable=1 index=0

No sound when S/PDIF video card is installed

Discover available modules and their order:

$ cat /proc/asound/modules
0 snd_hda_intel
1 snd_ca0106

Disable the undesired video card audio codec in /etc/modprobe.d/modprobe.conf:

# /etc/modprobe.d/modprobe.conf
install snd_hda_intel /bin/false

If both devices use the same module, it might be possible to disable one of them in the BIOS.

Poor sound quality or clipping

If you experience poor sound quality, try setting the PCM volume (in alsamixer) to a level such that gain is 0.

Pops When Starting and Stopping Playback

Some modules can power off your sound card when not in use. this can make an audible noise when powering down your sound card. If you find this annoying try "modinfo snd-MY-MODULE", and look for a module option that adjusts or disables this feature.

for example: to disable the power saving mode using snd-hda-intel add "options snd-hda-intel power_save=0" in /etc/modprobe.d/modprobe.conf. or try it with "modprobe snd-hda-intel power_save=0"

Alsamixer Does Not Run

If running alsamixer does not work and you wind up with the following error

alsamixer: function snd_ctl_open failed for default: No such device or directory

You should first check /etc/group to ensure that your current user is in the 'audio' group. Do not forget to log out and log in again for the group changes.

Then you might need to re-install your kernel. Run 'pacman -S linux' or whichever patchset you prefer to use.

S/PDIF Output Does Not Work

If the optical/coaxial digital output of your motherboard/sound card is not working or stopped working, and have already enabled and unmuted it in alsamixer, try running

iecset audio on

as root.

You can also put this command in rc.local as it sometimes it may stop working after a reboot.

HDMI Output Does Not Work

If the HDMI output of your motherboard/sound card is not working or stopped working, and have already enabled and unmuted it in alsamixer, try the following.

Query for Playback Devices:

 % aplay -l
 **** List of PLAYBACK Hardware Devices ****
 card 0: NVidia [HDA NVidia], device 0: ALC1200 Analog [ALC1200 Analog]
   Subdevices: 1/1
   Subdevice #0: subdevice #0
 card 0: NVidia [HDA NVidia], device 1: ALC1200 Digital [ALC1200 Digital]
   Subdevices: 1/1
   Subdevice #0: subdevice #0
 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI]
   Subdevices: 0/1
   Subdevice #0: subdevice #0

Now that we have the info for the HDMI device, try a test, In the example below, 0 is the card number and 3 is the device number.

 % aplay -D plughw:0,3 /usr/share/sounds/alsa/Front_Center.wav

If aplay does not output any errors, but still no sound is heared, "reboot" the receiver, monitor or tv set. Since the HDMI interface executes a handshake on connection, it might have noticed before that there was no audio stream embedded, and disabled audio decoding.

Note: If you're using an ATI card and linux 3.0, check your kernel/grub settings, you most likely need to use the '' boot parameter, or alternatively set up the option in your modprobe configuration files.

If the test is successful, create or edit your ~/.asoundrc file to set HDMI as the default audio device.

 pcm.!default {
      type hw
      card 0
      device 3

You can restart alsa with the example below or you may reboot, and audio should now work.

Note: Any programs running that use alsa will be closed.
# /etc/rc.d/alsa force-restart

Using udev to automatically turn HDMI audio on or off

The previous steps in this section are good for testing, or if you don't need to hotplug your HDMI cable often. We now consider the situation where you have a laptop and want HDMI audio only when you have the HDMI cable plugged in.

The code in this section comes from The Arch Linux Forums.

We create a udev rule (for processing the cable connect/disconnect events), two different ALSA configuration files, and a script that switches between these configurations.

First create /etc/alsa/ if it doesn't exist yet; it will house the configuration files and the switcher script:

[[ -d /etc/alsa/ ]] || sudo mkdir -v /etc/alsa

Place inside this directory the following files (which you may wish to tweak as you see fit):

#! /bin/bash
set -x

udevadm settle --quiet --timeout=16

read hdmi_status </sys/class/drm/card0-HDMI-A-1/status

if [[ -n $hdmi_status && -f "$alsa_dir/$asound_file" ]]; then
    ln -sf "$alsa_dir/$asound_file" /etc/asound.conf
    /etc/rc.d/alsa force-restart
pcm.!default {
      type hw
      card 0
      device 3
ctl.!default {
    type hw
    card 0
    device 3
pcm.!default {
      type hw
      card 0
      device 0
ctl.!default {
    type hw
    card 0
    device 0

You may now test that the switcher script works by running,

 sudo chmod +x /etc/alsa/hdmi-switch
 sudo /etc/alsa/hdmi-switch

You should find that you can successfully obtain HDMI audio output when you run the script while the cable is plugged in. You should be able to restore your default audio output by unplugging the cable and manually running the script again.

Once you have that working, we automate the process by adding the udev rule:

SUBSYSTEM=="drm", ACTION=="change", RUN+="/bin/bash /etc/alsa/hdmi-switch"

To make udev incorporate the new rule immediately, reload all rules:

  sudo udevadm control --reload-rules

HDMI Multi-channel PCM output does not work (Intel)

As of Linux 3.1 multi-channel PCM output through HDMI with a Intel card (Intel Eaglelake, IbexPeak/Ironlake,SandyBridge/CougarPoint and IvyBridge/PantherPoint) is not yet supported. Support for it has been recently added and expected to be available in Linux 3.2. To make it work in Linux 3.1 you need to apply the following patches:

No Adjustable PCM Channel

You may find that you lack adjustable PCM channel. In this case try to remove all sound-related stuff from MODULES section in /etc/rc.conf, except for snd-NAME-OF-MODULE.

HP TX2500

Add these 2 lines into /etc/modprobe.d/modprobe.conf:

options snd-cmipci mpu_port=0x330 fm_port=0x388
options snd-hda-intel index=0 model=toshiba position_fix=1
options snd-hda-intel model=hp (works for tx2000cto)

Skipping Sound When Playing MP3

If you have sound skipping when playing MP3 files and you have more then 2 speakers attached to your computer (i.e. > 2 speaker system), run alsamixer and disable the channels for the speakers that you DO NOT have (i.e. do not enable the sound for the center speaker if you do not have a center speaker.

Using a USB Headset and External USB Sound Cards

If you are using a USB headset with ALSA you can try using asoundconf (currently only available from the AUR) to set the headset as the primary sound output. note: before running please make sure you have usb audio module enabled

#modprobe snd-usb-audio

you can add this to /etc/rc.conf if you wish

# asoundconf is-active
# asoundconf list
# asoundconf set-default-card <chosen soundcard>

Crackling sound with USB devices

If you experience crackling sound on USB devices you can try tunning the snd-usb-audio for minimal latency.

Add this to your /etc/modprobe.d/modprobe.conf :

options snd-usb-audio nrpacks=1

source :

Error 'Unkown hardware' Appears After a Kernel Update

The following messages may be displayed during the start-up ALSA after the kernel update:

Unknown hardware "foo" "bar" ...
Hardware is initialized using a guess method
/usr/sbin/alsactl: set_control:nnnn:failed to obtain info for control #mm (No such file or directory)

Simply store ALSA mixer settings again (as root):

# alsactl -f /var/lib/alsa/asound.state store

HDA Analyzer

If the mappings to your audio pins(plugs) do not correspond but ALSA works fine, you could try HDA Analyzer -- a pyGTK2 GUI for HD-audio control can be found at the ALSA wiki. Try tweaking the Widget Control section of the PIN nodes, to make microphones IN and headphone jacks OUT. Referring to the Config Defaults heading is a good idea.

NOTE: the script is done by such way that it is incompatible with python3 (which is now shipped with ArchLinux) but tries to use it. The workaround is: open "", find all occurences of "python" (2 occurences - one on the first line, and the second on the last line) and replace them all by "python2".

NOTE2: the script requires root acces, but running it via su/sudo is bogus. Run it via kdesu or gtksu.


If you get no sound via SDL and ALSA cannot be chosen from the application. Try setting the environmental variable SDL_AUDIODRIVER to alsa.

# export SDL_AUDIODRIVER=alsa

Example configurations

See Advanced Linux Sound Architecture/Example Configurations.

External Resources