Advanced Linux Sound Architecture
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The Advanced Linux Sound Architecture (ALSA) is a Linux kernel component intended to provide device drivers for sound cards.
This document tells how to get ALSA working with 2.6 kernels. Also see how to allow multiple programs to play sound at once
See the Open Sound System page if you are looking for alternatives.
Installation
Kernel drivers
ALSA has been included in the 2.6 kernels and is included in all Arch kernel26* packages. If you build a custom kernel, do not forget to enable the correct ALSA driver.
All necessary modules should be detected and loaded automatically by udev. No special configuration has to be done unless you use ISA cards. NEVER 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.
Userspace utilities
- Required for native ALSA programs and administration
# pacman -Sy alsa-lib alsa-utils
- Recommended if you want to use applications with OSS sound support in combination with dmix:
# pacman -S alsa-oss
All ALSA programs will most likely have alsa-lib as a dependency.
Configuration
Set the default sound card
Telephony-capable modems and snd-pcsp (the internal PC speaker ALSA module) 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 2009-06-02 23:49 snd_ac97_codec -> ../../snd_ac97_codec lrwxrwxrwx 1 root root 0 2009-06-02 23:49 snd_intel8x0 -> ../../snd_intel8x0 lrwxrwxrwx 1 root root 0 2009-06-02 23:49 snd_intel8x0m -> ../../snd_intel8x0m lrwxrwxrwx 1 root root 0 2009-06-02 23:49 snd_pcm -> ../../snd_pcm lrwxrwxrwx 1 root root 0 2009-06-02 23:53 snd_pcsp -> ../../snd_pcsp lrwxrwxrwx 1 root root 0 2009-06-02 23:49 snd_timer -> ../../snd_timer
Then add the names of your sound card modules to:
options snd-intel8x0 index=0 options snd-pcsp index=1
These entries ensure that the Intel 82801DB-ICH4 sound card will become card 0 and the PC speaker will become card 1. NB: Some modules do not support indexing options eg. snd-hda-intel. If you experience problems on on reboot edit /etc/modprobe.d/modprobe.conf to just have the modules.
If you do not want snd-pcsp to load at all you can add the following to:
MODULES=(... !snd-pcsp)
If you want the PC speaker completely been disabled you can additionally add the following to:
MODULES=(... !pcspkr)
Making sure the sound modules are loaded
You can assume that udev will autodetect your sound properly, including the OSS compatibility modules. You can check this with the command
$ lsmod|grep '^snd' | column -t snd_usb_audio 69696 0 snd_usb_lib 13504 1 snd_usb_audio snd_rawmidi 20064 1 snd_usb_lib snd_hwdep 7044 1 snd_usb_audio snd_seq_oss 29412 0 snd_seq_midi_event 6080 1 snd_seq_oss snd_seq 46220 4 snd_seq_oss,snd_seq_midi_event snd_seq_device 6796 3 snd_rawmidi,snd_seq_oss,snd_seq snd_pcm_oss 45216 0 snd_mixer_oss 15232 1 snd_pcm_oss snd_intel8x0 27932 0 snd_ac97_codec 87648 1 snd_intel8x0 snd_ac97_bus 1792 1 snd_ac97_codec snd_pcm 76296 4 snd_usb_audio,snd_pcm_oss,snd_intel8x0,snd_ac97_codec snd_timer 19780 2 snd_seq,snd_pcm snd 43776 12 snd_usb_audio,snd_rawmidi,snd_hwdep,snd_seq_oss,snd_seq,snd_seq_device, snd_pcm_oss,snd_mixer_oss,snd_intel8x0,snd_ac97_codec,snd_pcm,snd_timer snd_page_alloc 7944 2 snd_intel8x0,snd_pcm
If the output looks similar, your sound drivers have been successfully autodetected (note that in this case, snd_intel8x0 and snd_usb_audio are the drivers for the hardware devices). 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
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. If you want any help on IRC or the forums, please post the output of the above commands. 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 # modprobe snd-pcm-oss
- 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).
Unmute the channels and test
In this section, we assume that you are logged in as root. If you want to perform these steps as an unprivileged user, you have to skip to the next section Setup Permissions first.
- Unmute Soundcard
The current version of ALSA installs with all channels muted by default, so even if installation completes successfully and all devices are working properly you will hear no sound. You will need to unmute the channels manually. It is recommended to use alsamixer to accomplish this. From the alsamixer text ui, the label "MM" below a channel indicates that the channel is muted, and "00" indicates that it is open. Press the 'm' key to toggle MM/00. Use arrow-keys left and right to navigate through the channels and the arrow-keys up and down to adjust the volume. Such things as Master and PCM and possibly Speaker will need to unmuted for your sound to work.
# amixer set Master 90% unmute # amixer set PCM 85% unmute
- Try to play a WAV file
# aplay /usr/share/sounds/alsa/Front_Center.wav
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
Setup Permissions
To be able to use the sound card as a user, follow these steps:
- Add your user to the audio group:
# gpasswd -a USERNAME audio
- Log your user out and back in to ensure the audio group is loaded.
Restore ALSA Mixer settings at startup
- Run alsactl store once to create /etc/asound.state.
# alsactl 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 restore
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 don't have GNOME Volume Control installed,
- Edit /etc/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
- 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 don't already have them.
# pacman -S ladspa swh-plugins
- If you haven't 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've just installed them. Once you've logged # out/restarted this shouldn't 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).
Troubleshooting
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. 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
Remember, ALSA installs with all channels muted by default (see previous section, unmuting your soundcard).
However, 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 blacklisting snd-pcsp from your modules array in rc.conf:
MODULES=(!snd-pcsp ... )
Note that this will disable your PC's internal speaker.
If that doesn't work, 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 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, edit rc.conf
and in the MODULES array blacklist the latter one so that it reads !snd_intel8x0m
afterwards.
Muting the "External Amplifier" in alsamixer
or amixer
may also help. See the ALSA wiki.
With Intel Corporation 82801 I (ICH9 Family) HD Audio Controller on laptop, you may need to add this line to
/etc/modprobe.d/sound
options snd-hda-intel model=laptop
or
options snd-hda-intel model=laptop enable=1 index=0
Otherwise, the pcspkr may not work, and only the headphone have sound.
Poor Sound Quality
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. Don't forget to log out and log in again for the group changes.
Then you might need to re-install your kernel. Run 'pacman -S kernel26' 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 the test is successful, edit/create asound.conf in /etc to set HDMI as the default audio device, reboot, and audio should now work. There might be a better way to do this??
% cat /etc/asound.conf pcm.!default { type plug slave.pcm { type hw card 0 device 3 } }
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 and snd-pcm-oss.
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
And don't forget to enable 'hal' in the DAEMONS section of your /etc/rc.conf.
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 attacked to your computer (i.e. > 2 speaker system), run alsamixer and disable the channels for the speakers that you DON'T have (i.e. don't enable the sound for the center speaker if you don't 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 modual enabled
#modprobe snd-usb-audio
you can add this to /etc/rc.conf if you wish
# asoundconf is-active # asoundconf list # asoundconf set-default-soundcard <chosen soundcard>
KDE Settings
- Start up KDE:
# startx
- Set up the volumes as you want them for this user (each user has their own settings):
# alsamixer
Log out and log back in as user xyz to get sound to work (I had to kill X, logout then log back in as user xyz, then start X and open Firefox and bam audio working on YouTube)
- KDE 3.3: Go to K Menu → Multimedia → KMix
- Choose Settings → Configure KMix...
- Uncheck the option "Restore volumes on logon"
- Press OK, and you should be all set. Now your volumes will be the same from the command line or within KDE.
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)
Then, try to store ALSA mixer settings again:
# alsactl 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.
Example configurations
The following should serve as a guide for more advanced setups. The configuration takes place in /etc/asound.conf as mentioned earlier in this article. None of the following configurations are guaranteed to be working.
Upmixing of stereo sources to 7.1 using dmix while saturated sources do not get upmixed
# 2008-11-15 # # This .asoundrc will allow the following: # # - upmix stereo files to 7.1 speakers. # - playback real 7.1 sounds, on 7.1 speakers, # - allow the playback of both stereo (upmixed) and surround(7.1) sources at the same time. # - use the 6th and 7th channel (side speakers) as a separate soundcard, i.e. for headphones # (This is called the "alternate" output throughout the file, device names prefixed with 'a') # - play mono sources in stereo (like skype & ekiga) on the alterate output # # Make sure you have "8 Channels" and NOT "6 Channels" selected in alsamixer! # # Please try the following commands, to make sure everything is working as it should. # # To test stereo upmix : speaker-test -c2 -Ddefault -twav # To test surround(5.1): speaker-test -c6 -Dplug:dmix6 -twav # To test surround(7.1): speaker-test -c6 -Dplug:dmix8 -twav # To test alternative output: speaker-test -c2 -Daduplex -twav # To test mono upmix: speaker-test -c1 -Dmonoduplex -twav # # # It may not work out of the box for all cards. If it doesnt work for you, read the comments throughout the file. # The basis of this file was written by wishie of #alsa, and then modified with info from various sources by # squisher. Svenstaro modified it for 7.1 output support. #Define the soundcard to use pcm.snd_card { type hw card 0 device 0 } # 8 channel dmix - output whatever audio, to all 8 speakers pcm.dmix8 { type dmix ipc_key 1024 ipc_key_add_uid false ipc_perm 0660 slave { pcm "snd_card" rate 48000 channels 8 period_time 0 period_size 1024 buffer_time 0 buffer_size 5120 } # Some cards, like the "nforce" variants require the following to be uncommented. # It routes the audio to the correct speakers. # bindings { # 0 0 # 1 1 # 2 4 # 3 5 # 4 2 # 5 3 # 6 6 # 7 7 # } } # upmixing - duplicate stereo data to all 8 channels pcm.ch71dup { type route slave.pcm dmix8 slave.channels 8 ttable.0.0 1 ttable.1.1 1 ttable.0.2 1 ttable.1.3 1 ttable.0.4 0.5 ttable.1.4 0.5 ttable.0.5 0.5 ttable.1.5 0.5 ttable.0.6 1 ttable.1.7 1 } # this creates a six channel soundcard # and outputs to the eight channel one # i.e. for usage in mplayer I had to define in ~/.mplayer/config: # ao=alsa:device=dmix6 # channels=6 pcm.dmix6 { type route slave.pcm dmix8 slave.channels 8 ttable.0.0 1 ttable.1.1 1 ttable.2.2 1 ttable.3.3 1 ttable.4.4 1 ttable.5.5 1 ttable.6.6 1 ttable.7.7 1 } # share the microphone, i.e. because virtualbox grabs it by default pcm.microphone { type dsnoop ipc_key 1027 slave { pcm "snd_card" } } # rate conversion, needed i.e. for wine pcm.2chplug { type plug slave.pcm "ch71dup" } pcm.a2chplug { type plug slave.pcm "dmix8" } # routes the channel for the alternative # 2 channel output, which becomes the 7th and 8th channel # on the real soundcard #pcm.alt2ch { # type route # slave.pcm "a2chplug" # slave.channels 8 # ttable.0.6 1 # ttable.1.7 1 #} # skype and ekiga are only mono, so route left channel to the right channel # note: this gets routed to the alternative 2 channels pcm.mono_playback { type route slave.pcm "a2chplug" slave.channels 8 # Send Skype channel 0 to the L and R speakers at full volume #ttable.0.6 1 #ttable.0.7 1 } # 'full-duplex' device for use with aoss pcm.duplex { type asym playback.pcm "2chplug" capture.pcm "microphone" } #pcm.aduplex { # type asym # playback.pcm "alt2ch" # capture.pcm "microphone" #} pcm.monoduplex { type asym playback.pcm "mono_playback" capture.pcm "microphone" } # for aoss pcm.dsp0 "duplex" ctl.mixer0 "duplex" # softvol manages volume in alsa # i.e. wine likes this pcm.mainvol { type softvol slave.pcm "duplex" control { name "2ch-Upmix Master" card 0 } } #pcm.!default "mainvol" # set the default device according to the environment # variable ALSA_DEFAULT_PCM and default to mainvol pcm.!default { @func refer name { @func concat strings [ "pcm." { @func getenv vars [ ALSA_DEFAULT_PCM ] default "mainvol" } ] } } # uncomment the following if you want to be able to control # the mixer device through environment variables as well #ctl.!default { # @func refer # name { @func concat # strings [ "ctl." # { @func getenv # vars [ ALSA_DEFAULT_CTL # ALSA_DEFAULT_PCM # ] # default "duplex" # } # ] # } #}
Surround51 incl. upmix stereo & dmix, swap L/R, bad speaker position in room
Bad practice but works fine for almost everything without additional per-program/file customization:
pcm.!default { type route ## forwards to the mixer pcm defined below slave.pcm dmix51 slave.channels 6 ## "Native Channels" stereo, swap left/right ttable.0.1 1 ttable.1.0 1 ## original normal left/right commented out # ttable.0.0 1 # ttable.1.1 1 ## route "native surround" so it still works but weaken signal (+ RL/RF swap) ## because my rear speakers are more like random than really behind me ttable.2.3 0.7 ttable.3.2 0.7 ttable.4.4 0.7 ttable.5.5 0.7 ## stereo => quad speaker "upmix" for "rear" speakers + swap L/R ttable.0.3 1 ttable.1.2 1 ## stereo L+R => join to Center & Subwoofer 50%/50% ttable.0.4 0.5 ttable.1.4 0.5 ttable.0.5 0.5 ttable.1.5 0.5 ## to test: "$ speaker-test -c6 -twav" and: "$ speaker-test -c2 -twav" } pcm.dmix51 { type dmix ipc_key 1024 # let multiple users share ipc_key_add_uid false # IPC permissions (octal, default 0600) # I think changing this fixed something - but I'm not sure what. ipc_perm 0660 # slave { ## this is specific to my hda_intel. Often hd:0 is just allready it; To find: $ aplay -L pcm surround51 # this rate makes my soundcard crackle # rate 44100 # this rate stops flash in firefox from playing audio, but I don't need that rate 48000 channels 6 ## Any other values in the 4 lines below seem to make my soundcard crackle, too period_time 0 period_size 1024 buffer_time 0 buffer_size 4096 } }
External Resources
More info can be found here: