Advanced Linux Sound Architecture
Related articles
The Advanced Linux Sound Architecture (ALSA) is a Linux kernel component which replaced 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.
Contents
- 1 Installation
- 2 Unmuting the channels
- 3 Configuration
- 4 High quality resampling
- 5 Upmixing/downmixing
- 6 Dmix
-
7 Troubleshooting
- 7.1 No sound in VirtualBox
- 7.2 Sound skipping while using dynamic CPU frequency scaling
- 7.3 Problems with availability to only one user at a time
- 7.4 Simultaneous playback problems
- 7.5 Random lack of sound on startup
- 7.6 Specific program problems
- 7.7 Model settings
- 7.8 Conflicting PC speaker
- 7.9 No microphone input
- 7.10 Crackling in microphone
- 7.11 Setting the default microphone/capture device
- 7.12 Internal microphone not working
- 7.13 No sound with onboard Intel sound card
- 7.14 No headphone sound with onboard Intel sound card
- 7.15 No sound when S/PDIF video card is installed
- 7.16 Poor sound quality or clipping
- 7.17 Pops when starting and stopping playback
- 7.18 S/PDIF output does not work
- 7.19 HDMI
- 7.20 HP TX2500
- 7.21 Sound skipping during playback
- 7.22 Crackling sound with USB devices
- 7.23 Error 'Unknown hardware' appears after a kernel update
- 7.24 HDA analyzer
- 7.25 No sound with SDL
- 7.26 Low sound workaround
- 7.27 Popping sound after resuming from suspension
- 7.28 Output is muted after reboot
- 7.29 Volume is too low
- 7.30 Verifying output parameters
- 7.31 Removing old ALSA state file (asound.state)
- 7.32 No sound in applications that use OpenAL
- 8 Tips and tricks
- 9 See also
Installation
ALSA is included in the default Arch Linux kernel (linux) as a set of modules, so manual installation 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 muted (as is the upstream default).
Users with a local login (at a virtual terminal or a display manager) have permission to play audio and change mixer levels. To allow this for a remote login, the user has to be added to the audio
group. Membership in the audio
group also allows direct access to devices, which can lead to applications grabbing exclusive output (breaking software mixing) and breaks fast-user-switching, and multiseat. Therefore, adding a user to the audio
group is not recommended, unless you specifically need to[1].
User space utilities
Install the alsa-utils package from the official repositories, which (among other utilities) contains the programs alsamixer
and amixer
. alsamixer provides an ncurses based interface for shell based configuration of audio devices. Also install the alsa-plugins package if you want high quality resampling, upmixing/downmixing and other advanced features.
If you want OSS applications to work with dmix install the alsa-oss package as well. Load the kernel modules snd_seq_oss
, snd_pcm_oss
and snd_mixer_oss
to enable OSS emulation.
Unmuting the channels
ALSA by default has all channels muted, all of which will need to be unmuted manually. This can be done using amixer:
$ amixer sset Master unmute
This can alternatively be done using alsamixer:
$ alsamixer
The MM
label 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 ←
and →
keys and unmute them by pressing the m
key. Use the ↑
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.
To get full 5.1 or 7.1 surround sound you will 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.
To enable your microphone, switch to the Capture tab with F4
and enable a channel with Space
.
To exit alsamixer, press Esc
.
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 audio is being outputted to the wrong device, try manually specifying it with the argument -D
.
$ speaker-test -D default -c 8
-D
accepts PCM channel names as values, which can be retrieved by running the following:
$ aplay -L | grep :CARD
default:CARD=PCH # 'default' is the PCM channel name sysdefault:CARD=PCH front:CARD=PCH,DEV=0 surround21:CARD=PCH,DEV=0 surround40:CARD=PCH,DEV=0 surround41:CARD=PCH,DEV=0 surround50:CARD=PCH,DEV=0 surround51:CARD=PCH,DEV=0 surround71:CARD=PCH,DEV=0
If that does not work, consult the Configuration and then Troubleshooting sections.
The alsa-utils package includes the systemd unit configuration files alsa-restore.service
and alsa-store.service
, which are activated during boot and before shutdown respectively. They do not need to be manually enabled using systemctl
.
For reference, ALSA stores its settings in /var/lib/alsa/asound.state
.
Configuration
Basic syntax
ALSA configuration files follow a simple syntax consisting of hierarchical value to parameter (key) assignments. Below are (modified) excerpts from asoundrc.txt, which is usually found in alsa-lib
package but can be also reached here.
Assignments and Separators
Assignments define a value of a given key. There are different assignment types and styles available.
Simple assignment
# This is a comment. Everything after the '#' symbol to the end of the line will be ignored by ALSA. key = value # Equal signs are usually left out, since space can also be used as an separator. key value # Equivalent to the example above.
Separators are used to indicate the start and end of an assignment, but using commas or whitespace is also possible.
Seperators
# The following three assignments are equivalent. key value0; key valueN; key value0, key valueN, key value0 key valueN key value0 key valueN
Compound assignments use braces as separators.
Compound assignment
key { subkey0 value0; subkeyN valueN; } key.subkey0 value0; # Equivalent to the example above. key.subkeyN valueN;
For easier reading, it is recommended to use first style for definitions including more then three keys.
Array definitions use brackets as separators.
Single array
key [ "value0"; "valueN"; ] key.0 "value0"; # Equivalent to the example above key.N "valueN";
Everything depends on user preferences when it comes to different styles of configuration, however one should avoid mixing different styles. Further information on basic configuration can be found here.
Data types
ALSA uses different data types for parameter values, which must be set in the users respective configuration file. Some keys accept multiple data types, while most do not. A list of configuration options and their respective type requirements for PCM plugins can be found here
Operation modes
There are different operation modes for parsing nodes, the default mode is merge and create. If operation mode is either merge/create or merge type checking is done. Only same type assignments can be merged, so strings cannot be merged with integers. Trying to define a simple assignment in default operation mode to a compound (and vice versa) will also not work.
Prefixes of operation modes:
- "+" -- merge and create
- "-" -- merge
- "?" -- do not override
- "!" -- override
Operation modes
# Merge/create - If a node does not exist, it is created. If it does exist and types match, # subkeyN is merged into key. key.subkeyN valueN; # Merge/create - Equivalent to above key.+subkeyN valueN; # Merge - Node key.subkeyN must already exist and must have same data type key.-subkeyN valueN; # No override - Ignore new assignment if key.subkeyN node already exists key.?subkeyN valueN; # Override - Removes subkeyN and all keys below it, then creates node key.subkeyN key.!subkeyN valueN;
Using override operation mode, when done correctly, is usually safe, however one should bear in mind, that there might be other necessary keys in a node for proper functioning.
An example of setting default device using "defaults" node
Assuming that "defaults" node is set in /usr/share/alsa/alsa.conf, where defaults.pcm.card and its ctl counterpart have assignment values "0" (type integer), user wants to set default pcm and control device to (third) sound card "2" or "SB" for an Azalia sound card.
Defaults node
defaults.ctl.card 2; # Sets default device and control to third card (counting begins with 0). defaults.pcm.card 2; # This does not change the data type. defaults.ctl.+card 2; # Equivalent to above. defaults.pcm.+card 2; defaults.ctl.-card 2; # Same effect on a default setup, however if defaults node was removed or defaults.pcm.-card 2; # type has been changed merge operation mode will result in no changes. defaults.pcm.?card 2; # This does nothing, since this assignment already exists. defaults.ctl.?card 2; defaults.pcm.!card "SB"; # The override operation mode is necessary here, because of defaults.ctl.!card "SB"; # different value types.
Using double quotes here automatically sets values data type to string, so in the above example setting defaults.pcm.!card "2" would result in retaining last default device, in this case card 0. Using double quotes for strings is not mandatory as long as no special characters are used, which ideally should never be the case. This may be irrelevant in other assignments.
Nesting
Sometimes it may be useful and even easier to read using nesting in configuration.
Nesting PCM plugins
pcm.azalia { type hw; card 0 } pcm.!default { type plug; slave.pcm "azalia" } # is equivalent to pcm.!default { type plug; slave.pcm { type hw; card 0; } } # which is also equivalent to pcm.!default.type plug; pcm.default.slave.pcm.type hw; pcm.default.slave.pcm.card 0;
Including configuration files
Include other configuration files
</path/to/configuration-file> # Include a configuration file <confdir:/path/to/configuration-file> # Reference to a global configuration directory
Set the default sound card
If your sound card order changes on boot, you can specify their order in any file ending with .conf
in /etc/modprobe.d
(/etc/modprobe.d/alsa-base.conf
is suggested).
For example, if you want your mia sound card to be #0:
/etc/modprobe.d/alsa-base.conf
options snd_mia index=0 options snd_hda_intel index=1
Use $ cat /proc/asound/modules
to get the loaded sound modules and their order. This list is usually all that is needed for the loading order. Use $ lsmod | grep snd
to get a devices & modules list. This configuration assumes you have one mia sound card using snd_mia
and one (e.g. onboard) card using snd_hda_intel
.
You can also provide an index of -2
to instruct ALSA to never use a card as the primary one. Distributions such as Linux Mint and Ubuntu use the following settings to avoid USB and other "abnormal" drivers from getting index 0
:
/etc/modprobe.d/alsa-base.conf
options bt87x index=-2 options cx88_alsa index=-2 options saa7134-alsa index=-2 options snd-atiixp-modem index=-2 options snd-intel8x0m index=-2 options snd-via82xx-modem index=-2 options snd-usb-audio index=-2 options snd-usb-caiaq index=-2 options snd-usb-ua101 index=-2 options snd-usb-us122l index=-2 options snd-usb-usx2y index=-2 options snd-pcsp index=-2 options snd-usb-audio index=-2
These changes require a system reboot.
See also [2]
Select the default PCM via environment variable
In your configuration file, preferably global, add:
pcm.!default { type plug slave.pcm { @func getenv vars [ ALSAPCM ] default "hw:Audigy2" } }
You need to replace the default line with the name of your card (in the example is Audigy2
). You can get the names with aplay -l
or you can also use PCMs like surround51. But if you need to use the microphone it is a good idea to select full-duplex PCM as default.
Now you can start programs selecting the sound card just changing the environment variable ALSAPCM
. It works fine for all program that do not allow to select the card, for the others ensure you keep the default card.
For example, assuming you wrote a downmix PCM called mix51to20
you can use it with mplayer using the commandline ALSAPCM=mix51to20 mplayer example_6_channel.wav
Instead of using new variables, you could set one of those mentioned in default global configuration.
/usr/share/alsa/alsa.conf
Variable name # Definition ALSA_CARD # pcm.default pcm.hw pcm.plughw ctl.sysdefault ctl.hw rawmidi.default rawmidi.hw hwdep.hw ALSA_CTL_CARD # ctl.sysdefault ctl.hw ALSA_HWDEP_CARD # hwdep.default hwdep.hw ALSA_HWDEP_DEVICE # hwdep.default hwdep.hw ALSA_PCM_CARD # pcm.default pcm.hw pcm.plughw ALSA_PCM_DEVICE # pcm.hw pcm.plughw ALSA_RAWMIDI_CARD # rawmidi.default rawmidi.hw ALSA_RAWMIDI_DEVICE # rawmidi.default rawmidi.hw
Alternative method
First you will have to find out the card and device id that you want to set as the default:
$ 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
For example, the last entry in this list has the card ID 2 and the device ID 0. To set this card as the default, you can either use the system-wide file /etc/asound.conf
or the user-specific file ~/.asoundrc
. You may have to create the file if it does not exist. Then insert the following options with the corresponding card.
pcm.!default { type hw card 2 } ctl.!default { type hw card 2 }
In most cases it is recommended to use sound card names instead of number references, which also solves boot order problem. Therefore the following would be correct for the above example.
pcm.!default { type hw card Audio } ctl.!default { type hw card Audio }
To get valid ALSA card names, use aplay:
$ aplay -l | awk -F \: '/,/{print $2}' | awk '{print $1}' | uniq
PCH
Alternatively use cat, which might return unused devices:
$ cat /proc/asound/card*/id
PCH ThinkPadEC
The 'pcm' options affect which card and device will be used for audio playback while the 'ctl' option affects which card is used by control utilities like alsamixer .
The changes should take effect as soon as you (re-)start an application (MPlayer etc.). You can also test with a command like aplay.
$ aplay -D default your_favourite_sound.wav
If you receive an error regarding your asound configuration, check the upstream documentation for possible changes to the config file format.
Verifying correct 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.
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. To solve this, you can try loading the modules manually:
- Locate the module for your sound card: ALSA Soundcard Matrix The module will be prefixed with 'snd-' (for example:
snd-via82xx
). - Load the module.
- Check for the device files in
/dev/snd
(see above) and/or try ifalsamixer
oramixer
have reasonable output. - Configure
snd-NAME-OF-MODULE
andsnd-pcm-oss
to load at boot.
Getting S/PDIF output
S/PDIF is a digital audio interface often used to connect a computer to a digital amplifier (such as a home theatre with 5.1/7.1 surround sound).
Depending on what shell you use, add the following line to your shell's configuration file:
amixer -c 0 cset name='IEC958 Playback Switch' on
You can see the name of your card's digital output with:
$ amixer scontrols
System-wide equalizer
Using ALSAEqual (provides UI)
Install the alsaequal package from the Arch User Repository.
After installing the package, add the following to your ALSA configuration file:
/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 { # If you do not want the equalizer to be your # default soundcard comment the following # line and uncomment the above line. (You can # choose it as the output device by addressing # it with specific apps,eg mpg123 -a equal 06.Back_In_Black.mp3) pcm.!default { type plug; slave.pcm plugequal; }
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 ~/.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 for example, you can make a symlink to your .alsaequal.bin
in his home...
Managing ALSAEqual states
Install alsaequal-mgr from the Arch User Repository.
Configure the equalizer as usual with
$ alsamixer -D equal
When you are satisfied with the state, you may give it a name ("foo" in this example) and save it:
$ alsaequal-mgr save foo
The state "foo" can then be restored at a later time with
$ alsaequal-mgr load foo
You can thus create different equalizer states for games, movies, music genres, VoIP apps, etc. and reload them as necessary.
See the project page and the help message for more options.
Using mbeq
Install the alsa-plugins, ladspa and swh-plugins packages if you do not already have them.
If you have not already created either an ~/.asoundrc
or a /etc/asound.conf
file, then create either one and insert the following:
/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 { # The following setting is just an 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 are 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" }
High quality resampling
When software mixing is enabled, ALSA is forced to resample everything to the same frequency (48 kHz by default when supported). By default, it will try to use the speexrate converter to do so, and fallback to low-quality linear interpolation if it is not available[3]. Thus, if you are getting poor sound quality due to bad resampling, the problem can be solved by simply installing alsa-plugins.
For even higher quality resampling, you can change the default rate converter to speexrate_medium
or speexrate_best
. Both perform well enough that in practice it does not matter which one you choose, so using the best converter is usually not worth the extra CPU cycles it requires.
To change the default converter place the following contents in your ~/.asoundrc
or /etc/asound.conf
:
/etc/asound.conf
defaults.pcm.rate_converter "speexrate_medium"
Upmixing/downmixing
Upmixing
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. We will use the upmix
plugin, included in 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.
The following example adds a new PCM channel that you can use for upmixing. If you want all sound sources to go through this channel, 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 this is not working, you have to setup your own dmixer for the upmixing PCM like this:
pcm.dmix6 { type asym playback.pcm { type dmix ipc_key 567829 slave { pcm "hw:0,0" channels 6 } } }
and use "dmix6" instead of "surround71". If you experience skipping or distorted sound, consider increasing the buffer_size (to 32768, for example) or use a high quality resampler.
Downmixing
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, use the vdownmix
plugin, included in alsa-plugins..
Again, in your configuration file, add this:
pcm.!surround51 { type vdownmix slave.pcm "default" } pcm.!surround40 { type vdownmix slave.pcm "default" }
Dmix
Mixing enables multiple applications to output sound at the same time. Most discrete sound cards support hardware mixing, which is enabled by default if available. Integrated motherboard sound cards (such as Intel HD Audio), usually do not support hardware mixing. On such cards, software mixing is done by an ALSA plugin called dmix
. This feature is enabled automatically if hardware mixing is unavailable.
To manually enable dmix, add the following to your ALSA configuration file:
/etc/asound.conf
pcm.dsp { type plug slave.pcm "dmix" }
Troubleshooting
No sound in VirtualBox
If you experience problems with VirtualBox, the following command might be helpful:
$ alsactl init
Found hardware: "ICH" "SigmaTel STAC9700,83,84" "AC97a:83847600" "0x8086" "0x0000" Hardware is initialized using a generic method
You might need to activate the ALSA output in your audio software as well.
Sound skipping while using dynamic CPU 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 it is 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 from 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 { ...
Simultaneous playback problems
If you are having problems with simultaneous playback, and if PulseAudio is installed, its default configuration is set to "hijack" the soundcard. Some users of ALSA may not want to use PulseAudio and are quite content with their current ALSA settings. One fix is to edit /etc/asound.conf
and comment out the following lines:
# Use PulseAudio by default pcm.!default { type pulse fallback "sysdefault" hint { show on description "Default ALSA Output (currently PulseAudio Sound Server)" } }
Commenting the following out also may help:
ctl.!default { type pulse fallback "sysdefault" }
This may be a much simpler solution than completely uninstalling PulseAudio.
Effectively, here is an example of a working /etc/asound.conf
:
pcm.dmixer { type dmix ipc_key 1024 ipc_key_add_uid 0 ipc_perm 0660 } pcm.dsp { type plug slave.pcm "dmix" }
Random lack of sound on startup
You can quickly test sound by running speaker-test
. If there is no sound, the error message might look something like
ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave Playback open error: -16 Device or resource busy
If you have no sound on startup, this may be because your system has multiple sound cards, and their order may sometimes change on startup. If this is the case, try setting the default sound card.
If you use mpd and the configuration tips above do not work for you, try reading this instead.
Timidity
Timidity can be the cause of missing audio. Try running:
$ systemctl status timidity
If it failed, try # killall -9 timidity
. If this solves the issue, then you should disable the timidity daemon to be started at boot.
Specific program problems
For other programs who insist on their own audio setup, e.g., XMMS or MPlayer, you would need to set their specific options.
For MPlayer or mpv, add the following line to the respective configuration file:
ao=alsa
For XMMS2, 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
, e.g.:
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 are 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 are hearing something when you record
As the pulseaudio wrapper is shown as "default" in alsamixer, you 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 does not 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. If you have enabled the snd_pcm_oss
, snd_mixer_oss
or snd_seq_oss
kernel modules previously (they are not loaded by default), try unloading them.
See also:
Crackling in microphone
If you are getting a crackling or popping from your microphone that cannot be resolved with ALSA settings or cleaning your microphone jack, try adding the following line to /etc/modprobe.d/modprobe.conf
:
options snd-hda-intel model=MODEL position_fix=3
This option will fix crackling on pure ALSA, but will cause issues to pulseaudio. To let Pulse use these settings effectively, edit /etc/pulse/default.pa
and find this line:
load-module module-udev-detect
And change it to this:
load-module module-udev-detect tsched=0
See the DMA-Position Problem in the kernel docs.
Setting the default microphone/capture device
Some applications (Pidgin, Adobe Flash) do not provide an option to change the capture device. It becomes a problem if your microphone is on a separate device (e.g. 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:
pcm.usb { type hw card U0x46d0x81d } pcm.!default { type asym playback.pcm { type plug slave.pcm "dmix" } capture.pcm { 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 the volume is enabled under the Capture
view in alsamixer. Add the following to /etc/modprobe.d/snd-hda-intel.conf
:
options snd-hda-intel enable_msi=1
Then reload the module:
# rmmod snd-hda-intel && modprobe snd-hda-intel
Now there should be an additional input under the previously mentioned Capture
view.
No sound with onboard Intel sound card
There may be a problem with two conflicting modules loaded, namely snd-intel8x0
and snd-intel8x0m
. In this case, blacklist snd-intel8x0m:
/etc/modprobe.d/modprobe.conf
blacklist snd-intel8x0m
Muting the "External Amplifier" in alsamixer
or amixer
may also help. See the ALSA wiki.
Unmuting the "Mix" setting in the mixer might help, also.
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:
- dell-m6
- dell-vostro
- generic
- laptop
- laptop-hpsense
- olpc-xo-1_5
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.
A list of available models is also available here. To know your chip name type the following command (with * being corrected to match your files). Note that some chips could have been renamed and do not directly match the available ones in the file.
$ grep Codec /proc/asound/card*/codec*
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 then we can use the enable
parameter from snd_hda_intel module; it is an array of booleans that can enable/disable the desired sound card.
options snd_hda_intel enable=1,0
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.
If snd-usb-audio driver has been loaded, you could try to enable softvol
in /etc/asound.conf
file. Example configuration for the first audio device:
pcm.!default { type plug slave.pcm "softvol" } pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:0" period_size 4096 buffer_size 131072 rate 50000 } bindings { 0 0 1 1 } } pcm.dsnooper { type dsnoop ipc_key 1024 slave { pcm "hw:0" channels 2 period_size 4096 buffer_size 131072 rate 50000 } bindings { 0 0 1 1 } } pcm.softvol { type softvol slave { pcm "dmixer" } control { name "Master" card 0 } } ctl.!default { type hw card 0 } ctl.softvol { type hw card 0 } ctl.dmixer { type hw card 0 }
Pops when starting and stopping playback
Some modules (e.g. snd_ac97_codec and snd_hda_intel) can power off your sound card when not in use. This can make an audible noise (like a crack/pop/scratch) when turning on/off your sound card. Sometimes even when move the slider volume, or open and close windows (KDE4). If you find this annoying try modinfo snd_MY_MODULE
, and look for a module option that adjusts or disables this feature.
Example: disable the power saving mode and solve cracking sound trough speakers problem, using snd_hda_intel add in /etc/modprobe.d/modprobe.conf
:
options snd_hda_intel power_save=0
or:
options snd_hda_intel power_save=0 power_save_controller=N
You can also try it with modprobe snd_hda_intel power_save=0
before.
You may also have to unmute the 'Line' ALSA channel for this to work. Any value will do (other than '0' or something too high).
Example: on an onboard VIA VT1708S (using the snd_hda_intel module) these cracks occured even though 'power_save' was set to 0. Unmuting the 'Line' channel and setting a value of '1' solved the problem.
Source: https://www.kernel.org/doc/Documentation/sound/alsa/powersave.txt
If you use a laptop, pm-utils will change power_save
back to 1 when you go onto battery power even if you disable power saving in /etc/modprobe.d
. Disable this for pm-utils by disabling the script that makes the change (see Disabling a hook for more information):
# touch /etc/pm/power.d/intel-audio-powersave
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 the following:
$ iecset audio on
You can also put this command in an enabled systemd service as it sometimes it may stop working after a reboot.
HDMI
Output does not work
The procedure described below can be used to test HDMI audio. Before proceeding, make sure you have enabled and unmuted the output with alsamixer
.
Connect your PC to the Display via HDMI cable and enable the display with xrandr.
Use aplay -l
to get the discover the card and device number. For example:
$ aplay -l
**** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC892 Analog [ALC892 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 1: ALC892 Digital [ALC892 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0
Send sound to the device. Following the example in the previous step, you would send sound to card 1
, device 3
:
$ aplay -D plughw:1,3 /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. If you are using a standalone window manager, you may need to have sound playing while plugging in the HDMI cable.
mplay and other application could be configured to use special HDMI device as audio output. But flashplugin could only use default device. The following method is used to override default device. But you need to change it back when your TV is disconnected from HDMI port.
If the test is successful, create or edit your ~/.asoundrc
file to set HDMI as the default audio device.
~/.asoundrc
pcm.!default { type hw card 1 device 3 }
Or if the above configuration does not work try:
~/.asoundrc
defaults.pcm.card 1 defaults.pcm.device 3 defaults.ctl.card 1
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:
5.1 sound goes to wrong speakers
Sound can be redirected to the intended speakers using ALSA's remap
function. To do so, add the following to /etc/asound.conf
:
pcm.!hdmi-remap { type asym playback.pcm { type plug slave.pcm "remap-surround51" } } pcm.!remap-surround51 { type route slave.pcm "hw:0,3" ttable { 0.0= 1 1.1= 1 2.4= 1 3.5= 1 4.2= 1 5.3= 1 } }
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)
Sound skipping during playback
Run alsamixer, and if channels exist for nonexistent output devices then disable them (e.g. alsamixer showing a center speaker but you not having one).
Crackling sound with USB devices
Try modifying the parameters of the snd-usb-audio
kernel_module for minimal latency via modprobe.conf
.
/etc/modprobe.d/modprobe.conf
options snd-usb-audio nrpacks=1
For more information, see the following: Tuning USB devices for minimal latency
Error 'Unknown hardware' appears after a kernel update
The following messages may be displayed during ALSA's initialization:
Unknown hardware "foo" "bar" ... Hardware is initialized using a guess method /usr/bin/alsactl: set_control:nnnn:failed to obtain info for control #mm (No such file or directory)
or:
Found hardware: "HDA-Intel" "VIA VT1705" "HDA:11064397,18490397,00100000" "0x1849" "0x0397" Hardware is initialized using a generic method /usr/bin/alsactl: set_control:1328: failed to obtain info for control #1 (No such file or directory) /usr/bin/alsactl: set_control:1328: failed to obtain info for control #2 (No such file or directory) /usr/bin/alsactl: set_control:1328: failed to obtain info for control #25 (No such file or directory) /usr/bin/alsactl: set_control:1328: failed to obtain info for control #26 (No such file or directory)
Simply store ALSA mixer settings again:
# alsactl -f /var/lib/alsa/asound.state store
It may be necessary configure ALSA again with alsamixer
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.
No sound with SDL
If you get no sound using SDL based applications, try setting the environment variable SDL_AUDIODRIVER
to alsa
.
Low sound workaround
If you are facing low sound even after maxing out your speakers/headphones, you can give the softvol plugin a try. Add the following to /etc/asound.conf
.
pcm.!default { type plug slave.pcm "softvol" } pcm.softvol { type softvol slave { pcm "dmix" } control { name "Pre-Amp" card 0 } min_dB -5.0 max_dB 20.0 resolution 6 }
After the changes are loaded successfully, you will see a Pre-Amp
section in alsamixer. You can adjust the levels there.
Popping sound after resuming from suspension
You might hear a popping sound after resuming the computer from suspension. This can be fixed by editing /etc/pm/sleep.d/90alsa
and removing the line that says aplay -d 1 /dev/zero
Output is muted after reboot
Run the following command:
# alsactl restore
If the problem persists, verify that the Auto-Mute
option in alsamixer is set to Disabled
.
Volume is too low
Run |alsamixer and try to increase the value of the sliders, unmuting channels if necessary. Note that if you have many sliders, you may have to scroll to the right to see any missing sliders.
If all the sliders are maxed out, and the volume is still too low, you can try running the script at http://www.alsa-project.org/hda-analyzer.py to reset your codec settings:
$ wget http://www.alsa-project.org/hda-analyzer.py $ su -c 'python2 hda-analyzer.py'
The script assumes the python
binary refers to Python 2, which is incorrect on Arch by default. To avoid this issue run the following command:
$ sed -i 's/python %s/python2 %s/' hda-analyzer_run.py
Close the analyzer, and when prompted as to whether you want to reset the codecs, say "yes".
If the volume is *still* too low, run alsamixer
again: resetting the codecs may have caused new sliders to become enabled, and some of them may be set to a low value.
Verifying output parameters
Check the contents of /proc/asound/cardX/pcmYp/subZ/hw_params
, where X
, Y
, and Z
are system dependent.
In order to find this file, execute the following command while outputting anything via ALSA:
$ find /proc/asound/ -name hw_params | xargs -I FILE grep -v -l "closed" FILE | grep '/proc/asound/card./pcm.p/sub./hw_params'
If nothing is playing there should be no results.
Here is an example output for audio with a bit depth of 24 bits and a sampling frequency of 44.1 kilohertz:
cat /proc/asound/card1/pcm0p/sub0/hw_params
access: RW_INTERLEAVED format: S24_3LE subformat: STD channels: 2 rate: 44100 (44100/1) period_size: 5513 buffer_size: 22050
More info is available in the ALSA documentation.
Removing old ALSA state file (asound.state)
The alsa-utils package provides alsa-store.service
which automatically stores the current ALSA state to /var/lib/alsa/asound.state
upon system shutdown. This can be problematic for users who are trying to reset their current ALSA state as the asound.state
file will be recreated with the current state upon every shutdown (e.g., attempting to remove user-defined channels from the mixer). The alsa-store.service
service may be temporarily disabled by creating the following empty file:
# mkdir -p /etc/alsa # touch /etc/alsa/state-daemon.conf
The presence of state-daemon.conf
prevents alsa-store.service
from saving asound.state
during shutdown. After disabling this service, the asound.state
file may be removed as such:
# rm /var/lib/alsa/asound.state
After rebooting, the previous ALSA state should be lost and the current state should be reset to defaults. Re-enable alsa-store.service
by deleting the condition file we created:
# rm /etc/alsa/state-daemon.conf
On the next shutdown, the asound.state
file should be recreated with ALSA defaults. The file may also be generated immediately using:
# alsactl store
No sound in applications that use OpenAL
OpenAL defaults to pulseaudio, to change the order, add the following configuration to /etc/openal/alsoft.conf
:
drivers=alsa,pulse
Tips and tricks
Hot-plugging a USB sound card
The following udev rule can be used to automatically make a USB sound card the primary output device when the card is plugged in. To do so, create the following file with the specified contents and.
/etc/udev/rules.d/00-local.rules
KERNEL=="pcmC[D0-9cp]*", ACTION=="add", PROGRAM="/usr/bin/sh -c 'K=%k; K=$${K#pcmC}; K=$${K%%D*}; echo defaults.ctl.card $$K > /etc/asound.conf; echo defaults.pcm.card $$K >>/etc/asound.conf'" KERNEL=="pcmC[D0-9cp]*", ACTION=="remove", PROGRAM="/usr/bin/sh -c 'echo defaults.ctl.card 0 > /etc/asound.conf; echo defaults.pcm.card 0 >>/etc/asound.conf'"
For information regarding loading udev rules see the following: Udev#Loading_new_rules