Advanced Linux Sound Architecture

Related articles
/Configuration examples
/Troubleshooting
PC speaker
PipeWire
PulseAudio
Sound system
List of applications/Multimedia#Volume control
The
Advanced Linux Sound Architecture
(ALSA) provides
kernel
driven sound card drivers. It replaces the original
Open Sound System
(OSS).
Besides the sound device drivers, ALSA also bundles a user space driven library for application developers. They can then use those ALSA drivers for high level API development. This enables direct (kernel) interaction with sound devices through ALSA libraries.
Tip
An explanation of ALSA related terminology—
interface
,
card
,
device
(a
card
is not a
device
),
subdevice
, and more—can be found on
Wikipedia:Advanced Linux Sound Architecture#Concepts
.
Installation
The ALSA drivers are part of the Linux kernel. The ALSA library (
alsa-lib
) is usually installed as a
dependency
. Therefore, manual installation is not necessary.
udev
will automatically detect your hardware and select needed drivers at boot time, therefore, your sound should already be working.
However, your sound may be initially muted. If it is, see
#Unmuting the channels
.
Firmware
The
Sound Open Firmware
(SOF) (
sof-firmware
) is usually required for laptops—they tend to utilize Cadence Tensilica Xtensa architecture
DSPs
, see the list of the
supported platforms
. In case of the missing firmware the
journal
will provide messages such as:
error: sof firmware file is missing
error: failed to load DSP firmware -2
error: sof_probe_work failed err: -2
For more SOF troubleshooting information, see
Overview of Intel hardware platforms
.
The
linux-firmware-cirrus
package is needed for laptops with Cirrus Logic
smart amplifiers
. See also:
Audio drivers for Cirrus Logic CS35L54/56/57/63 Boosted Smart Amplifiers
,
archlinux/packaging/packages/linux-firmware#19
.
The
linux-firmware-intel
package is needed for some Intel audio devices.
The
alsa-firmware
package contains firmware that may be required for
certain
sound cards.
See also
#Cards and modules
and
Linux firmware
.
ALSA utilities
Install
the
alsa-utils
package. This contains (among other utilities) the
alsamixer(1)
and
amixer(1)
utilities.
amixer
is a shell command to change audio settings, while
alsamixer
provides a more intuitive
ncurses
based interface for audio device configuration.
ALSA and systemd
The
alsa-utils
package comes with
systemd
unit configuration files
alsa-restore.service
and
alsa-state.service
by default.
These are automatically installed and activated during installation (via package provided symlink to
sound.target
). The options are as follows:
alsa-restore.service
reads
/var/lib/alsa/asound.state
on boot and writes updated values on shutdown, provided
/etc/alsa/state-daemon.conf
does not exist. As
/etc/alsa/state-daemon.conf
is not created without a conscious action of the user, it is the default method.
alsa-state.service
(Re-)Starts alsactl in daemon mode to continuously keep track of, and persist, volume changes, under the condition that the user has consciously created
/etc/alsa/state-daemon.conf
.
Evidently, both methods are mutually exclusive. You can decide for one of the two approaches depending on your requirements. To edit these units, see
systemd#Editing provided units
. You can check their status using
systemctl
.
For further information, see
alsactl(1)
.
User privileges
Local users have permission to play audio and change mixer levels. To allow remote users to use ALSA, you need to
add
those users to the
audio
group.
Note
Adding users to the
audio
group allows direct access to devices. Keep in mind that this allows applications to exclusively reserve output devices. This may break software mixing or fast-user-switching on multi-seat systems. Therefore, adding a user to the
audio
group is
not
recommended by default unless you specifically
need to
.
OSS emulation
OSS emulation
is the ability to intercept
OSS
calls and reroute them through ALSA instead. This emulation layer is useful e.g. for legacy applications which try to open
/dev/dsp
and write sound data to them directly. Without OSS or the emulation library,
/dev/dsp
will be missing, and the application will not produce any sound.
If you want OSS applications to work with
dmix
, install the
alsa-oss
package as well.
Load
the
snd_pcm_oss
and
snd_seq_oss
kernel modules. Configure them to
load at boot
.
Unmuting the channels
By default, ALSA has all channels muted. Those have to be unmuted manually.
Unmute with amixer
Unmuting the sound card's master volume can be done by using
amixer
:
$ amixer sset Master unmute
$ amixer sset Speaker unmute
$ amixer sset Headphone unmute
Unmute with alsamixer
Unmuting the sound card can be done using
alsamixer
:
$ alsamixer
The
MM
label below a channel indicates that the channel is muted, and
OO
indicates that it is open.
Scroll to the
Master
and
PCM
channels with the
Left
and
Right
keys and unmute them by pressing the
m
key.
Use the
Up
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.
Note
If gain is set above 0 dB, audible distortion can become present.
Unmute 5.1/7.1 sound
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
. (Those are channel names with
Intel HD Audio
; they may vary with different hardware)
Note
Please take note that this will not automatically upmix stereo sources (like most music). In order to accomplish that, see
#Upmixing
.
Enable the microphone
To enable your microphone, switch to the Capture tab with
F4
and enable a channel with
Space
. See
/Troubleshooting#Microphone
if microphone does not work.
Test your changes
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:PCH -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:PCH' is the PCM channel name for -D
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
Additional notes
If your system has more than one soundcard, then you can switch between them by pressing
F6
Some cards need to have digital output muted or disabled in order to hear analog sound and vice versa.
Some machines, like the Thinkpad T61, have a
Speaker
channel which must be unmuted and adjusted as well.
Some machines, like the Dell E6400, may also require the
Front
and
Headphone
channels to be unmuted and adjusted.
If your volume adjustments seem to be lost after you reboot, try running
alsamixer
as root.
Driver configuration
For more information, see
Advanced Linux Sound Architecture - Driver Configuration guide
.
To reload ALSA driver configuration you have to reload corresponding modules. Before doing this, all processes using the corresponding ALSA driver—such as
PipeWire
—must be stopped. To identify processes using sound
device files
, utilize
fuser(1)
:
# fuser --all --verbose /dev/snd/*
See also:
ALSA SoundCard Matrix
alsactl(1)
Kernel modules#Manual module handling
Cards and modules
Run
lspci -k -nn -d ::0401; lspci -k -nn -d ::0403
to identify the modules in use for PCI devices.
Tip
::0401
means
Multimedia audio controller
subclass of the
Multimedia controller
PCI device class.
::0403
means
Audio device
subclass.
Kernel driver in use
in
lspci
output is the
loaded module
.
Kernel modules
are
capable
of handling the device, see
lspci(8) § k
.
Run
lsusb --verbose --tree | grep --after-context=1 'Class=Audio'
for USB devices.
Run
lsmod | grep '^snd'
to get a full list of loaded sound modules.
Run
cat /proc/asound/cards
to get the list of your sound cards with their corresponding
indexes
(
card numbers
).
Tip
For the format of
/proc/asound/cards
, see the
General Overview
section of the ALSA library API
Control Interface
.
For general information about
/proc/asound
procfs
tree, see
Proc Files of ALSA Drivers
.
Run
cat /proc/asound/modules
to get the card indexes with their corresponding module names.
Card index
If you want to change your sound card order (or if your sound card order changes on boot, and you want to make it permanent),
reserve
the index for the given driver with the
slots
option of the
snd
module. See also
Kernel module#Setting module options
.
Tip
The name
slot
comes from the OSS and is equivalent to the ALSA term
index
.
The following sample assumes you want your USB sound card always be the first (i.e. with index
0
), no matter when the module is loaded (e.g. the card could be unplugged on boot):
/etc/modprobe.d/alsa-base.conf
options snd slots=snd_usb_audio
When a module name is prepended with an exclamation mark (
!
), the corresponding index will be given for any modules but that name. For example, reserve the first index (
0
) for any modules but
snd_usb_audio
to avoid USB sound cards from getting it:
options snd slots=!snd_usb_audio
You can also provide an index of
-2
to instruct ALSA to never use a card as the primary one: negative value is interpreted as a
bitmask of permissible indexes
. The alternative to the previous sample using the
index
option of the specific module:
options snd_usb_audio index=-2
If several sound cards use the same module, and their order is always the same, you can change the order with just
index
option. The following sample assumes there are two audio cards using the HD Audio module (e.g. an integrated audio card and HDMI output of non-integrated video card), and you want to swap their indexes:
options snd_hda_intel index=1,0
Tip
The sample above reads as "the first sound card which uses
snd_hda_intel
gives the index
1
, and the second one gives the index
0
". Which card is first and which one is second is determined by
udev
.
ALSA module option often take not just a single value, but an array of values, each one setting the option's value but only for one of the sound cards that a given module is handling. Look for the presence of the
array of
notion in the output of:
$ modinfo --field=parm
module_name
| column --separator=':' --table --table-columns-limit=2
The
slots
option can be combined with the
index
one
as long as they do not conflict
:
options snd slots=,snd_hda_intel,snd_hda_intel,snd_usb_audio,snd_usb_audio,snd_usb_audio
options snd_hda_intel index=2,1
options snd_usb_audio index=3,4,5 vid=0x
VID_3
,0x
VID_4
,0x
VID_5
pid=0x
PID_3
,0x
PID_4
,0x
PID_5
Tip
slots=,
means "do not reserve the index zero", so e.g. some USB card—not mentioned in the
snd-usb-audio
options—plugged in after a boot could be the first.
The
snd_usb_audio
module allows to specify the cards with
vid
and
pid
options, so you do not rely on the order from
udev
.
See also:
MultipleCards
MultipleUSBAudioDevices
Card disabling
To disable all cards controlled by a given kernel module,
prevent the module from loading
using
install
or
module_blacklist
approach.
Note
Blacklisting using the
blacklist
keyword in a
modprobe.d
configuration file will
not
work.
To select which card should be disabled, use the
enable
option
of a kernel module. For example, disable the second card operated by a module:
/etc/modprobe.d/alsa-base.conf
options
module_name
enable=1,0
See also
/Troubleshooting#Codec probing
for an HD Audio card codec disabling.
Library configuration
The system configuration file is
/etc/asound.conf
, and the per-user configuration file is
~/.asoundrc
.
The syntax of library configuration—i.e. whitespace, line continuation, comments, including configuration files, punctuators (separators), assignments, compound assignments, operation modes—is explained in
Configuration files
.
ALSA library configuration is loaded for each instance of the library, so to reload it, all you have to do is restart the programs that are using it.
For more information, see:
Configuring ALSA
.asoundrc article
on the ALSA unofficial wiki
.asoundrc article
on the ALSA project wiki
Basic syntax
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.
Warning
Overriding
pcm
node itself will most definitely make ALSA unusable, since every plugin definition will be deleted. Therefore,
do not use
!pcm.key
unless you are making a configuration from scratch.
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;
Set the default sound card
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 error.

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.
Note
From a configuration point of view, those are not equivalent to setting a compound "default" PCM device, since most users specify addressing type in there also, which actually may be the same, but the assignment itself still differs. Also,
defaults.pcm.card
is referred to multiple times in ALSA configuration files, usually as a fallback assignment, where different environment variables take precedence.
Setting the default sound card via defaults node
Putting the
previous example
regarding
defaults.pcm.card
and
defaults.pcm.device
into practice, assuming we have 2 cards with index 0 and 1 respectively and wish to simply change the default card to index 1, would lead to the following configuration in
/etc/asound.conf
or the user-specific
~/.asoundrc
to change both the playback and the mixer control card.
defaults.pcm.card 1
defaults.ctl.card 1
Select the default PCM via environment variable
Probably, it is enough to set ALSA_CARD to the name of the device.  First, get the names with
aplay -l
, then set ALSA_CARD to the name which comes after the colon and before the bracket; e.g. if you have
card 1: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0]
then set
ALSA_CARD=HDMI
.
Other variables are also checked in the default global configuration
/usr/share/alsa/alsa.conf
. By looking there for constructs of the form
vars [ ... ]
, the following table emerges:
Variable name
Used by
1
ALSA_CARD
pcm.default , pcm.hw , pcm.plughw , ctl.sysdefault , ctl.hw ,  rawmidi.default , rawmidi.hw , hwdep.hw
2
ALSA_CTL_CARD
ctl.sysdefault , ctl.hw
3
ALSA_HWDEP_CARD
hwdep.default , hwdep.hw
4
ALSA_HWDEP_DEVICE
hwdep.default , hwdep.hw
5
ALSA_PCM_CARD
pcm.default , pcm.hw , pcm.plughw
6
ALSA_PCM_DEVICE
pcm.hw  , pcm.plughw
7
ALSA_RAWMIDI_CARD
rawmidi.default , rawmidi.hw
8
ALSA_RAWMIDI_DEVICE
rawmidi.default , rawmidi.hw
Alternatively, you can override the behavior in your own configuration file, preferably the global one (
/etc/asound.conf
).  Add:
pcm.!default {
   type plug
   slave.pcm {
       @func getenv
       vars [ ALSAPCM ]
       default "hw:Audigy2"
   }
}
In this case as well, replace
Audigy2
with the name of your device. 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 select the sound card when starting programs by just changing the environment variable
ALSAPCM
. It works fine for all programs 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
Note
Pay attention to default addressing type.
Addressing hardware directly
Warning
This setting leaves the device unavailable to other applications. This method is only recommended if it is a part of a more sophisticated setup
~/.asoundrc
or if user deliberately wants to address sound card directly (digital output through
IEC958
or dedicated music server for example). This setting without sound server or
mixing
is not supported by some applications such as
Chromium
.
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 index (card number)
2
, card ID strings
Audio
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 Audio #or card 2
   hint {
    show on # for some applications
   }
}

ctl.!default {
   type hw
   card Audio
}

pcm.dmixer {
	type dmix
	ipc_key 2048
	slave {
		pcm "hw:Audio" #or "hw:2"
	}
}

pcm.dsnooper {
	type dsnoop
	ipc_key 2048
	slave {
		pcm "hw:Audio"
	}
}
It is recommended to use sound card ID strings instead of number references to overcome the boot order problem.
Note
This method could be problematic if your system has several cards of the same ID string (with
_1
,
_2
, … suffixes appended). For more information, see
Identify two identical audio devices
.
pcm.dmixer
and
pcm.dsnooper
are spares for applications which does not support without
mixing
.
The factual accuracy of this article or section is disputed.
Reason:
General method to select ALSA device should be introduced such as environment value. (Discuss in
Talk:Advanced Linux Sound Architecture
)
For example,
chromium -alsa-output-device=pcm.dmixer -alsa-input-device=pcm.dsnooper
enables mixing for Chromium tempolary.
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 (e.g. MPlayer). You can also test with a command like
aplay
:
$ aplay -D default:PCH
your_favourite_sound.wav
If you receive an error regarding your asound configuration, check the
upstream documentation
for possible changes to the configuration file format.
Plugins
Install
the
alsa-plugins
package if you need to enable
#Upmixing
,
#Downmixing
,
#High quality resampling
and other advanced features.
For more information, see
PCM (digital audio) plugins
.
Software mixing
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.
Note
Software mixing is enabled by default for sound cards which do not support hardware mixing.
dmix
is not enabled by default for digital output (S/PDIF) and will require the configuration snippet below.
To manually enable
dmix
, add the following to your ALSA configuration file:
/etc/asound.conf
pcm.dsp {
    type plug
    slave.pcm "dmix"
}
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 the
alsa-plugins
package.
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 the
alsa-plugins
package.
Again, in your configuration file, add this:
pcm.!surround51 {
   type vdownmix
   slave.pcm "default"
}
pcm.!surround40 {
   type vdownmix
   slave.pcm "default"
}
Note
This might not be enough to make downmixing working; see
[1]
. So, you might also need to add
pcm.!default "plug:surround51"
or
pcm.!default "plug:surround40"
. Only one
vdownmix
plug can be used; if you have 7.1 channels, you will need to use
surround71
instead the configuration above. A good example, which includes a configuration that makes both
vdownmix
and
dmix
working, can be found in
[2]
.
Multiband EQ
Multiband EQ
(mbeq)
is a fairly typical multiband graphical equalizer. It is implemented using a
fast Fourier transform
(FFT), so it takes quite a lot of CPU power, but should have less phase effects than an equivalent filter implementation. If the input signal is at too low sample rate, then the top bands will be ignored—the highest useful band will always be a high shelf.
mbeq
is part of Steve Harris'
LADSPA
plugin suite
.
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:
            #   50 Hz,  100 Hz,  156 Hz,  220 Hz,  311 Hz,   440 Hz,   622 Hz,  880 Hz,
            # 1250 Hz, 1750 Hz, 2500 Hz, 3500 Hz, 5000 Hz, 10000 Hz, 20000 Hz
            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"
}
ALSAEqual
System-wide
Install the
alsaequal
AUR
package.
After installing the package, add the following to your ALSA configuration file:
/etc/asound.conf
ctl.equal {
    type equal;
}

pcm.plugequal {
    type equal;
    # Normally, the equalizer feeds into dmix so that audio
    # from multiple applications can be played simultaneously:
    slave.pcm "plug:dmix";
    # If you want to feed directly into a device, specify it instead of dmix:
    #slave.pcm "plughw:0,0";
}

# Configuring pcm.!default will make the equalizer your default sink
pcm.!default {
# If you do not want the equalizer to be your default,
# give it a different name, like pcm.equal commented below
# Then you can choose it as the output device by addressing
# it in individual apps, for example mpg123 -a equal 06.Back_In_Black.mp3
# pcm.equal {
    type plug;
    slave.pcm plugequal;
}
To change your equalizer settings, run
$ alsamixer -D equal
Note that the equalizer configuration 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 their home directory.
Specific outputs only
If you wish to apply an equalizer to a specific output device only (for example your speakers connected to the S/PDIF output, but not your headphones connected to the headphone jack), but also want be able to output from multiple applications and to both output devices simultaneously, you need to create two
dmix
devices that feed into their respective devices (
slave.pcm
) directly. The following works for stereo output and maintains a regular stereo input, applying the equalizer to the S/PDIF output only.
/etc/asound.conf
#
#  (capture.pcm)  <-- dnsoop
#        |
# !default                               --> dmixa
#        |                               |
#  (playback.pcm) --> stereo2quad ==> quad
#                                        |
#                                        --> softvol --> plugequal --> dmixd
#

# dmix for analog output
pcm.dmixa {
  type dmix
  ipc_key 1024
  ipc_perm 0666
  slave.pcm "hw:PCH,0"
  slave {
    period_time 0
    period_size 1024
    buffer_size 4096
    channels 2
  }
  bindings {
    0 0
    1 1
  }
}

# dmix for digital output
pcm.dmixd {
  type dmix
  ipc_key 2048
  ipc_perm 0666
  slave.pcm "hw:PCH,1"
  slave {
    period_time 0
    period_size 1024
    buffer_size 4096
    channels 2
  }
  bindings {
    0 0
    1 1
  }
}

# equalizer with controls
pcm.plugequal {
  type equal
  slave {
    pcm "plug:dmixd"
  }
}
ctl.equal {
 type equal
}

# Volume control for S/PDIF
pcm.softvol {
    type softvol
    slave.pcm "plug:plugequal"
    control {
        name "S/PDIF"
    }
}

# multi:
# "a" (analog)  -> dmix,
# "d" (digital) -> softvol -> plugequal -> dmix
pcm.quad {
    type multi
    slaves {
      a.pcm "dmixa"
      a.channels 2
      d.pcm "plug:softvol" # detour via softvol and equalizer
      d.channels 2
    }
    bindings {
      0 { slave a; channel 0; }
      1 { slave a; channel 1; }
      2 { slave d; channel 0; }
      3 { slave d; channel 1; }
    }
}

# stereo to quad
pcm.stereo2quad {
  type route
  slave.pcm "quad"
  ttable [
    [ 1 0 1 0 ]
    [ 0 1 0 1 ]
  ]
}

# playback to stereo to quad, capture as usual
pcm.!default {
  type asym
  playback.pcm "plug:stereo2quad"
  capture.pcm "plug:dnsoop"
}
Managing states
Install the
alsaequal-mgr
AUR
package.
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
This, however, only restores
~/.alsaequal.bin
. You then have to update the equalizer by
alsamixer -D equal
.
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.
Tips and tricks
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. Thus, if you are getting poor sound quality due to bad resampling, the problem can be solved by simply installing the
alsa-plugins
package.
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
:
defaults.pcm.rate_converter "speexrate_medium"
Note
It is also possible to use
libsamplerate
converters, which are only about half as fast as the
speexrate
converters but do not achieve much higher quality.
It is also possible to use
Rate Converter Plugin Using libavresample
that use
FFmpeg
.
lavcrate_high
and
lavcrate_higher
are equal to
Kodi
low
and
medium
quality resamplers respectively.
Some applications (like MPlayer and its forks) do their own resampling by default because some ALSA drivers have incorrect delay reporting when resampling is enabled (hence leading to AV desynchronization), so changing this setting will not have any effect unless you configure them to use ALSA resampling.
Disabling auto mute on startup
Auto-Mute Mode
can be configured on startup with
amixer
. For example, to disable it:
# amixer -c 0 sset "Auto-Mute Mode" Disabled
Alternatively, the ncurses based interface can be utilized through
alsamixer
. In order to save any modifications, use:
# alsactl store
or
# alsactl daemon
See also
#ALSA and systemd
.
Hot-plugging a USB sound card
See
Writing Udev rules for ALSA
.
Simultaneous output
You might want to play music via external speakers connected via mini jack and internal speakers simultaneously. This can be done by unmuting
Auto-Mute
item using
alsamixer
or
amixer
:
$ amixer sset "Auto-Mute" unmute
and then unmuting other required items, such as
Headphones
,
Speaker
,
Bass Speaker
...
Note
If you have a crackling sound through headphones connector (mini-jack) after, see
/Troubleshooting#Crackling sound through mini-jack (headphones connector)
.
Keyboard volume control
Map
the following commands to your volume keys:
XF86AudioRaiseVolume
,
XF86AudioLowerVolume
,
XF86AudioMute
.
To raise the volume:
amixer set Master 5%+
To lower the volume:
amixer set Master 5%-
To toggle mute/unmute of the volume:
amixer set Master toggle
Virtual sound device using snd_aloop
You might want a jack alternative to create a virtual recording or play device in order to mix different sources, using the
snd_aloop
module:
modprobe snd_aloop
List your new virtual devices using:
aplay -l
now you can for example using ffmpeg:
ffmpeg -f alsa -i hw:1,1,0 -f alsa -i hw:1,1,1 -filter_complex amerge output.mp3
In the hw:R,W,N phrase, R is your virtual card device number. W should be set to 1 for recording devices, or 0 for playing. N is your sub device. You can use all the virtual devices available and play/stop using  applications like mplayer:
mplayer -ao alsa:device=hw=1,0,0 fileA
mplayer -ao alsa:device=hw=1,0,1 fileB
Another thing you could do with this approach is using
festival
to generate a voice into a recording stream using a script like this:
#!/bin/sh
echo "$1" | iconv -f utf-8 -t iso-8859-1 | text2wave  > "_tmp_.wav"
mplayer -ao alsa:device=hw=2,0,0 "_tmp.wav"
rm "_tmp.wav"
Reconfiguring input/output ports
The
alsa-tools
package contains the
hdajackretask
tool, which can be used (on Intel HDA cards) to reconfigure the sound card input/output ports; for instance, to turn a microphone jack into a headphone jack.
apulse
apulse
AUR
provides an
alternative partial implementation
of the
PulseAudio
API. It lets you use ALSA for applications that support only PulseAudio for sound. Usage is simply:
$ apulse
application
See also
ALSA project wiki
ALSA unofficial wiki
A close look at ALSA
Linux ALSA sound notes
Pulse-code modulation (PCM)