PulseAudio

Related articles
/Examples
/Troubleshooting
PipeWire
PulseAudio
is a general purpose sound server intended to run as a middleware between your applications and your hardware devices, either using
Advanced Linux Sound Architecture
(ALSA) or
Open Sound System
(OSS). It also offers easy network streaming across local devices using
Avahi
if enabled. While its main purpose is to ease audio configuration, its modular design allows more advanced users to configure the daemon precisely to best suit their needs.
Note
Some confusion may occur between ALSA and PulseAudio. ALSA includes a Linux kernel component with sound card drivers, as well as a userspace component,
libasound
.
[1]
PulseAudio builds only on the kernel component, but offers compatibility with
libasound
through
pulseaudio-alsa
.
[2]
Installation
Install
the
pulseaudio
package.
Some PulseAudio modules are not included in the main package and must be installed separately if needed:
pulseaudio-alsa
for PulseAudio to manage ALSA as well, see
#ALSA
pulseaudio-bluetooth
for
Bluetooth
support (BlueZ), see the
Bluetooth headset
page
pulseaudio-equalizer
for equalizer sink (
qpaeq
)
pulseaudio-jack
for
JACK
sink, source and jackdbus detection
pulseaudio-lirc
for infrared volume control with
LIRC
pulseaudio-zeroconf
for Zeroconf (
Avahi
/DNS-SD) support
Front-ends
There are a number of front-ends available for controlling the PulseAudio daemon:
Console
ncpamixer
— Ncurses mixer for PulseAudio inspired by
pavucontrol
.
https://github.com/fulhax/ncpamixer
||
ncpamixer
AUR
pacmixer
—
alsamixer(1)
alike for PulseAudio.
https://github.com/KenjiTakahashi/pacmixer
||
pacmixer
AUR
PAmix
— Ncurses PulseAudio mixer similar to
pavucontrol
.
https://github.com/patroclos/PAmix
||
pamix
AUR
pamixer
— PulseAudio command line mixer.
https://github.com/cdemoulins/pamixer
||
pamixer
pavolume
— Simple command-line volume control for PulseAudio with libnotify messages.
https://github.com/sseemayer/pavolume
||
pavolume-git
AUR
Ponymix
— Command line mixer for PulseAudio.
https://github.com/falconindy/ponymix
||
ponymix
AUR
pulseaudio-ctl
— Control PulseAudio volume from the shell or mapped to keyboard shortcuts.
https://github.com/graysky2/pulseaudio-ctl
||
pulseaudio-ctl
AUR
pulsemixer
— CLI and curses mixer for PulseAudio. Discontinued development.
https://github.com/GeorgeFilipkin/pulsemixer
||
pulsemixer
Graphical
KMix
—
KDE
volume control application supporting several platforms including PulseAudio, system tray applet configurable.
https://apps.kde.org/kmix/
||
kmix
MicTray
— Lightweight system tray application which lets you control the microphone state and volume using PulseAudio.
https://github.com/Junker/MicTray
||
mictray
AUR
pa-applet
— System tray applet for PulseAudio with volume bar.
https://github.com/fernandotcl/pa-applet
||
pa-applet-git
AUR
pa-notify
— PulseAudio or PipeWire volume notification daemon.
https://github.com/ikrivosheev/pa-notify
||
pa-notify
AUR
pasystray
— System tray applet for PulseAudio.
https://github.com/christophgysin/pasystray
||
pasystray
plasma-pa
—
KDE
Plasma applet for audio volume management using PulseAudio.
https://invent.kde.org/plasma/plasma-pa
||
plasma-pa
PulseAudio Equalizer
— LADSPA based multiband equalizer for PulseAudio.
https://github.com/pulseaudio-equalizer-ladspa/equalizer
||
pulseaudio-equalizer-ladspa
PulseAudio Graph Control
— Electron-based volume and graph control for PulseAudio.
https://github.com/futpib/pagraphcontrol#readme
||
pagraphcontrol-git
AUR
PulseAudio Preferences
— Simple GTK configuration dialog for PulseAudio.
https://freedesktop.org/software/pulseaudio/paprefs/
||
paprefs
PulseAudio Volume Control
— Simple GTK volume control tool ("mixer") for PulseAudio.
https://freedesktop.org/software/pulseaudio/pavucontrol/
||
pavucontrol
PulseAudio Volume Control (Qt)
— Mixer for PulseAudio (Qt port of
pavucontrol
).
https://github.com/lxqt/pavucontrol-qt
||
pavucontrol-qt
PulseAudio Volume Meter
— Simple GTK volume meter for PulseAudio. Discontinued development.
http://0pointer.de/lennart/projects/pavumeter/
||
pavumeter
AUR
PulseEffects
— Audio effects for PulseAudio applications.
https://github.com/wwmm/easyeffects/tree/pulseaudio-legacy
||
pulseeffects-legacy
AUR
Volctl
— Per-application system tray applet volume control and OSD for PulseAudio.
https://buzz.github.io/volctl/
||
volctl
AUR
Xfce PulseAudio Panel Plugin
— PulseAudio plugin for
Xfce
4 panel.
https://goodies.xfce.org/projects/panel-plugins/xfce4-pulseaudio-plugin
||
xfce4-pulseaudio-plugin
Configuration
By default, PulseAudio is configured to automatically detect all sound cards and manage them. It takes control of all detected ALSA devices and redirects all audio streams to itself, making the PulseAudio daemon the central configuration point. The daemon should work mostly out of the box, only requiring a few minor tweaks.
While PulseAudio usually runs fine out of the box and requires only minimal configuration, advanced users can change almost every aspect of the daemon by either altering the default configuration file to disable modules or writing your own from scratch.
PulseAudio runs as a server daemon that can run either system-wide or on per-user basis using a client/server architecture. The daemon by itself does nothing without its
modules
except to provide an API and host dynamically loaded modules. The audio routing and processing tasks are all handled by various modules, including PulseAudio's native protocol itself (provided by
module-native-protocol-unix
). Clients reach the server through one of many protocol modules that will accept audio from external sources, route it through PulseAudio and eventually have it go out through a final other module. The output module does not have to be an actual sound output: it can dump the stream into a file, stream it to a broadcasting server such as
Icecast
, or even just discard it.
You can find a detailed list of all available modules at
PulseAudio Loadable Modules
. To enable them you can just add a line
load-module
module-name-from-list
to
~/.config/pulse/default.pa
.
Configuration files
PulseAudio will first look for configuration files in the home directory
~/.config/pulse/
, and if they are not found, the system-wide configuration from
/etc/pulse/
will be applied.
Tip
It is strongly suggested not to edit system-wide configuration files, but rather edit user ones. Create the
~/.config/pulse
directory, then copy the system configuration files into it and edit according to your need.
Make sure you keep user configuration in sync with changes to the packaged files in
/etc/pulse/
. Otherwise, PulseAudio may refuse to start due to configuration errors.
There is usually no need to add your user to the
audio
group, as PulseAudio uses
udev
and
logind
to give access dynamically to the currently "active" user. Exceptions would include running the machine headless so that there is no currently "active" user.
daemon.conf
This is the main configuration file to configure the daemon itself. It defines base settings like the default sample rates used by modules, resampling methods, realtime scheduling and various other settings related to the server process. These can not be changed at runtime without restarting the PulseAudio daemon. The defaults are sensible for most users, see the
pulse-daemon.conf(5)
man page
for additional information. Boolean options accepts any of these:
true
,
yes
,
on
and
1
as well as
false
,
no
,
off
and
0
.
Note
PulseAudio does not perform tilde expansion on paths in this file. Use absolute paths for any files.
Option
Description
daemonize
Controls whether the server will daemonize itself and return. Set to
no
when debugging so you can see the debugging information on the terminal.
resample-method
Which resampler to use when audio with incompatible sample rates needs to be passed between modules (e.g. playback of 96kHz audio on hardware which only supports 48kHz). The available resamplers can be listed with
pulseaudio --dump-resample-methods
. Choose the best tradeoff between CPU usage and audio quality for the present use-case.
Tip
In some cases PulseAudio will generate a high CPU load. This can happen when multiple streams are resampled (individually). If this is a common use-case in a workflow, it should be considered to create an additional sink at a matching sample rate which can then be fed into the main sink, resampling only once.
avoid-resampling
With
avoid-resampling = yes
, PulseAudio automatically configures the hardware to the sample rate which the application uses, if the hardware supports this sample rate (needs
PA 11
or higher)
Warning
Enabling this feature might cause audio distortion, therefore it is disabled by default, see the
release notes
for more information.
enable-remixing
When the input and output have a different channel count (for example, outputting a 6 channel movie into a stereo sink), PulseAudio can either remix all the channels (default,
yes
) or just trivially map the channels by their name (left goes to left, right to right, all others ignored) when
no
.
system-instance
If set to
yes
, run the daemon as a
system-wide
instance.
Highly discouraged
as it can introduce security issues. Useful on
Multiseat
systems, or headless systems that have no real local users. Defaults to
no
.
flat-volumes
If set to
yes
, scales the device-volume with the volume of the "loudest" application. For example, raising the VoIP call volume will raise the hardware volume and adjust the music-player volume so it stays where it was, without having to lower the volume of the music-player manually. Defaults to
no
.
Note
When enabled, this can sometimes be confusing and some applications, unaware of this feature, can set their volume to 100% at startup, potentially blowing your speakers or your ears.
realtime-scheduling
If your
kernel
supports realtime scheduling (i.e.
realtime kernel
), set this to
yes
to ensure PulseAudio can deliver low-latency glitch-free playback. You can adjust
realtime-priority
as well to have it use the correct priority, especially when
JACK
is also running on the system.
nice-level
Since PulseAudio runs in userspace and involves inter-process communication, audio can be subject to dropouts if the daemon does not have enough CPU time to process the audio. The default usually is enough, but can be tweaked to give PulseAudio the wanted priority over (or below) other applications.
exit-idle-time
If you want to run PulseAudio only when needed and use ALSA otherwise, you can set a delay in seconds after which the daemon will automatically shutdown after all clients are disconnected. Set it to -1 to disable this feature.
log-level
When debugging, you may want to increase the logging level of the daemon to see exactly why a specific module fails to load. High logging levels will sometimes print useful information such as detected minimum latency for the system, which can then be used to tweak
default-fragments
and
default-fragment-size-msec
.
default-sample-format
This usually does not need to be changed, but if your sound card's native format is different, performance and quality can be improved by setting the right format here.
default-sample-rate
The default sample rate used by PulseAudio unless overriden at module level. Change this if your sound card does not support 44100Hz or if you wish to upsample all audio. See previous note about CPU usage.
alternate-sample-rate
To fix a common limitation where movies at 48000Hz were needlessly downsampled to 44100Hz, some modules support changing their sample rate dynamically to avoid resampling when possible. See manual for more in-depth information. This usually does not need to be changed.
default-sample-channels
The default number of channels when not specified. Usually do not need any change as you can configure more channels on per-module basis.
default-fragments
Audio samples are split into multiple fragments of
default-fragment-size-msec
each. The larger the buffer is, the less likely audio will skip when the system is overloaded. On the downside this will increase the overall latency. Increase this value if you have issues.
default-fragment-size-msec
The size in milliseconds of each fragment. This is the amount of data that will be processed at once by the daemon.
default.pa
This file is a startup script and is used to configure modules. It is actually parsed and read after the daemon has finished initializing and additional commands can be sent at runtime using
pactl(1)
or
pacmd(1)
. The startup script can also be provided on the command line by starting PulseAudio in a terminal using
pulseaudio -nC
. This will make the daemon load the CLI module and will accept the configuration directly from the command line, and output resulting information or error messages on the same terminal. This can be useful when debugging the daemon or just to test various modules before setting them permanently on disk. The manual page is quite self-explanatory, consult
pulse-cli-syntax(5)
for the details of the syntax.
The default configuration also loads
module-gsettings
to apply settings specified by
paprefs
.
Tip
Rather than being a complete copy,
~/.config/pulse/default.pa
can start with the line
.include /etc/pulse/default.pa
and then just override the defaults.
Run
pacmd list-sinks | grep -Ei 'index:|name:'
to list available sinks. The present default sink is marked with an asterisk.
Edit
~/.config/pulse/default.pa
to insert/alter the set-default-sink command using the sink's name as the numbering cannot be guaranteed repeatable.
system.pa
This file is a startup script used in place of
default.pa
when PulseAudio is running in system-wide mode.
client.conf
This is the configuration file read by every PulseAudio client application. It is used to configure runtime options for individual clients. It can be used to set and configure the default sink and source statically as well as allowing (or disallowing) clients to automatically start the server if not currently running. If autospawn is enabled, clients will automatically start PulseAudio if it is not already running when a client attempts to connect to it. This can be useful if you do not want PulseAudio to always be running to conserve system resources. Otherwise, you really should have it start with your X11 session.
Configuration command
The main command to configure a server during runtime is
pacmd
. Run
pacmd --help
for a list options, or just run
pacmd
to enter the shell interactive mode and
Ctrl+d
to exit. All modifications will immediately be applied.
Once your new settings have been tested and meet your needs, edit the
default.pa
accordingly to make the change persistent. See
PulseAudio/Examples
for some basic settings.
Tip
Leave the
load-module module-default-device-restore
line in the
default.pa
file untouched. It will allow you to restart the server in its default state, thus dismissing any wrong setting.
It is important to understand that the "sources" (processes, capture devices) and "sinks" (sound cards, servers, other processes) accessible and selectable through PulseAudio depend upon the current hardware "Profile" selected. These "Profiles" are those ALSA "pcms" listed by the command
aplay -L
, and more specifically by the command
pacmd list-cards
, which will include a line "index:", a list beginning "profiles:", and a line "active profile: <...>" in the output, among other things. "Profiles" correspond to different card input/output configurations, notably the number of available input/output channels.
The "active profile" can be set with the command
pacmd set-card-profile INDEX PROFILE
, with
no
comma separating INDEX and PROFILE, where INDEX is just the number on the line "index:" and a PROFILE name is everything shown from the beginning of any line under "profiles:" to just
before
the colon and first space, as shown by the command
pacmd list-cards
. For instance,
pacmd set-card-profile 0 output:analog-stereo+input:analog-stereo
.
It may be easier to select a "profile" with a graphical tool like
pavucontrol
, under the
Configuration
tab, or KDE System Settings, under the
Sound
tab. Each audio "card", which are those devices listed by the command
aplay -l
, or again by the command
pacmd list-cards
, will have its own selectable "profile". When a "profile" has been selected, the then available "sources" and "sinks" can be seen by using the commands
pacmd list-sources
and
pacmd list-sinks
. Note that the "index" of the available sources and sinks will change each time a card profile is changed.
The selected "Profile" can be an issue for some applications, especially the Adobe Flash players, typically
/usr/lib/mozilla/plugins/libflashplayer.so
and
/usr/lib/PepperFlash/libpepflashplayer.so
. Often, these Flash players will only work when one of the Stereo profiles is selected, and otherwise, will play video with no sound, or will simply "crash". When all else fails, you might try selecting a different profile.
Of course, when configuring some variation of Surround Sound in PulseAudio, the appropriate Surround profile will have to be selected, before Surround Sound will work, or in order to do things like remap the speaker channels.
If the only profile you seem to have is "HiFi", this means that you are using
ALSA Use Case Manager
profiles instead of PulseAudio profiles. See
PulseAudio/Examples#Disabling UCM/"HiFi"
for information on how to get back to using PulseAudio profiles.
Connection and authentication
Since PulseAudio runs as a daemon as the current user, clients needs to know where to find the daemon socket to connect to it as well as a shared random cookie file clients use to authenticate with it. By default, clients should be able to locate the daemon without problem using environment variables, X11 root window properties, the
default-server
option in
client.conf
and finally by trying the default location
$XDG_RUNTIME_DIR/pulse/native
(typically
unix:/run/user/
user-id
/pulse/native
). However, if you have clients that needs to access PulseAudio outside of your X11 session like
mpd
running as a different user, you will need to tell it how to connect to your PulseAudio instance. See
PulseAudio/Examples#Allowing multiple users to share a PulseAudio daemon
for a complete example. An authentication cookie containing random bytes is enabled by default to ensure audio does not leak from one user to another on a multi-user system. If you already control who can access the server using user/group permissions, you can disable the cookie by passing
auth-cookie-enabled=0
to
module-native-protocol-unix
.
Environment variables
These two variables are the important ones in order for libpulse clients to locate PulseAudio if you moved its socket to somewhere else. See
pulseaudio(1)
for more details and other useful environment variables clients will read.
Variable
Definition
PULSE_SERVER
Defines where the server is. It takes a protocol prefix like
unix:
or
tcp
followed by the path or IP of the server. Example:
unix:/home/pulse/native-sock
.
PULSE_COOKIE
Point this to the location of a file that contains the random cookie generated by PulseAudio. This file will be read by clients and its content sent to the server, thus the file has to be readable by all audio clients. It does not need to be the same file, as long as its content matches the one the daemon uses.
X11 properties
When using SSH X11 forwarding (i.e. when the
DISPLAY
and
SSH_CONNECTION
environment variables
are present), libpulse clients also use window properties on the root window of the X11 server to help find the daemon.
X11 properties can be queried using
xprop -root
, or with
pax11publish -d
to read pulse-specific properties.
pax11publish
can also be used to update the properties from environment variables (
pax11publish -e
, or
pax11publish -r
to remove them entirely). If possible, it is recommended to let PulseAudio do it by itself using the
module-x11-publish
module or the
start-pulseaudio-x11
command.
Variable
Definition
PULSE_SERVER
String value (
xprop -root -f PULSE_SERVER 8s -set PULSE_SERVER 127.0.0.1:4713
or
pax11publish -e -S 127.0.0.1:4713
), works the same as the environment variable of the same name.
PULSE_COOKIE
String value that contains the hexadecimal representation of the authentication cookie.
Running
PulseAudio on Arch has
pulseaudio.socket
enabled by default for the
systemd/User
instance. This means that PulseAudio will automatically start when needed.
Note
To disable
pulseaudio.socket
,
mask
the
pulseaudio.socket
user unit
. This will allow you to have
pulseaudio
installed without applications using it, e.g. you do not need sound or you are using an alternate sound server.
This systemd-based approach takes precedence over the
autospawn
option described in
pulse-client.conf(5)
.
pulseaudio
comes with option disabled by default, so you need not worry about setting it yourself.
[3]
Many
desktop environments
support
XDG Autostart
. In those desktop environments, PulseAudio will be launched automatically regardless of the socket activation status.
For more information, see
PulseAudio: Running
.
Stopping
Stop
the
pulseaudio.socket
and
pulseaudio.service
user units
.
Back-end configuration
ALSA
Warning
Do
not
attempt to change the ALSA configuration files while using the default PulseAudio configuration. The default configuration grabs the hardware devices directly in order to allow all the on-the-fly configurations using the GUIs. Changes to the ALSA configurations will very likely be ignored by PulseAudio and ALSA applications will break randomly while trying to access an ALSA device already used by PulseAudio. If you intend to change the ALSA configurations, also configure PulseAudio manually to output to your own ALSA device and play nice with your configuration.
If you have applications that do not support PulseAudio explicitly but rely on ALSA, these applications will try to access the sound card directly via ALSA and will therefore bypass PulseAudio. PulseAudio will thus not have access to the sound card any more. As a result, all applications relying on PulseAudio will not be working any more, leading to
this issue
. To prevent this, you will need to install the
pulseaudio-alsa
package. It contains the necessary
/etc/alsa/conf.d/99-pulseaudio-default.conf
for configuring ALSA to use PulseAudio. Also make sure that
~/.asoundrc
does not exist, as it would override the
/etc/asound.conf
file.
Also install
lib32-libpulse
and
lib32-alsa-plugins
if you run a x86_64 system and want to have sound for 32-bit
multilib
programs like
Wine
and
Steam
.
To prevent applications from using
OSS emulation
and bypassing PulseAudio (thereby preventing other applications from playing sound), make sure the module
snd_pcm_oss
is not being loaded at boot. If it is currently loaded (
lsmod | grep oss
), disable it by executing:
# rmmod snd_pcm_oss
Enable DTS via ALSA
To enable PulseAudio DTS (Digital Theater System) via ALSA install
dcaenc
AUR
package and enable it:
/etc/asound.conf
<confdir:pcm/dca.conf>
Finally restart PulseAudio. If experience volume issues with your DTS device and/or PulseAudio, you may fix it by looking for more setting option at
dcaenc's GitLab
.
Expose PulseAudio sources, sinks and mixers to ALSA
Although
pulseaudio-alsa
contains the necessary configuration file to allow ALSA applications to use PulseAudio's default device, ALSA's
pulse
plugin is more versatile than that:
~/.asoundrc (or /etc/asound.conf)
# Create an alsa input/output using specific PulseAudio sources/sinks
pcm.pulse-example1 {
    type pulse
    device "my-combined-sink" # name of a source or sink
    fallback "pulse-example2" # if combined not available
}

pcm.pulse-example2 {
    type pulse
    device "other-sound-card" # name of a source or sink
    # example: device "alsa_output.pci-0000_00_1b.0.analog-stereo"
}

# Create an ALSA mixer using specific PulseAudio sources/sinks
# these can be tested with "alsamixer -D pulse-example3"
ctl.pulse-example3 {
    type pulse
    device "my-output" # name of source or sink to control

    # example: always control the laptop speakers:
    # device "alsa_output.pci-0000_00_1b.0.analog-stereo"
    fallback "pulse-example4" # supports fallback too
}

# Mixers also can control a specific source and sink, separately:
ctl.pulse-example4 {
    type pulse
    sink "my-usb-headphones"
    source "my-internal-mic"

    # example: output to HDMI, record using internal
    sink "alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1"
    source "alsa_input.pci-0000_00_1b.0.analog-stereo"
}

# These can override the default mixer (example: for pnmixer integration)
ctl.!default {
    type pulse
    sink "alsa_output.pci-0000_01_00.1.hdmi-stereo-extra1"
    source "alsa_input.pci-0000_00_1b.0.analog-stereo"
}
The
source code
can be read to know all available options.
ALSA/dmix without grabbing hardware device
Note
This section describes alternative configuration, which is generally
not
recommended.
You may want to use ALSA directly in most of your applications while still being able to use applications which require PulseAudio at the same time. The following steps allow you to make PulseAudio use dmix instead of grabbing ALSA hardware device.
Remove the
pulseaudio-alsa
package, which provides compatibility layer between ALSA applications and PulseAudio. After this your ALSA applications will use ALSA directly without being hooked by PulseAudio.
In
/etc/pulse/default.pa
, comment or delete the
load-module module-udev-detect
line and add the following lines:
/etc/pulse/default.pa
load-module module-alsa-sink device=dmix
load-module module-alsa-source device=dsnoop
Optional:
If you use
kmix
you may want to control ALSA volume instead of PulseAudio volume: set
KMIX_PULSEAUDIO_DISABLE=1
as an
environment variable
.
Now, reboot your computer and try running ALSA and PulseAudio applications at the same time. They both should produce sound simultaneously.
Use
pavucontrol
to control PulseAudio volume if needed.
OSS
There are multiple ways of making OSS-only programs output to PulseAudio:
ossp
Install the
ossp
package and
start
osspd.service
.
padsp wrapper
Programs using OSS can work with PulseAudio by starting it with
padsp(1)
(included with
libpulse
):
$ padsp
OSSprogram
A few examples:
$ padsp aumix
$ padsp sox foo.wav -t ossdsp /dev/dsp
You can also add a custom wrapper script like this:
/usr/local/bin/
OSSprogram
#!/bin/sh
exec padsp /usr/bin/
OSSprogram
"$@"
Make sure
/usr/local/bin
comes before
/usr/bin
in your
PATH
.
Note
Some applications (e.g.
Factorio
with the OSS audio driver) do not work when the
module-udev-detect
has the option
tsched=0
.
GStreamer
gst-plugins-good
provides the
pulseaudio
plugin for applications using
GStreamer
.
OpenAL
OpenAL Soft should use PulseAudio by default, but can be explicitly configured to do so:
/etc/openal/alsoft.conf
drivers=pulse,alsa
libao
Edit the libao configuration file:
/etc/libao.conf
default_driver=pulse
Be sure to remove the
dev=default
option of the
alsa
driver or adjust it to specify a specific PulseAudio sink name or number.
Note
You could possibly also keep the libao standard of outputting to the
alsa
driver and its default device if you install
pulseaudio-alsa
since the ALSA default device then
is
PulseAudio.
Audio post-processing
PulseEffects
PulseEffects
is a GTK advanced utility for applying several audio effects (e.g. Noise reduction, Equalizer etc.) to audio input and output.
Note
PulseEffects new version (
EasyEffects
) only supports
PipeWire
. You need to install the legacy version (
pulseeffects-legacy
AUR
or
pulseeffects-legacy-git
AUR
) to use it with PulseAudio.
You may need to also install its optional dependency
lsp-plugins
in order to get plugins to work. If PulseEffects plugins are greyed out after installing plugins, trying to start the daemon produces an error, or no devices are shown in the
Settings > PulseAudio
tab, consider clearing the cache as shown in
[4]
.
A collection of PulseEffects presets can be found in
community presets
.
Equalization
If you want to use a different equalizer rather that the one integrated in
#PulseEffects
, there are the following options.
LADSPA module
Install
pulseaudio-equalizer-ladspa
, an equalizer based on LADSPA
swh-plugins
. Launch
pulseaudio-equalizer-gtk
GUI and tweak the parameters to match your expectations.
Integrated module
PulseAudio has an integrated 10-band equalizer system. In order to use it, install
pulseaudio-equalizer
and load
module-equalizer-sink
:
Warning
PulseAudio equalizer module is considered
unstable and might be removed from PulseAudio
.
$ pactl load-module module-equalizer-sink
Also load
module-dbus-protocol
if your configuration does not load it by default.
To start the GUI, run
qpaeq
.
Note
If
qpaeq
has no effect, install
pavucontrol
and change "ALSA Playback on" to "FFT based equalizer on ..." while the media player is running.
To load the equalizer module on every boot, create a
.pa
file in
/etc/pulse/default.pa.d/
or edit
~/.config/pulse/default.pa
and add the following lines:
### Load the integrated PulseAudio equalizer
load-module module-equalizer-sink
Note
The equalizer sink needs to be loaded after the master sink is already available.
Dynamic Range Compression
Dynamic range compression
can be done with
#PulseEffects
, however PulseEffects might introduce much overhead and latency to audio stream, so if you only need a compression effect and a minor load on the system, other options are available using a
module-ladspa-sink
.
Steve Harris plugin
Steve Harris LADSPA is a set of plugins containing various compression modules. Install
swh-plugins
and edit the configuration as the following:
~/.config/pulse/default.pa
.include /etc/pulse/default.pa

set-default-sink
your_card_sink_name
load-module module-ladspa-sink sink_name=shw_sc4 sink_master=
your_card_sink_name
plugin=sc4_1882 label=sc4 control=,,,,,,
set-default-sink shw_sc4
You have to specify your card sink name, get it from
pacmd list-sinks
. In order to apply the changes, stop and restart PulseAudio. The above configuration has empty control options using the default values.
To tweak the module with custom control parameters, fill them respecting the right order.
Control option
Description
RMS/peak (0/1)
The blanace between the RMS and peak envelope followers. RMS is generally better for subtle, musical compression and peak is better for heavier, fast compression and percussion.
Attack time (ms)
The attack time in milliseconds.
Release time (ms)
The release time in milliseconds.
Threshold level (dB)
The point at which the compressor will start to kick in.
Ratio (1:n)
The gain reduction ratio used when the signal level exceeds the threshold. 1 means no compression; higher values stronger compression.
Knee radius (dB)
The distance from the threshold where the knee curve starts.
Makeup gain (dB)
Controls the gain of the makeup input signal in decibels.
Other plugins can be found in
Steve Harris' LADSPA Plugin Documentation
.
Calf plugin
For a more professional compressor, you can use the one developed by
Calf Studio Gear
[
dead link
2025-08-16—SSL error]
. Install
calf-ladspa
AUR
and edit the configuration as the following
~/.config/pulse/default.pa
.include /etc/pulse/default.pa

set-default-sink
your_card_sink_name
load-module module-ladspa-sink sink_name=calf_comp_x2 sink_master=
your_card_sink_name
plugin=veal label=Compressor control=,,,,,,,,,,
set-default-sink calf_comp_x2
The plugin has 11 control options. If you want to insert custom values, read the following table and do not forget to specify them in the right order.
Control option
Default
Min
Max
Type
Info
Bypass
0
0
1
Bool
Level in
1
0.015625
64
Float db
Threshold
0.125
0.000976563
1
Float dbFs
For example, to set -18 db, the right value is 10^(-18/20) = 0.158
Ratio
2
1
20
Float
Attack
20
0.01
2000
Float ms
Release
250
0.01
2000
Float ms
Makeup
1
1
64
Float db
Knee
2.828427125
1
8
Float db
RMS/Peak
0
0
1
Bool
0 = RMS; 1 = Peak
Stereo Link
0
0
1
Bool
0 = Average; 1 = Max
Mix
1
0
1
Float
Percentage
To understand the meaning of every single option, read the
Calf Compressor Documentation
[
dead link
2025-08-16—SSL error]
.
Microphone echo/noise cancellation
Arch does not load the PulseAudio echo-cancellation module by default, therefore, we have to add it in
/etc/pulse/default.pa.d/
. First you can test if the module is present with
pacmd
and entering
list-modules
. If you cannot find a line showing
name: <module-echo-cancel>
you have to create:
/etc/pulse/default.pa.d/noise-cancellation.pa
### Enable Echo/Noise-Cancellation
load-module module-echo-cancel use_master_format=1 aec_method=webrtc aec_args="analog_gain_control=0 digital_gain_control=1" source_name=echoCancel_source sink_name=echoCancel_sink
set-default-source echoCancel_source
set-default-sink echoCancel_sink
then restart PulseAudio:
$ pulseaudio -k
$ pulseaudio --start
and check if the module is activated by starting
pavucontrol
. Under
Recording
, the input device should show
Echo-Cancel Source Stream from
.
Turning on
beamforming=1
in the
aec_args
can also significantly reduce background noise if you have more than one microphone (which is common on many new laptops). However, beamforming requires specifying your
mic_geometry
(see below).
If you want existing streams to be automatically moved to the new sink and source, you have to load the
module-switch-on-connect
with
ignore_virtual=no
before.
Note
If you plug in a USB sound card or headset, or you have for example a 5.1 Speaker configuration and plug in a headset on your front audio connectors after you have loaded the
module-echo-cancel
, you have to manually unload and load the
module-echo-cancel
again, because unfortunately there is no way to tell the module that it should automatically switch to the new default 'source_master' and 'source_sink'. See
[5]
.
Possible 'aec_args' for 'aec_method=webrtc'
This article or section is out of date.
Reason:
Some args were
removed
in PulseAudio 17.0. (Discuss in
Talk:PulseAudio
)
Here is a list of possible 'aec_args' for 'aec_method=webrtc' with their default values
[6]
[7]
:
analog_gain_control=1
- Analog AGC - 'Automatic Gain Control' done over changing the volume directly - Will most likely lead to
distortions
.
digital_gain_control=0
- Digital AGC - 'Automatic Gain Control' done in post processing (higher CPU load).
experimental_agc=0
- Allow enabling of the webrtc experimental AGC mechanism.
agc_start_volume=85
- Initial volume when using AGC - Possible values 0-255 - A too low initial volume may prevent the AGC algorithm from ever raising the volume high enough
[8]
.
high_pass_filter=1
- ?
noise_suppression=1
- Noise suppression.
voice_detection=1
- VAD - Voice activity detection.
extended_filter=0
- The extended filter is more complex and less sensitive to incorrect delay reporting from the hardware than the regular filter. The extended filter mode is disabled by default, because it seemed produce worse results during double-talk
[9]
. Enable this option if your microphone or speaker has a larger latency, for example, if you use a wireless microphone or some HDMI TVs as speaker.
intelligibility_enhancer=0
- Some bits for webrtc intelligibility enhancer.
drift_compensation=0
- Drift compensation to allow echo cancellation between different devices (such as speakers on your laptop and the microphone on your USB webcam). - only possible with "mobile=0".
beamforming=0
- This can significantly reduce background noise. See
[10]
[11]
mic_geometry=x1,y1,z1,x2,y2,z2
- Only with "beamforming=1".
target_direction=a,e,r
- Only with "beamforming=1". Note: If the module does not want to load with this argument, set azimuth (a) to the desired value, but set elevation (e) and radius (r) to 0.
mobile=0
- ?
routing_mode=speakerphone
- Possible Values "quiet-earpiece-or-headset,earpiece,loud-earpiece,speakerphone,loud-speakerphone" - only valid with "mobile=1".
comfort_noise=1
- ? - only valid with "mobile=1".
Disable audio post processing in certain applications
If you are using the
module-echo-cancel
, you probably do not want other applications to do additional audio post processing. Here is a list for disabling audio post processing in following applications:
Mumble:
Configure > Settings > Audio Input
Echo Cancellation
: select
Disabled
Noise suppression
: select
Disabled
Max. Amplification
: set slider to
1.00
TeamSpeak:
Tools -> Options -> Capture
Uncheck: 'Typing attenuation', 'Remove background noise', 'Echo cancellation' and 'Echo reduction (Ducking)'
Firefox: see
Firefox tweaks#Disable WebRTC audio post processing
Steam:
Settings > Voice > Advanced Options > Show
Disable the following sliders:
Echo cancellation
,
Noise cancellation
,
Automatic volume/gain control
Script for reloading module-echo-cancel
Since the module-echo-cancel is not always needed, or must be reloaded if the source_master or sink_master has changed, it is nice to have a easy way to load or reload the module-echo-cancel.
Create
the following script and make it
executable
:
echoCancelEnable.sh
#!/bin/sh
aecArgs="$*"
# If no "aec_args" are passed on to the script, use this "aec_args" as default:
[ -z "$aecArgs" ] && aecArgs="analog_gain_control=0 digital_gain_control=1"
newSourceName="echoCancelSource"
newSinkName="echoCancelSink"

# "module-switch-on-connect" with "ignore_virtual=no" (needs PulseAudio 12 or higher) is needed to automatically move existing streams to a new (virtual) default source and sink.
if ! pactl list modules short | grep "module-switch-on-connect.*ignore_virtual=no" >/dev/null 2>&1; then
	echo Load module \"module-switch-on-connect\" with \"ignore_virtual=no\"
	pactl unload-module module-switch-on-connect 2>/dev/null
	pactl load-module module-switch-on-connect ignore_virtual=no
fi

# Reload "module-echo-cancel"
echo Reload \"module-echo-cancel\" with \"aec_args=$aecArgs\"
pactl unload-module module-echo-cancel 2>/dev/null
if pactl load-module module-echo-cancel use_master_format=1 aec_method=webrtc aec_args=\"$aecArgs\" source_name=$newSourceName sink_name=$newSinkName; then
	# Set a new default source and sink, if module-echo-cancel has loaded successfully.
	pacmd set-default-source $newSourceName
	pacmd set-default-sink $newSinkName
fi
To run the script easily from the graphical environment, you can create a
desktop launcher
for it.
Recurrent neural network noise suppression (RNNoise)
Installing the package
noise-suppression-for-voice
will allow real-time noise suppression based on RNNoise: Learning Noise Suppression
[12]
. Configuration details can be found on the projects Github site
[13]
.
An alternative is
noisetorch
AUR
which is built on top of RNNoise. There is not only input noise cancellation but also an output.
Applications
QEMU
Refer to
QEMU#Creating an audio backend
for a detailed guide on how to configure PulseAudio within
QEMU
.
AlsaMixer.app
Install
pulseaudio-alsa
and make
alsamixer.app
AUR
dockapp for the
windowmaker
use PulseAudio, e.g.:
$ AlsaMixer.app --device pulse
Here is a two examples where the first one is for ALSA and the other one is for PulseAudio. You can run multiple instances of it. Use the
-w
option to choose which of the control buttons to bind to the mouse wheel.
$ AlsaMixer.app -3 Mic -1 Master -2 PCM --card 0 -w 1
$ AlsaMixer.app --device pulse -1 Capture -2 Master -w 2
See
#ALSA
for details.
XMMS2
Make it switch to PulseAudio output:
$ xmms2 server config output.plugin pulse
and to ALSA:
$ xmms2 server config output.plugin alsa
To make XMMS2 use a different output sink, e.g.:
$ xmms2 server config pulse.sink alsa_output.pci-0000_04_01.0.analog-stereo.monitor
See also
the official guide
.
KDE Plasma Workspaces and Qt
PulseAudio will automatically be used by KDE / Qt applications. It is supported by default in the KDE sound mixer. For more information see the
KDE page in the PulseAudio wiki
.
If the phonon-gstreamer backend is used for
Phonon
, GStreamer should also be configured as described in
#GStreamer
.
Audacious
Audacious
natively supports PulseAudio. In order to use it, set
File > Settings > Audio > Output plugin > PulseAudio Output
.
Music Player Daemon (MPD)
Configure
MPD
to use PulseAudio. See also
Music Player Daemon/Tips and tricks#PulseAudio
.
MPlayer
MPlayer
natively supports PulseAudio output with the
-ao pulse
option. It can also be configured to default to PulseAudio output, in
~/.mplayer/config
for per-user, or
/etc/mplayer/mplayer.conf
for system-wide:
/etc/mplayer/mplayer.conf
ao=pulse
mpv
mpv
supports PulseAudio same as written for
#MPlayer
. Configuration in
~/.config/mpv/mpv.conf
per-user, or
/etc/mpv/mpv.conf
system-wide.
guvcview
guvcview
when using the PulseAudio input from a
Webcam
may have the audio input suspended resulting in no audio being recorded.  You can check this by executing:
$ pactl list sources
If the audio source is "suspended" then create the folowing
.pa
file:
/etc/pulse/default.pa.d/no-module-suspend-on-idle.pa
unload-module module-suspend-on-idle
And then either restarting PulseAudio or your computer will only idle the input source instead of suspending it.  guvcview will then correctly record audio from the device.
Networked audio
This article or section is a candidate for merging with
PulseAudio/Examples#PulseAudio over network
.
Notes:
No need for two separate sections. (Discuss in
Talk:PulseAudio
)
One of PulseAudio's unique features is its ability to stream audio from clients over TCP to a server running the PulseAudio daemon reliably within a LAN. Ensure that client and server systems agree on the time (i.e., use NTP), or audio streams may be choppy or may not work at all. For a more detailed guide visit the
official PulseAudio documentation
.
To enable the TCP module on the server (the computer that actually outputs sound), create the following
.pa
file:
/etc/pulse/default.pa.d/tcp.pa
load-module module-native-protocol-tcp
Or you can use the
paprefs
GUI application (root is not required). Go to
Network Server > Enable network access to local sound devices
.
To make sure
module-native-protocol-tcp
is loaded on the server, you can use:
$ pacmd list-modules | grep module-native-protocol-tcp
It is a requirement that both the client and server share the same cookie. Ensure that the clients and server share the same cookie file found under
~/.config/pulse/cookie
. It does not matter whose cookie file you use (the server or a client's), just that the server and client(s) share the same one.
If it is undesirable to copy the cookie file from clients, anonymous clients can access the server by passing
auth-anonymous
to
module-native-protocol-tcp
on the server (again in
/etc/pulse/default.pa.d/
):
load-module module-native-protocol-tcp auth-anonymous=1
It is also possible to authenticate based on client IP address:
load-module module-native-protocol-tcp auth-ip-acl=127.0.0.1;192.168.0.0/24
Change the LAN IP subnet to match that of those clients you wish to have access to the server.
Starting system-wide on boot
The PulseAudio daemon normally starts as a user service when a user logs in and attempts to play some sort of audio. For running a dedicated PulseAudio server accepting client connections over TCP, the daemon must be started on boot as a system service.  Note
that in most desktop use cases, system mode likely is not the right choice
.
To run PulseAudio in a system mode, first we need to set up users and groups needed by system-wide PulseAudio server instance
[14]
:
Add user
pulse
. PulseAudio daemon switches to this user after starting.
# useradd -d /var/run/pulse -s /usr/bin/nologin -G audio pulse
Optionally add user
pulse
to the
bluetooth
group, if you have it (BlueZ) and want PulseAudio to use Bluetooth.
# usermod -aG bluetooth pulse
Add group
pulse-access
. This group is used by PulseAudio server for access control.
# groupadd pulse-access
Add users to
pulse-access
group, if you want them to have access to the system-wide PulseAudio instance.
# usermod -aG pulse-access root
Create the service
pulseaudio.service
in
/etc/systemd/system
containing the following:
/etc/systemd/system/pulseaudio.service
[Unit]
Description=Sound Service

[Service]
# Note that notify will only work if --daemonize=no
Type=notify
ExecStart=/usr/bin/pulseaudio --daemonize=no --exit-idle-time=-1 --disallow-exit=true --system --disallow-module-loading
Restart=always

[Install]
WantedBy=default.target
Tip
When PulseAudio starts in the system mode, it will change its user and group from
root
to
pulse
.
[15]
The user and group name is
hard-coded
: starting the service as pulse user does not work and PulseAudio complains that it is not root.
Then
enable
pulseaudio.service
at the system level. You will also need to disable the user-level PulseAudio service across the whole system by
masking
the
pulseaudio.socket
with the
--global
flag.
This is necessary even if you are accessing the system over SSH, to make sure the user-level PulseAudio service will never start.
When PulseAudio starts in the system mode,
/etc/pulse/system.pa
is used instead of
default.pa
, so be sure to put any necessary settings in
system.pa
.
Selecting the server
For a single shell or command you can set the
PULSE_SERVER
environment variable
to the host name or IP address of the desired PulseAudio server:
$ env PULSE_SERVER=
server-hostname-or-ip
mplayer test.mp3
Alternatively, you can create or modify
~/.config/pulse/client.conf
or
/etc/pulse/client.conf
to set a default-server persistently:
default-server =
server-hostname-or-ip
It is also possible to specify multiple servers separated by spaces which are subsequently tried by PulseAudio
[16]
:
default-server =
server1
backup
Tips and tricks
This article or section is a candidate for merging with
PulseAudio/Examples
.
Notes:
Same topic. (Discuss in
Talk:PulseAudio
)
Keyboard volume control
Map
the following commands to your volume keys:
XF86AudioRaiseVolume
,
XF86AudioLowerVolume
and
XF86AudioMute
.
First find out which sink corresponds to the audio output you would like to control.
To list available sinks:
$ pactl list sinks short
Suppose sink 0 is to be used, to raise the volume:
$ sh -c "pactl set-sink-mute 0 false ; pactl set-sink-volume 0 +5%"
To lower the volume:
$ sh -c "pactl set-sink-mute 0 false ; pactl set-sink-volume 0 -5%"
To mute/unmute the volume:
$ pactl set-sink-mute 0 toggle
To mute/unmute the microphone:
$ pactl set-source-mute 1 toggle
Tip
To have keyboard shortcuts operate always on the default sink, specify
@DEFAULT_SINK@
as the sink number, for example
pactl set-sink-mute @DEFAULT_SINK@ toggle
.
For more advanced control, such as limiting the maximum volume, consider using one of the
console front-ends
.
Play sound from a non-interactive shell (systemd service, cron)
Replace
user
with the user running PulseAudio:
# machinectl shell --uid=
user
.host /usr/bin/paplay /usr/share/sounds/freedesktop/stereo/complete.oga
X11 Bell Events
To get PulseAudio to handle X11 bell events, run the following commands after the X11 session has been started:
$ pactl upload-sample /usr/share/sounds/freedesktop/stereo/bell.oga bell-window-system
$ pactl load-module module-x11-bell display=$DISPLAY
Or use configuration files
/etc/pulse/default.pa.d/
or
~/.config/pulse/default.pa
:
~/.config/pulse/default.pa
.include /etc/pulse/default.pa

# audible bell
load-sample-lazy bell-window-system /usr/share/sounds/freedesktop/stereo/bell.oga
load-module module-x11-bell
To adjust the volume of the X11 bell, run the following command:
$ xset b 100
100 is a percentage. This requires the
xorg-xset
package. See
Autostarting
for a way to run these commands automatically when the X11 session is started.
Switch on connect
The
switch-on-connect
module switches the output sound to new devices when connected. For example, if you plug in a USB headset, the output will be switched to that. If you unplug it, the output will be set back to the last device.
This module is disabled by default for being too aggressive, but can be enabled by adding the line
load-module module-switch-on-connect
to your
~/.config/pulse/default.pa
.
Note
This does not work with some devices, see
PulseAudio/Troubleshooting#ALSA channels mute when headphones are plugged/unplugged improperly
.
Script for switching output ports
Some sound cards present the option of multiple analog outputs, being switchable through using PulseAudio ports. But switching manually can become a chore, so you can automate this with the
pactl
command.
List available ports:
$ LC_ALL=C.UTF-8 pactl list sinks
Sink #0
	...
	Ports:
		analog-output-speaker: Speakers (type: Speaker, priority: 10000, availability unknown)
		analog-output-headphones: Headphones (type: Headphones, priority: 9900, not available)
	Active Port: analog-output-speaker
Current port can be obtained through:
$ LC_ALL=C.UTF-8 pactl list sinks | grep "Active Port" | cut -d ' ' -f 3-
Switch the active port:
$ pactl set-sink-port
sink-index
port
This process can be automated through a simple script. This script then can be given a shortcut by the user:
~/pa.sh (or anything the user wants)
#!/bin/sh

# This script uses notify-send from the libnotify package to warn the user of the currently swapped to port.
# User could adapt it to their needs or change it.

CURRENT_PORT=$(LC_ALL=C.UTF-8 pactl list sinks | grep "Active Port" | cut -d ' ' -f 3-)

if [ "$CURRENT_PORT" = "analog-output-speaker" ] ; then
  pactl set-sink-port 0 "analog-output-headphones"
  notify-send --expire-time=2000 "PulseAudio" "Headphones"
else
  pactl set-sink-port 0 "analog-output-speaker"
  notify-send --expire-time=2000 "PulseAudio" "Speakers"
fi
This script is intended to swap between two ports. First checking the current port then swapping it. Users might need to change the sink index number and the port names to fit their machine.
Disable muting media on entering voice call (module-role-cork)
When entering a voice call (e.g. in Microsoft Teams, maybe others too) any media applications might be muted. To disable this behaviour you can simply disable this module in PulseAudio configuration:
/etc/pulse/default.pa.d/no-cork.pa
unload-module module-role-cork
Advanced configuration and use cases
See
PulseAudio/Examples
.
Troubleshooting
See
PulseAudio/Troubleshooting
.
See also
PulseAudio official website
, including documentation
PulseAudio under the hood