PulseAudio

There are a number of front-ends available for controlling the PulseAudio daemon:

• pacmixer — Alsamixer alike for PulseAudio.

https://github.com/KenjiTakahashi/pacmixer || pacmixer^AUR

• PAmix — Ncurses PulseAudio mixer similar to pavucontrol.

https://github.com/patroclos/PAmix || pamix-git^AUR

• pa-applet — System tray applet for PulseAudio with volume bar.

https://github.com/fernandotcl/pa-applet || pa-applet-git^AUR

• plasma-pa — KDE Plasma applet for audio volume management using PulseAudio

https://invent.kde.org/plasma/plasma-pa || plasma-pa

• PulseAudio Volume Meter — Simple GTK volume meter for PulseAudio. Discontinued development.

http://0pointer.de/lennart/projects/pavumeter/ || pavumeter^AUR

• Xfce PulseAudio Panel Plugin — PulseAudio plugin for Xfce4 panel.

https://goodies.xfce.org/projects/panel-plugins/xfce4-pulseaudio-plugin || xfce4-pulseaudio-plugin

PulseAudio will first look for configuration files in the home directory , and if they are not found, the system-wide configuration from will be applied.

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 man page for additional information. Boolean options accepts any of these: true, , and 1 as well as , , and .

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 or . 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 for the details of the syntax.

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.

The main command to configure a server during runtime is . Run for a list options, or just run 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 accordingly to make the change persistent. See PulseAudio/Examples for some basic settings.

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 , and more specifically by the command , 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 "profile:" to just before the colon and first space, as shown by the command . For instance, .

It may be easier to select a "Profile" with a graphical tool like , under the "Configuration" tab, or KDE System Settings, "Multimedia/Audio and Video Settings", under the "Audio Hardware Setup" tab. Each audio "Card", which are those devices listed by the command , or again by the command , 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 and . 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 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.

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 and finally by trying the default location (). 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 to module-native-protocol-unix.

These two variables are the important ones in order for libpulse clients to locate PulseAudio if you moved its socket to somewhere else. See for more details and other useful environment variables clients will read.

PulseAudio also uses window properties on the root window of the X11 server to help find the daemon. Since environment variables cannot be modified after child processes are started, X11 properties are more flexible because they are more easily changed because they are globally shared. As long as applications receive a environment variable, it can read the most up-to-date values. X11 properties can be queried using , or with pax11publish -d to read pulse-specific properties. can also be used to update the properties from environment variables (, or 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. The following table is there only for completeness, you should not ever need to manually set these variables by hand.

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 does not exist, as it would override the file.

Also install and 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 ALSA's 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 (), disable it by executing:

# rmmod snd_pcm_oss

To enable PulseAudio DTS (Digital Theater System) via ALSA install package and enable it:

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.

Although pulseaudio-alsa contains the necessary configuration file to allow ALSA applications to use PulseAudio's default device, ALSA's 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.

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 package pulseaudio-alsa, which provides compatibility layer between ALSA applications and PulseAudio. After this your ALSA applications will use ALSA directly without being hooked by Pulse.

• Create a configuration file in to unload the autodetection modules and load back-end drivers statically. Add device parameters as follows:

• Optional: If you use you may want to control ALSA volume instead of PulseAudio volume: set 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 to control PulseAudio volume if needed.

There are multiple ways of making OSS-only programs output to PulseAudio:

Install ossp package and start .

Programs using OSS can work with PulseAudio by starting it with padsp (included with PulseAudio):

$ 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:

Make sure comes before /usr/bin in your .

Install , or if your intended program has a legacy GStreamer implementation.

OpenAL Soft should use PulseAudio by default, but can be explicitly configured to do so:

By default, OpenAL does not allow pulseaudio to move audio streams to a different device. To change this, add the allow-moves option:

Edit the libao configuration file: Be sure to remove the dev=default option of the alsa driver or adjust it to specify a specific Pulse sink name or number.

PulseEffects is a GTK advanced utility for applying several audio effects (e.g. Noise reduction, Equalizer etc.) to audio input and output.

You may need to also install its optional dependency 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 .

A collection of PulseEffects presets can be found in community presets.

If you want to use a different equalizer rather that the one integrated in #PulseEffects, there are the following options.

Install pulseaudio-equalizer-ladspa, an equalizer based on LADSPA . Launch GUI and tweak the parameters to match your expectations.

PulseAudio has an integrated 10-band equalizer system. In order to use it, install pulseaudio-equalizer and read the following instructions.

Load the equalizer sink and dbus-protocol module

$ pactl load-module module-equalizer-sink
$ pactl load-module module-dbus-protocol

To start the GUI, run .

To load the equalizer and dbus module on every boot, create a .pa file in or edit and add the following lines:

### Load the integrated PulseAudio equalizer and D-Bus module
load-module module-equalizer-sink
load-module module-dbus-protocol

Dynamic range compression can be done with #PulseEffects. Anyway 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 LADSPA is a set of plugins containing various compression modules. Install and edit the configuration as the following

You have to specify your card sink name, get it from . 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.

Other plugins can be found in Steve Harris' LADSPA Plugin Documentation.

For a more professional compressor, you can use the one developed by Calf Studio Gear. Install and edit the configuration as the following

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.

Arch does not load the PulseAudio echo-cancellation module by default, therefore, we have to add it in . First you can test if the module is present with and entering list-modules. If you cannot find a line showing you have to create:

then restart Pulseaudio:

$ pulseaudio -k
$ pulseaudio --start

and check if the module is activated by starting . Under the input device should show .

Turning on 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 before.

Here is a list of possible 'aec_args' for 'aec_method=webrtc' with their default values :

• - Analog AGC - 'Automatic Gain Control' done over changing the volume directly - Will most likely lead to distortions.
• - Digital AGC - 'Automatic Gain Control' done in post processing (higher CPU load).
• - 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 .
• - ?
• - Noise suppression.
• - VAD - Voice activity detection.
• - 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 . 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 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".
• - This can significantly reduce background noise. See
  • - Only with "beamforming=1".
  • - 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.
• - ?
  • - 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".

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:
  1. Configure -> Settings -> Check 'Advanced' check box -> Audio Input
  2. Echo: Select 'Disabled'
  3. Noise Suppression: Set slider to 'Off'
  4. Max. Aplification: Set slider to '1.0'
• TeamSpeak:
  1. Tools -> Options -> Capture
  2. Uncheck: 'Typing attenuation', 'Remove background noise', 'Echo cancellation' and 'Echo reduction (Ducking)'
• Firefox: see Firefox tweaks#Disable WebRTC audio post processing
• Steam:
  1. In window "Friends List" -> Manage friends list settings (gear symbol) -> VOICE -> Show Advanced Settings
  2. Set the following sliders to "OFF": "Echo cancellation", "Noise cancellation", "Automatic volume/gain control"
• Skype:
  1. Tools -> Settings... -> Audio & Video -> Microphone -> Automatically adjust microphone settings -> off

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:

To run the script easily from the graphical environment, you can create a desktop launcher for it.

Installing the package will allow real-time noise suppression based on RNNoise: Learning Noise Suppression . Configuration details can be found on the projects Github site . One can install Cadmus (cadmus-deb^AUR or ) which is a GUI frontend for @werman's Pulse Audio real-time noise suppression plugin.

Another alternative is which is also build on top of RNNoise. There is not only input noise cancellation but also an output.

Refer to QEMU#Creating an audio backend for a detailed guide on how to configure pulseaudio within QEMU.

Make dockapp for the 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 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

Make it switch to pulseaudio output:

$ nyxmms2 server config output.plugin pulse

and to alsa:

$ nyxmms2 server config output.plugin alsa

To make xmms2 use a different output sink, e.g.:

$ nyxmms2 server config pulse.sink alsa_output.pci-0000_04_01.0.analog-stereo.monitor

See also the official guide .

PulseAudio will automatically be used by KDE/Qt4 applications. It is supported by default in the KDE sound mixer. For more information see the KDE page in the PulseAudio wiki.

One useful tidbit from that page is that should be loaded. This usually happens automatically at login through the script ; if you find that the module is not loaded automatically you can consider adding it manually to a configuration file in . See #Switch on connect for possible conflicts with the module-switch-on-connect.

If the phonon-gstreamer backend is used for Phonon, GStreamer should also be configured as described in #GStreamer.

Audacious natively supports PulseAudio. In order to use it, set Audacious Preferences -> Audio -> Current output plugin to 'PulseAudio Output Plugin'.

Configure MPD to use PulseAudio. See also MPD/Tips and Tricks#PulseAudio.

MPlayer natively supports PulseAudio output with the option. It can also be configured to default to PulseAudio output, in for per-user, or /etc/mplayer/mplayer.conf for system-wide:

mpv supports PulseAudio same as written for #MPlayer. Configuration in per-user, or system-wide.

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:

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.

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 :

1. Add user pulse. PulseAudio daemon switches to this user after starting.
2. Optionally add user pulse to the bluetooth group, if you have it (bluez) and want PulseAudio to use bluetooth.

   # usermod -aG bluetooth pulse

3. Add group pulse-access. This group is used by PulseAudio server for access control.
4. Add users to pulse-access group, if you want them to have access to the system-wide PulseAudio instance.

Create the service in 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

Then enable at the system level. You will also need to disable the user-level PulseAudio service across the whole system:

# systemctl --global mask pulseaudio.socket

This is necessary even if you are accessing the system over SSH, to make sure the user-level PulseAudio service will never start.

For a single shell or command you can set the 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 or 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:

default-server = server1 backup

See Keyboard shortcuts#Xorg to bind the following commands to your volume keys: , 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

Set before the command (replace with the ID of the user running PulseAudio):

$ XDG_RUNTIME_DIR=/run/user/user_id paplay /usr/share/sounds/freedesktop/stereo/complete.oga

Or use machinectl:

# machinectl shell .host --uid=user_id /usr/bin/paplay /usr/share/sounds/freedesktop/stereo/complete.oga

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 or :

To adjust the volume of the X11 bell, run the following command:

$ xset b 100

100 is a percentage. This requires the package. See Autostarting for a way to run these commands automatically when the X11 session is started.

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 to your .

Some sound cards present the option of multiple analog outputs, being switchable through using Pulseaudio profiles. But switching manually can become a chore, so you can use the following commands to switch it:

$ pactl set-sink-port 'number of the card' 'port'

This will set the default output to whatever port you chose. Example:

$ pactl set-sink-port 0 "analog-output;output-speaker" 

The values can be easily obtained using:

$ pactl list

Current output can be obtained through:

$ pactl list sinks | grep "active profile"| cut -d ' ' -f 3-

This process can be automated through a simple script. This script then can be given a shortcut by the user:

This script is intended to swap between two profiles. First checking the current profile then swapping it. Users are required to change the field 'active profile' according to the language pactl reports. Users might need to change the number of the card and the output to fit their machine.

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: