PulseAudio
PulseAudio is a general purpose sound server intended to run as a middleware between your applications and your hardware devices, either using ALSA or 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.
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 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.
- pacmixer — Alsamixer alike for PulseAudio.
- PAmix — Ncurses PulseAudio mixer similar to pavucontrol.
- pamixer — PulseAudio command line mixer.
- pavolume — Simple command-line volume control for PulseAudio with libnotify messages.
- Ponymix — Command line mixer for PulseAudio.
- pulseaudio-ctl — Control PulseAudio volume from the shell or mapped to keyboard shortcuts.
- pulsemixer — CLI and curses mixer for PulseAudio
Graphical
- KMix — KDE volume control application supporting several platforms including PulseAudio, system tray applet configurable.
- MicTray — Lightweight system tray application which lets you control the microphone state and volume using PulseAudio.
- pa-applet — System tray applet for PulseAudio with volume bar.
- pasystray — System tray applet for PulseAudio.
- plasma-pa — KDE Plasma applet for audio volume management using PulseAudio
- PulseAudio Equalizer — LADSPA based multiband equalizer for PulseAudio.
- PulseAudio Graph Control — Electron-based volume and graph control for PulseAudio.
- PulseAudio Manager — Simple GTK frontend for PulseAudio. Discontinued development.
- PulseAudio Preferences — Simple GTK configuration dialog for PulseAudio.
- PulseAudio Volume Control — Simple GTK volume control tool ("mixer") for PulseAudio.
- PulseAudio Volume Control (Qt) — Mixer for PulseAudio (Qt port of pavucontrol).
- PulseAudio Volume Control (Sandsmark) — Lightweight fork of the LXQt's pavucontrol, with missing features from pavucontrol implemented, bug fixes and unnecessary dependencies removed.
- PulseAudio Volume Meter — Simple GTK volume meter for PulseAudio. Discontinued development.
- PulseEffects — Audio effects for PulseAudio applications.
- Volctl — Per-application system tray applet volume control for PulseAudio.
- Xfce PulseAudio Panel Plugin — PulseAudio plugin for Xfce4 panel.
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.
- It is strongly suggested not to edit system-wide configuration files, but rather edit user ones. Create the
~/.config/pulsedirectory, 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
audiogroup, 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.
| 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), pulse 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 |
flat-volumes 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 yes upstream, but to no within Arch.
Note: The default behavior upstream 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. This is why Arch defaults to the classic (ALSA) behavior by setting this to
no. |
| realtime-scheduling | If your kernel supports realtime scheduling (for instance, Realtime kernel or Linux-ck), 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 pulse 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 user by pulse 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-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.
- Rather than being a complete copy,
~/.config/pulse/default.pacan start with the line.include /etc/pulse/default.paand 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.pato insert/alter the set-default-sink command using the sink's name as the numbering cannot be guaranteed repeatable.
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.
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 "profile:" 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, "Multimedia/Audio and Video Settings", under the "Audio Hardware Setup" 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.
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 and finally by trying the default location (unix:/run/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
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 DISPLAY= environment variable, it can read the most up-to-date values. 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. The following table is there only for completeness, you should not ever need to manually set these variables by hand.
| Variable | Definition |
|---|---|
PULSE_SERVER |
String value (xprop -root -f PULSE_SERVER 8s -set PULSE_SERVER "unix:/tmp/pulse-sock"), 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.
- To disable
pulseaudio.socket, make sure that$XDG_CONFIG_HOME/systemd/user/exists and runsystemctl --user mask pulseaudio.socket. - 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
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 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 (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 dcaencAUR 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 Github.
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
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 apps will use ALSA directly without being hooked by Pulse.
- Create a configuration file in
/etc/pulse/default.pa.d/to unload the autodetection modules and load back-end drivers statically. Add device parameters as follows:
/etc/pulse/default.pa.d/load-audio-drivers-statically.pa
unload-module module-udev-detect unload-module module-detect 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:
$ echo export KMIX_PULSEAUDIO_DISABLE=1 > ~/.kde4/env/kmix_disable_pulse.sh $ chmod +x ~/.kde4/env/kmix_disable_pulse.sh
- 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 ossp package and start osspd.service.
padsp wrapper
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:
/usr/local/bin/OSSProgram
#!/bin/sh exec padsp /usr/bin/OSSprogram "$@"
Make sure /usr/local/bin comes before /usr/bin in your PATH.
tsched=0.GStreamer
Install gst-plugins-good, or gstreamer0.10-good-pluginsAUR if your intended program has a legacy GStreamer implementation.
OpenAL
OpenAL Soft should use PulseAudio by default, but can be explicitly configured to do so:
/etc/openal/alsoft.conf
drivers=pulse,alsa
By default, OpenAL does not allow pulseaudio to move audio streams to a different device. To change this, add the allow-moves option:
/etc/openal/alsoft.conf
[pulse] allow-moves=true
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 Pulse sink name or number.
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.
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 [3].
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 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 qpaeq.
To load the equalizer and dbus 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 and D-Bus module load-module module-equalizer-sink load-module module-dbus-protocol
Dynamic Range Compression
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 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. |
| Amplitude (dB) | The level of the input signal, in decibels. |
| Gain reduction (dB) | The degree of gain reduction applied to the 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. Install calf-ladspaAUR 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. | |||||
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.
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 [4].Possible 'aec_args' for 'aec_method=webrtc'
Here is a list of possible 'aec_args' for 'aec_method=webrtc' with their default values [5][6]:
-
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 [7]. -
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 [8]. 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 [9][10]-
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 -> Check 'Advanced' check box -> Audio Input
- Echo: Select 'Disabled'
- Noise Suppression: Set slider to 'Off'
- Max. Aplification: Set slider to '1.0'
- 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:
- In window "Friends List" -> Manage friends list settings (gear symbol) -> VOICE -> Show Advanced Settings
- Set the following sliders to "OFF": "Echo cancellation", "Noise cancellation", "Automatic volume/gain control"
- Skype:
- Tools -> Settings... -> Audio & Video -> Microphone -> Automatically adjust microphone settings -> off
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/bash 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-voiceAUR will allow real-time noise suppression based on RNNoise: Learning Noise Suppression [11]. Configuration details can be found on the projects Github site [12]. One can install Cadmus (cadmus-debAUR or cadmus-appimageAUR) which is a GUI frontend for @werman's Pulse Audio real-time noise suppression plugin.
Another alternative is noisetorchAUR which is also build on top of RNNoise. There is not only input noise cancellation but also an output.
Applications
QEMU
Refer to QEMU#Host for a detailed guide on how to configure pulseaudio within QEMU.
AlsaMixer.app
Make alsamixer.appAUR dockapp for the windowmakerAUR 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
XMMS2
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 [13].
KDE Plasma Workspaces and Qt4
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 load-module module-device-manager should be loaded. This usually happens automatically at login through the script /usr/bin/start-pulseaudio-x11; if you find that the module is not loaded automatically you can consider adding it manually to a configuration file in /etc/pulse/default.pa.d/. 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
In order to use PulseAudio, set Edit → Preferences… → Devices → Playback → Device in Audacious to “default” or “pulse”. These devices are added to the drop-down list by pulseaudio-alsa.
Music Player Daemon (MPD)
Configure MPD to use PulseAudio. See also MPD/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
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
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
Enable the TCP module on the server(the computer that actually outputs sound), create the folowing .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. To do this, create pulseaudio.service in /etc/systemd/system containing the following:
[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 Restart=always [Install] WantedBy=default.target
Then enable pulseaudio.service at the system level.
# systemctl enable --now pulseaudio
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.
Selecting the Server
For a single shell or command you can set the environment variable $PULSE_SERVER 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 ~/.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[14]:
default-server = server1 backup
Tips and tricks
Keyboard volume control
See Keyboard shortcuts#Xorg to bind 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
- To have keyboard shortcuts operate always on the default sink, specify
@DEFAULT_SINK@as the sink number, for examplepactl 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)
Set XDG_RUNTIME_DIR before the command (replace user_id 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
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
This is a default enabled module used to switch the output sound to the newly connected device. 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 used to be quite buggy but got a lot of attention in PulseAudio 8.0 and should work quite well now.
Script for switching analog outputs
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:
~/pa.sh (or anything the user wants)
#!/bin/bash
# This script uses kdialog notification to warn the user of the currently swapped to profile. User could adapt it to their needs or change it.
CURRENT_PROFILE=$(pactl list sinks | grep "active profile"| cut -d ' ' -f 3-)
if [ "$CURRENT_PROFILE" = "analog-output;output-speaker" ] ; then
pactl set-sink-port 0 "analog-output;output-headphones-1"
kdialog --title "Pulseaudio" --passivepopup "Headphone" 2 &
else
pactl set-sink-port 0 "analog-output;output-speaker"
kdialog --title "Pulseaudio" --passivepopup "Speaker" 2 &
fi
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.
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