PulseAudio (or PA for short) is a sound server that provides a number of features on top of the low-level audio interface ALSA on Linux, such as:
- Networking support (P2P and server mode);
- Per-application volume controls;
- Better cross-platform support;
- Dynamic latency adjustment, which can be used to save power
- 1 Installation
- 2 Configuration
- 3 Troubleshooting
- 4 External resources
For motherboards containing Intel HDA sound cards, use the following kernel option for improved power-saving:
Device Drivers ---> <*> Sound card support ---> <*> Advanced Linux Sound Architecture ---> HD-Audio ---> (2048) Pre-allocated buffer size for HD-audio driver
Portage knows the
pulseaudio as a global USE flag for enabling support for PulseAudio in other packages. Enabling this USE flag will pull in media-sound/pulseaudio automatically:
USE flags for media-sound/pulseaudio A networked sound server with an advanced plugin system
||Build the X11 publish module to export PulseAudio information through X11 protocol for clients to make use. Don't enable this flag if you want to use a system wide instance. If unsure, enable this flag.||local|
||Add support for media-libs/alsa-lib (Advanced Linux Sound Architecture)||global|
||Request installing media-plugins/alsa-plugins with PulseAudio plugin enabled. This ensures that clients supporting ALSA only will use PulseAudio.||local|
||Use libasyncns for asynchronous name resolution.||local|
||Enable Bluetooth Support||global|
||Use Linux capabilities library to control privilege||global|
||Enable dbus support for anything that needs it (gpsd, gnomemeeting, etc)||global|
||Build the doxygen-described API documentation.||local|
||Enable the equalizer module (requires sci-libs/fftw).||local|
||Use sys-libs/gdbm to store PulseAudio databases. Recommended for desktop usage. This flag causes the whole package to be licensed under GPL-2 or later.||local|
||Add support to dev-libs/glib-based mainloop for the libpulse client library, to allow using libpulse on glib-based programs.||local|
||Use GConf to store user preferences on streams and so on. Don't enable this flag if you want to use a system wide instance. If unsure, enable this flag.||local|
||Add support for x11-libs/gtk+ (The GIMP Toolkit)||global|
||Add support for IP version 6||global|
||Add support for the JACK Audio Connection Kit||global|
||Use dev-libs/libressl as SSL provider (might need ssl USE flag), packages should not depend on this USE flag||global|
||Build with support for converting sample rates using libsamplerate||global|
||Add support for lirc (Linux's Infra-Red Remote Control)||global|
||Build with native HSP backend for bluez 5.||local|
||Enable optimization support for ARM NEON processors||global|
||Build with oFono HFP backend for bluez 5, requires net-misc/ofono.||local|
||Use dev-lang/orc for just-in-time optimization of array operations||global|
||Enable OSS sink/source (output/input). Deprecated, upstream does not support this on systems where other sink/source systems are available (i.e.: Linux). The padsp wrapper is now always build if the system supports OSS at all.||local|
||Add support for the Qt GUI/Application Toolkit version 4.x||global|
||Makes PulseAudio use RealtimeKit (sys-auth/rtkit) to get real-time priority while running.||local|
||!!internal use only!! Security Enhanced Linux support, this must be set by the selinux profile or breakage will occur||global|
||Add support for Sound eXchange (SoX)||global|
||Use dev-libs/openssl to provide support for RAOP (AirPort) streaming.||local|
||Allow preparation and installation of the system-wide init script for PulseAudio. Since this support is only supported for embedded situations, do not enable without reading the upstream instructions at http://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/WhatIsWrongWithSystemWide/ .||local|
||Build with sys-apps/systemd support to replace standalone ConsoleKit.||local|
||Add support for TCP wrappers||global|
||Workaround to pull in packages needed to run with FEATURES=test. Portage-2.1.2 handles this internally, so don't set it in make.conf/package.use anymore||global|
||Enable virtual/udev integration (device discovery, power and storage device support, etc)||global|
||Uses the webrtc.org AudioProcessing library for enhancing VoIP calls greatly in applications that support it by performing acoustic echo cancellation, analog gain control, noise suppression and other processing.||local|
||Builds a special paravirtualized module for running in a Xen guest.||local|
||Support for DNS Service Discovery (DNS-SD)||global|
After setting USE flags be sure to update the system so the changes take effect:
emerge --ask --changed-use --deep @world
- media-sound/pavucontrol - Pulseaudio Volume Control, a GTK+ based mixer for PulseAudio.
- media-sound/paprefs - PulseAudio Preferences, a configuration dialogue for PulseAudio.
- KDE's Phonon integrated PulseAudio configuration and mixing, but it is not as powerful as pavucontrol or paprefs.
This is the only supported configuration by upstream and Gentoo developers.
To make this possible, ACLs (Access Control Lists) are required:
File systems ---> Pseudo filesystems ---> [*] Virtual memory file system support (former shm fs) [*] Tmpfs POSIX Access Control List
If a desktop profile is not being used, check that sys-auth/consolekit or sys-apps/systemd are installed with the
acl USE flag enabled. ConsoleKit should be running on systems using OpenRC as the init system:
* status: started
If not, enable it at boot time:
rc-update add consolekit default
When finished, verify the permissions are working correctly:
getfacl /dev/snd/controlC0 | grep -Eo "user:.+:" | cut -d: -f2
For ConsoleKit/Systemd Logind managed permissions to be respected, no one may be part of the audio group (not even even the user pulse). Remove any and all users from the audio group:
gpasswd -d <user> audio
Configuring other applications
Some applications need to be configured to output to PulseAudio by default. A detailed list of these can be found on the PulseAudio wiki's PerfectSetup page.
If you previous had followed this article, upon upgrade to a fixed version (there should also be an einfo reminder), please remove the asound.conf file.
The media-plugins/alsa-plugins must be installed with the
pulseaudio USE flag enabled:
emerge --ask media-plugins/alsa-plugins
You need to enable the following module in /etc/pulse/default.pa:
Several GConf keys must be set:
- Manual with gconftool:
gconftool-2 -t string --set /system/gstreamer/0.10/default/audiosink pulsesink
gconftool-2 -t string --set /system/gstreamer/0.10/default/audiosrc pulsesrc
Some programs might still require GStreamer output plugin to be manually set to PulseAudio Audio Sink.
Enable the following module in /etc/pulse/default.pa:
Also eselect the PulseAudio implementation:
eselect esd set esdcompat
Set the following in /etc/libao.conf:
Set the following in /etc/openal/alsoft.conf:
drivers = pulse
Set the following in /etc/mplayer/mplayer.conf:
Using PulseAudio without udev/systemd is not recommended. It should be noted that this breaks both auto-detection and hot-plugging -- needlessly making your life harder -- as well as disabling ConsoleKit, essentially voiding the article. Please, if you have the possibility to use udev (and you most certainly have), just use it.
If you are using ALSA as a PulseAudio sink (output) and routing ALSA apps to PA but not using udev, you must set a specific device to be used. Otherwise, PulseAudio will use the ALSA device "default" as the sink, which may be routed back to PulseAudio, forming a loop. To avoid this, add the parameter device=hw:0,0 (you can find the correct IDs by running aplay -l). In the following example, we use two soundcards, of which card 0, device 0 is used as a sink (audio output, e.g. speakers) and card 1, device 0 as a source (audio input, e.g. microphone). PulseAudio will still be able to access other cards than these but it needs these settings to avoid looping the default device in this setup.
load-module module-alsa-sink device=hw:0,0 load-module module-alsa-source device=hw:1,0
If you forget to specify this when using the ALSA sink/source modules, you may not notice any problems until you restart PulseAudio (e.g. by logging out and back in or rebooting). You will be left with (obviously) no audio, a slow desktop environment and hanging applications until you resolve the loop, restart alsasound and kill all running pulseaudio processes.
These instructions are for setting up a headless pulse audio server. Meaning a server which has no display on it but does have speakers. This provides the ability to use the remote server's speakers for audio output.
You will get warned in a dozen places for doing this, but it is the proper method.
First configure USE flags and emerge the package. The system-wide USE flag is masked, so we have to unmask it.
mkdir -p /etc/portage/profile
echo "-system-wide" >> /etc/portage/profile/use.mask
echo "media-sound/pulseaudio system-wide" >> /etc/portage/package.use
emerge --ask --oneshot pulseaudio
Add the following two lines somewhere in the system.pa file:
load-module module-native-protocol-tcp auth-ip-acl=184.108.40.206/24 load-module module-alsa-sink
Replace 220.127.116.11/24 with the network mask that you want to be able to access the server.
If you have multiple ALSA devices, you will need to specify the device to use by adding device or device_id to the module-alsa-sink module
Tell the init script that we really do want to do this, and then start it up:
echo "PULSEAUDIO_SHOULD_NOT_GO_SYSTEMWIDE=1" >> /etc/conf.d/pulseaudio
rc-update add pulseaudio default
rc-service pulseaudio start
pacmd load-module module-tunnel-sink server=18.104.22.168
For a more permanent solution you can add the following to your default.pa file
load-module module-tunnel-sink server=22.214.171.124
Now in the pulse audio volume control you should see the remote server listed under Output Devices. Under playback you should have a button next to the Mute audio button that when clicked will let you switch that audio stream to whichever output you want.
The equalizer module is considered unstable and might be removed from pulseaudio. For more, see the mailing list.
Make sure you installed pulseaudio with the
equalizer USE flag enabled.
Enabling the required modules
Add the following two lines somewhere in the default.pa file :
load-module module-dbus-protocol load-module module-equalizer-sink
Restart the pulseaudio instance. This should be as easy as:
Choosing the equalizer sink
The command should list the index and name of the equalizer sink:
pacmd list-sinks | grep -B1 -e "name:.*equalizer"
Use pavucontrol or a similar program to select the equalizer sink for sound output. It may be listed as a device starting with FFT based equalizer.
Control the equalizer levels
The equalizer levels can now be controlled with the Qt GUI called qpaeq.
- Short sound events (e.g. the terminal bell) distort ongoing sound streams (e.g. music)
No sound after installation
If you have no sound while using ALSA, consider unmuting the sound card. Launch alsamixer and make sure each column has a green
00 under it (use the m key to toggle mute/unmute). Install media-sound/pavucontrol and check if there is any output on the pavucontrol panel when playing an audio.
Enable debug mode
To get more informations you need to set the following in /etc/pulse/daemon.conf:
Afterward restart the daemon:
Audio/Video out of sync
When using PulseAudio over your local network, you can experience out-of-sync problems. Solve this by adding tsched=0:
load-module module-udev-detect tsched=0
This disables time scheduling. Afterwards restart the daemon:
If the only playback device is the Dummy Output, PulseAudio cannot access your sound devices. Either the user has no permissions (see section Permissions or another program blocks the access. Try:
fuser -v /dev/snd/*
It shows the relevant program. Close the program and reconfigure it to use PulseAudio.
No guarantees on actual latencies
Currently PA provides whatever latency at that moment is possible be it some milliseconds to hundreds of milliseconds without regard to what applications ask for.
In case of buffer under-run latencies are never decreased
Currently, if a buffer under-run occurs, PA buffers for longer increasing latency, but it then never tries to buffer for less until restart.
Re-sampling using up a lot of CPU time
Re-sampling can require quite a lot of computational power, PA defaults are rather conservative but in certain cases can still take a significant toll, in such cases edit /etc/pulse/daemon.conf and consider changing resample-method to something less CPU intensive, default-sample-format and default-sample-rate can also affect CPU utilization with higher bit-depth and larger difference in sample-rate generally needing more resources (e.g. re-sampling 44.1 kHz to 48 kHz is faster than re-sampling either to 192 kHz). Since re-sampling is done per each channel per input, channel configuration and number of applications can affect performance as well.
Starting with version 7.0 there is also soxr resamplers made available by enabling the sox USE flag. In particular resample-method = soxr-mq should provide acceptable quality while even the higher quality and hence slower soxr-hq is still cheaper than the default speex-float-1. But be warned that the soxr resamplers have roughly 5-20 times higher latency than speex-float, in terms of time the worst case for soxr-mq/hq can be as high as 20 ms while soxr-vhq latency can in few specific setups reach over 27 ms. In terms of feeling 20 ms can range from unnoticeable to irritating depending on person and use case (the usual PA latency's lower bound is around 20-25 ms and more commonly often around 70-90 ms, for comparison).
Using a version of PA with Orc support can noticeably decrease CPU usage, too. Also PA has the ability, if certain conditions are met, to automatically switch sinks between common sampling rates, this effectively can avoid some re-sampling.