PulseAudio

PulseAudio (or PA for short) is Article description::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.

Prerequisites
PulseAudio can use but does not need any of:


 * sys-apps/systemd or
 * sys-auth/elogind (remember to add  to global USE flag).

Kernel
For motherboards containing Intel HDA sound cards, use the following kernel option for improved power-saving:

CONFIG_HIGH_RES_TIMERS is needed to avoid  errors and degraded audio when some applications are using pulseaudio:

See also the permissions section for how to configure the kernel to provide the correct permissions for PA. Also see ALSA for setting the right kernel options for sound card detection.

Global
Several packages are aware of the USE flag for enabling support for PulseAudio in other packages. Enabling this USE flag will pull in automatically:

Emerge
After setting USE flags be sure to update the system so the changes take effect:

Additional software

 * - Pulseaudio Volume Control, a GTK based mixer for PulseAudio. Alternatively, is the Qt based version.
 * - CLI and curses mixer for PulseAudio
 * - 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.
 * - replacement for  USE-flag, may have better sound quality and allows for a codec selection in

Permissions
PulseAudio uses udev to dynamically give access to the soundcards to the currently "active" user.

To make this possible, ACLs (Access Control Lists) are required:

If a desktop profile is not being used, check that  are installed with the   USE flag enabled, and (if using OpenRC) that  is installed.

When finished, verify the permissions are working correctly:

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.


 * ALSA

The must be installed with the   USE flag enabled:

You need to enable the following module in :
 * OSS

Several GConf keys must be set:
 * GStreamer


 * Manual with :

Enable the following module in :
 * ESD

Also the PulseAudio implementation:

Set the following in :
 * libao

Set the following in :
 * OpenAL

Set the following in :
 * MPlayer

Without udev or systemd
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.

Headless server
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.

Server
First configure USE flags and emerge the package. The system-wide USE flag is masked, so we have to unmask it.

Add the following two lines somewhere in the system.pa file:

Replace 1.2.3.0/24 with the network mask that you want to be able to access the server.

Tell the init script that we really do want to do this, and then start it up:

Client
For a more permanent solution you can add the following to your default.pa file

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.

Allow multiple users to use PulseAudio concurrently
In some situations, like software isolation, it may be desirable to run some programs as another users and have access to the Pulseaudio daemon. By default, the PulseAudio daemon does not accept connections by secondary users.

The following configuration runs PulseAudio daemon using a UNIX socket that accepts connections from other users.

The above configuration allows different users to access the same sound server. Users do not need to be in the audio, pulse-access or pulse groups. PulseAudio server may need to be executed at startup with the command pulseaudio -D. For example, in case you are using i3wm:

Equalizer
Make sure you installed pulseaudio with the  USE flag enabled.

Enabling the required modules
Add the following two lines somewhere in the default.pa file :

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:

Use or a similar program like  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.

Known issues

 * 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 and make sure each column has a green   under it (use the  key to toggle mute/unmute). Install  and check if there is any output on the pavucontrol panel when playing audio.
 * After upgrading the system: pulseaudio's user configuration files can become corrupt. Deleting configuration files and forcing fresh ones to be generated by restarting the daemon may fix no sound condition.

Corrupted audio
Problem: Audio has strange distortion or static when playing an output.

Solution: There is probably an issue with the Pulse server. Kill and restart the server:

Enable debug mode
To get more informations you need to set the following in :

Afterward restart the daemon:

Audio/Video out of sync
Problem: Out-of-sync problems experienced when using PulseAudio over a local network.

Solution: Add  to the  file in order to disables time scheduling:

Restart the daemon:

Dummy output
If the only playback device is the Dummy Output, PulseAudio cannot access the system sound devices. Either the user has no permissions (see section Permissions or another program is blocking access. Try:

It shows the relevant program. Close the program and reconfigure it to use PulseAudio.

No guarantees on actual latencies
Currently PA provides whatever latency is possible at the moment. This can be be milliseconds to hundreds of a millisecond without regard to what applications require.

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 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).

grsec and PulseAudio
Make sure the CONFIG_GRKERNSEC_SYSFS_RESTRICT kernel symbol is not enabled when using a grsecurity kernel. PulseAudio’s module-udev-detect needs to access to discover what cards are available on the system, and that kernel option disallows this for anyone but root.

Volume gets randomly louder or reset to 100%
This may be caused by misbehaving PulseAudio-enabled programs when PulseAudio has flat volumes feature enabled. Disable it in the daemon config:

In pavucontrol, unable to change output device for applications that use OpenALsoft
It can happen that you want to change the output device for an application, but trying to change it in pavucontrol has no effect. If this application use OpenALsoft, this one has a configuration option that inhibits sink changes.

To disable this option, you can create a configuration file for OpenALsoft:



Or, by running :

External resources

 * Getting DTS 5.1+ sound via S/PDIF or HDMI using PulseAudio
 * PulseAudio's Frequently Asked Questions - official FAQ
 * PulseAudio: The Perfect Setup
 * More general troubleshooting tips
 * Why you should care about PulseAudio (and how to start doing it)