Difference between revisions of "PulseAudio"

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=== My Bluetooth device is paired but does not play any sound ===
=== My Bluetooth device is paired but does not play any sound ===
[[Bluetooth#My_device_is_paired_but_no_sound_is_played_from_it|See the article in Bluetooth section]]
[[Bluetooth#My_device_is_paired_but_no_sound_is_played_from_it|See the article in Bluetooth section]]
Starting from PulseAudio 2.99 and bluez 4.101 you should '''avoid''' using Socket interface. Do NOT add
to your /etc/bluetooth/audio.conf.
If you face problems with A2DP and PA 2.99 make sure you have SBC library:
pacman -S sbc
=== Subwoofer stops working after end of every song  ===
=== Subwoofer stops working after end of every song  ===

Revision as of 07:51, 1 December 2012

PulseAudio is the default sound server that serves as a proxy to sound applications using existing kernel sound components like ALSA or OSS. Since ALSA is included in Arch Linux by default so the most common deployment scenarios include PulseAudio with ALSA.

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Note: Pulseaudio requires D-Bus to function.
Note: Most X11 environments start pulseaudio automatically with the X11 session.

In the unlikely event that pulseaudio is not automatically called upon entering X, it can can be started with:

$ pulseaudio --start

PulseAudio can be stopped with:

$ pulseaudio --kill


Newer pulseaudio versions have an intergrated 10-band equalizer system. In order to use the equalizer do the following:

Load equalizer sink module

$ pactl load-module module-equalizer-sink

Install and run the gui frontend

# pacman -S --needed python2-pyqt
$ qpaeq
Note: If qpaeq has no effect, install pavucontrol and change "ALSA Playback on" to "FFT based equalizer on ..." while the media player is running.

Load equalizer module on every boot

Edit the file /etc/pulse/default.pa with your favorite editor and append the following lines:

### Load the integrated pulseaudio equalizer module
load-module module-equalizer-sink

Backend Configuration

Tango-view-refresh-red.pngThis article or section is out of date.Tango-view-refresh-red.png

Reason: Arch has moved to systemd and rc.conf is now deprecated. (Discuss in Talk:PulseAudio#)


Note: Optional PKGs are needed only if running x86_64 and wanting to have sound for 32 bit programs (like Wine).

For the applications that do not support PulseAudio and support ALSA it is recommended to install the PulseAudio plugin for ALSA. This package also contains the necessary /etc/asound.conf for configuring ALSA to use PulseAudio.

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 in the MODULES array in /etc/rc.conf. If it is currently loaded (lsmod|grep oss), disable it by executing:

# rmmod snd_pcm_oss


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


Start ossp with:

rc.d start osspd

Afterwards, add it to DAEMONS in rc.conf.

padsp wrapper (part of PulseAudio)

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

$ padsp OSSprogram

A few examples:

$ padsp aumix
$ padsp sox foo.wav -t ossdsp /dev/dsp

One can also rename the OSSprogram-bin binary and replace it with a script like this:

if test -x /usr/bin/padsp; then
    exec /usr/bin/padsp /usr/bin/OSSprogram-bin "$@"
    exec /usr/bin/OSSprogram "$@"


To make GStreamer use PulseAudio, you need to install gstreamer0.10-good-plugins, execute gstreamer-properties (part of gnome-media package) and select PulseAudio Sound Server in both Audio Input and Output. Alternatively, this can be done by setting the gconf variables /system/gstreamer/0.10/default/audiosink to pulsesink and /system/gstreamer/0.10/default/audiosrc to pulsesrc:

 $ 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 applications (like Rhythmbox) ignore the audiosink property, but rely instead on musicaudiosink, which cannot be configured using gstreamer-properties but needs to be manually set using gconf-editor or the gconftool-2:

 $ gconftool-2 -t string --set /system/gstreamer/0.10/default/musicaudiosink pulsesink


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



Edit the libao configuration file:



PulseAudio is a drop-in replacement for the enlightened sound daemon (ESD). While PulseAudio is running, ESD clients should be able to output to it without configuration.

Desktop Environments

General X11

Note: As mentioned previously, PulseAudio is very likely launched automatically via either /etc/X11/xinit/xinitrc.d/pulseaudio or the files in /etc/xdg/autostart/ if users have some DE installed.

Check to see if PulseAudio is running:

$ ps aux | grep pulse
facade   1794  0.0  0.0 360464  6532 ?        S<l  15:33   0:00 /usr/bin/pulseaudio --start
facade   1827  0.0  0.0  68888  2608 ?        S    15:33   0:00 /usr/lib/pulse/gconf-helper

If Pulseaudio is not running and users are using X, the following will start PulseAudio with the needed the X11 plugins manually:

$ start-pulseaudio-x11

If you are not running Gnome, KDE or XFCE and your ~/.xinitrc does not source the scripts in /etc/X11/xinit/xinitrc.d (such as is done in the example file /etc/skel/.xinitrc) then you can launch PulseAudio on boot by adding the following line to ~/.xinitrc:



As of GNOME 3, GNOME fully integrates with PulseAudio and no extra configuration is needed.


PulseAudio is not a drop-in replacement for aRts. Users of KDE 3 cannot use PulseAudio. However note, recent versions of PulseAudio may have eliminated the prohibition:

See: http://www.pulseaudio.org/wiki/PerfectSetup KDE 3 uses the artsd sound server by default. However, artsd itself can be configured to use an Esound backend. Edit kcmartsrc (either in /etc/kde or /usr/share/config for global configuration or .kde/share/config to configure only one user) like this:

Arguments=\s-F 10 -S 4096 -a esd -n -s 1 -m artsmessage -c drkonqi -l 3 -f

KDE Plasma Workspaces and Qt4

PulseAudio, it will be used by KDE/Qt4 applications. For more information see the KDE page in the PulseAudio wiki.

PulseAudio support has been merged into KMix, the default KDE sound mixer.

One useful tidbit from that page is to add load-module module-device-manager to /etc/pulse/default.pa.

Additionally, the kdeplasma-applets-veromixAUR is available in the AUR as a KDE alternative to KMix or pavucontrol.


Applications running under Xfce can take advantage of PulseAudio. To manage PulseAudio settings you can use pavucontrol.



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

Java/OpenJDK 6

Create a wrapper for the java executable using padsp as seen on the Java sound with Pulseaudio page.

Music Player Daemon (MPD)

configure MPD to use PulseAudio.


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:


Skype (x86_64 only)

Install lib32-libpulse, otherwise the following error will occur when trying to initiate a call: "Problem with Audio Playback".


No sound after install

Muted audio device

If one experiences no audio output via any means while using ALSA, attempt to unmute the sound card. To do this, launch alsamixer and make sure each column has a green 00 under it (this can be toggled by pressing 'm')

$ alsamixer -c 0

Bad configuration files

If after starting pulseaudio, the system outputs no sound, it may be necessary to delete the contents of ~/.pulse. Pulseaudio will automatically create new configuration files on its next start.

Flash Content

Since Adobe Flash does not directly support PulseAudio the recommended way is to configure ALSA to use the virtual PulseAudio soundcard.

Alternatively you may try out libflashsupport-pulseAUR from the AUR.

Note: This may invariably crash the flash plugin.

No cards

If PulseAudio starts, run pacmd list. If no cards are reported, make sure that the ALSA devices are not in use:

$ fuser -v /dev/snd/*
$ fuser -v /dev/dsp

Make sure any applications using the pcm or dsp files are shut down before restarting PulseAudio.

The only device shown is "dummy output"

This may be caused by different reasons, one of them being the .asoundrc file in $HOME is taking precedence over the systemwide /etc/asound.conf.

The user file is modified also by the tool asoundconf or by its graphical variant asoundconf-gtk (the latter is named "Default sound card" in the menu) as soon as it runs. Prevent the effects of .asoundrc altogether by commenting the last line like this:



It may be that another output device set as preferred in phonon. Make sure that every setting reflects the preferred output device at the top, and check the playback streams tab in kmix to make sure that applications are using the device for output.

Bluetooth headset replay problems

Some user report huge delays or even no sound when the bluetooth connection does not send any data. This is due to an idle-suspend-module that puts the related sinks/sources automatically into suspend. As this can cause problems with headset, the responsible module can be deactivated.

1. cp /etc/pulse/default.pa ~/.pulse/default.pa
2. comment out the "load-module module-suspend-on-idle" line in ~/.pulse/default.pa
3. pulseaudio -k && pulseaudio --start

More information

Automatically switch to Bluetooth or USB headset

Add the following to your /etc/pulse/default.pa:

# automatically switch to newly-connected devices
load-module module-switch-on-connect

Pulse overwrites ALSA settings

Pulseaudio usually overwrites the ALSA settings- for example set with alsamixer- at start up, even when the alsa daemon is loaded. Since there seems to be no other way to restrict this behaviour, a workaround is to restore the alsa settings again after pulseaudio had started. Add the following command to .xinitrc .bash_login or any other autostart file:

restore_alsa() {
 while [ -z "`pidof pulseaudio`" ]; do
  sleep 0.5
 alsactl -f /var/lib/alsa/asound.state restore 
restore_alsa &

Daemon startup failed

Try resetting PulseAudio. To do that:

$ pulseaudio --kill
$ killall pulseaudio
$ killall -9 pulseaudio
$ rm -rf ~/.pulse*
$ rm -rf /tmp/pulse*

Afterwards, start PulseAudio again.


If one cannot launch the PulseAudio Device Chooser, first (re)start the Avahi daemon as follows:

$ rc.d restart avahi-daemon

Glitches, skips or crackling

The newer implementation of PulseAudio sound server uses a timer-based audio scheduling instead of the traditional interrupt-driven approach.

Timer-based scheduling may expose issues in some ALSA drivers. On the other hand, other drivers might be glitchy without it on, so check to see what works on your system.

To turn timer-based scheduling off, replace the line:

load-module module-udev-detect 

in /etc/pulse/default.pa by:

load-module module-udev-detect tsched=0

Then restart the PulseAudio server.

Do the reverse to enable timer-based scheduling, if not already enabled by default.

Please report any such cards to PulseAudio Broken Sound Driver page

Setting the default fragment number and buffer size in Pulseaudio

1. Finding out your audio device parameters

Run the following Bash commands to find your sound card buffering settings:

echo autospawn = no >> ~/.pulse/client.conf
killall pulseaudio
LANG=C timeout --foreground -k 10 -s kill 10 pulseaudio -vvvv 2>&1 | grep device.buffering -B 10
sed -i '$d' ~/.pulse/client.conf

For each sound card detected by Pulseaudio, you will see output similar to this:

I: [pulseaudio] source.c:     alsa.long_card_name = "HDA Intel at 0xfa200000 irq 46"
I: [pulseaudio] source.c:     alsa.driver_name = "snd_hda_intel"
I: [pulseaudio] source.c:     device.bus_path = "pci-0000:00:1b.0"
I: [pulseaudio] source.c:     sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
I: [pulseaudio] source.c:     device.bus = "pci"
I: [pulseaudio] source.c:     device.vendor.id = "8086"
I: [pulseaudio] source.c:     device.vendor.name = "Intel Corporation"
I: [pulseaudio] source.c:     device.product.name = "82801I (ICH9 Family) HD Audio Controller"
I: [pulseaudio] source.c:     device.form_factor = "internal"
I: [pulseaudio] source.c:     device.string = "front:0"
I: [pulseaudio] source.c:     device.buffering.buffer_size = "768000"
I: [pulseaudio] source.c:     device.buffering.fragment_size = "384000"

Take note the buffer_size and fragment_size values for the relevant sound card.

2. Calculate your fragment size in msecs and number of fragments

Pulseaudio's default sampling rate and bit depth are set to 44100Hz @ 16 bits.

With this configuration, the bit rate we need is 44100*16 = 705600 bits per second. That's 1411200 bps for stereo.

Let's take a look at the parameters we've found in the previous step:

device.buffering.buffer_size = "768000" => 768000/1411200 = 0.544217687075s = 544 msecs
device.buffering.fragment_size = "384000" => 384000/1411200 = 0.272108843537s = 272 msecs

3.Modify Pulseaudio's configuration file

Edit the configuration file located at /etc/pulse/daemon.conf using the editor of your choice.

For example:

sudo vi /etc/pulse/daemon.conf

Locate & uncomment (remove leading semicolons) these lines:

; default-fragments = X
; default-fragment-size-msec = Y

In the previous step, we calculated the fragment size parameter. The number of fragments is simply buffer_size/fragment_size, which in this case (544/272) is 2.

Edit the lines to use your calculated settings:

default-fragment-size-msec = 272
default-fragments = 2

Save the file.

4.Restart the Pulseaudio daemon

pulseaudio -k
pulseaudio --start

Source: kwevej @ Linux Mint Forums

Laggy sound

This issue is due to incorrect buffer sizes. Edit /etc/pulse/daemon.conf

Either disable any modifications (if any) to these entries, or, if issue still exists, uncomment and change them in the following way:

default-fragments = 8
default-fragment-size-msec = 5

Choppy, overdriven sound

Choppy sound in pulsaudio can result from wrong settings for the sample rate in /etc/pulse/daemon.conf. Try changing the line

; default-sample-rate = 44100


default-sample-rate = 48000

and restart the PulseAudio server.

If one experiences choppy sound in applications using openAL, change the sample rate in /etc/openal/alsoft.conf:

frequency = 48000

Setting the PCM volume above 0dB can cause clipping of the audio signal. Running alsamixer -c0 will allow you to see if this is the problem and if so fix it.

Volume adjustment does not work properly

Check: /usr/share/pulseaudio/alsa-mixer/paths/analog-output.conf.common

If the volume does not appear to increment/decrement properly using alsamixer or amixer, it may be due to pulseaudio having a larger number of increments (65537 to be exact). Try using larger values when changing volume (e.g. amixer set Master 655+).

Volume gets louder every time a new application is started

Per default, it seems as if changing the volume in an application sets the global system volume to that level instead of only affecting the respective application. Applications setting their volume on startup will therefore cause the system volume to "jump".

Fix this by uncommenting

flat-volumes = no



and then restarting PulseAudio by executing

pulseaudio --kill && pulseaudio --start

When Pulse comes back after a few seconds, applications will not alter the global system volume anymore but have their own volume level again.

Note: A previously installed and removed pulseaudio-equalizer may leave behind remnants of the setup in $HOME/.pulse/default.pa which can also cause maximized volume trouble. Comment that out as needed.

No mic on ThinkPad T400/T500/T420


alsamixer -c 0

Maximize the volume of/unmute the "Internal Mic".

Once you see the device with

arecord -l

you might still need to adjust the settings. The microphone and the audio jack are duplexed. Set the configuration of the internal audio in pavucontrol to Analog Stereo Duplex.

No mic input on Acer Aspire One

Install pavucontrol, unlink the microphone channels and turn down the left one to 0. Reference: http://getsatisfaction.com/jolicloud/topics/deaf_internal_mic_on_acer_aspire_one#reply_2108048

Sound output is only mono on M-Audio Audiophile 2496 sound card

Add the following to /etc/pulseaudio/default.pa:

load-module module-alsa-sink sink_name=delta_out device=hw:M2496 format=s24le channels=10 channel_map=left,right,aux0,aux1,aux2,aux3,aux4,aux5,aux6,aux7
load-module module-alsa-source source_name=delta_in device=hw:M2496 format=s24le channels=12 channel_map=left,right,aux0,aux1,aux2,aux3,aux4,aux5,aux6,aux7,aux8,aux9
set-default-sink delta_out
set-default-source delta_in

Static Noise in Microphone Recording

If we are getting static noise in skype, gnome-sound-recorder, arecord, etc.'s recordings then the sound card samplerate is incorrect. That is why there is static noise in linux microphone recordings. To fix this We need to set sample-rate in /etc/pulse/daemon.conf for the sound hardware.

1. Determine soundcards in the system

This requires alsa-utils and related packages to be installed:

$  arecord --list-devices


   **** List of CAPTURE Hardware Devices ****
   card 0: Intel [HDA Intel], device 0: ALC888 Analog [ALC888 Analog]
     Subdevices: 1/1
     Subdevice #0: subdevice #0
   card 0: Intel [HDA Intel], device 2: ALC888 Analog [ALC888 Analog]
     Subdevices: 1/1
     Subdevice #0: subdevice #0

soundcard is hw:0,0

2. Determine sampling-rate of the sound card

arecord -f dat -r 60000 -D hw:0,0 -d 5 test.wav


  "Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 60000 Hz, Stereo
  Warning: rate is not accurate (requested = 60000Hz, got = 96000Hz)
  please, try the plug plugin

observe, the got = 96000Hz, this is the max sample-rate of our card.

3. Setting the soundcard's sampling rate into pulse audio configuration

the default sample-rate in pulseaudio is

grep "sample-rate" /etc/pulse/daemon.conf


   ; default-sample-rate = 44100

It is 44100 and is disabled. Let us set our sound card's settings into pulseaudio configuation file

su -c "sed 's/; default-sample-rate = 44100/default-sample-rate = 96000/g' -i /etc/pulse/daemon.conf"

Let us verify the changes to deamon.conf

grep "sample-rate" /etc/pulse/daemon.conf 


  default-sample-rate = 96000

and it is done.

4. Restart pulseaudio to apply the new settings

pulseaudio --kill
pulseaudio --start

5. Finally check by recording and playing it back

Let us record some voice using mic for say 10 seconds. Make sure the mic is not muted and all

arecord -f cd -d 10 test-mic.wav

After 10 seconds, let us play the recording...

aplay test-mic.wav

Now hopefully, there is no static noise in microphone recording anymore.

My Bluetooth device is paired but does not play any sound

See the article in Bluetooth section

Starting from PulseAudio 2.99 and bluez 4.101 you should avoid using Socket interface. Do NOT add


to your /etc/bluetooth/audio.conf. If you face problems with A2DP and PA 2.99 make sure you have SBC library:

pacman -S sbc 

Subwoofer stops working after end of every song

Known issue: https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/494099

To fix this, must edit: /etc/pulse/daemon.conf and enable enable-lfe-remixing :

enable-lfe-remixing = yes

Pulseaudio uses wrong microphone

If Pulseaudio uses the wrong microphone, and changing the Input Device with Pavucontrol did not help, take a look at alsamixer. It seems that Pavucontrol does not always set the input source correctly.

$ alsamixer

press F6 and choose your sound card, e.g. HDA Intel. Now press F5 to display all items. Try to find the item: Input Source. With the up/down arrow keys you are able to change the input source.
Now try if the correct microphone is used for recording.

Choppy Sound with Analog Surround Sound Setup

The low-frequency effects (LFE) channel is not remixed per default. To enable it the following needs to be set in /etc/pulse/daemon.conf :

enable-lfe-remixing = yes

External links