Difference between revisions of "PulseAudio"

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If you are unable to select 5/7.1 channel output for a working HDMI device, then turning off "stream device reading" in {{ic|/etc/pulse/default.pa}} might help.  
If you are unable to select 5/7.1 channel output for a working HDMI device, then turning off "stream device reading" in {{ic|/etc/pulse/default.pa}} might help.  
See [[Pulseaudio#Fallback device is not respected|Fallback device is not respected]]
See [[#Fallback device is not respected]].
==== KDE Plasma Workspaces ====
==== KDE Plasma Workspaces ====

Revision as of 05:46, 4 May 2014

PulseAudio is a sound server commonly used by desktop environments like GNOME or KDE. It 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, the most common deployment scenarios include PulseAudio with ALSA.




Warning: If you have per-user copies of configuration files (such as client.conf, daemon.conf or default.pa) in ~/.config/pulse/ or ~/.pulse/, make sure you keep them in sync with changes to the packaged files in /etc/pulse/. Otherwise, PulseAudio may refuse to start due to configuration errors.
Note: Most X11 environments start PulseAudio automatically with the X11 session.

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

$ pulseaudio --start

PulseAudio can be stopped with:

$ pulseaudio -k


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

Load equalizer sink and dbus-protocol module

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

Install and run the GUI front-end

Install python-pyqt4 and execute:

$ 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 and dbus 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 and D-Bus module
load-module module-equalizer-sink
load-module module-dbus-protocol

Back-end configuration


Note: Optional packages 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 being loaded at boot. If it is currently loaded (lsmod | grep oss), disable it by executing:

# rmmod snd_pcm_oss

ALSA/dmix without grabbing hardware device

Note: This section describes alternative configuration, which is generally not recommended.

You may want to use ALSA directly in most of your applications and to be able to use other applications, which constantly 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.
  • Edit /etc/pulse/default.pa.
Find and uncomment lines which load back-end drivers. Add device parameters as follows. Then find and comment lines which load autodetect modules.
load-module module-alsa-sink device=dmix
load-module module-alsa-source device=dsnoop
# load-module module-udev-detect
# load-module module-detect
  • Optional: If you use kdemultimedia-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.


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


Install ossp package and start osspd service.

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

You can also add a custom wrapper script like this:

exec padsp /usr/bin/OSSprogram "$@"

Make sure /usr/local/bin comes before /usr/bin in your PATH.


To make GStreamer use PulseAudio, you need to install gst-plugins-good or gstreamer0.10-good-plugins.


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



Edit the libao configuration file:

# /etc/libao.conf

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

Note: You could possibly also keep the libao standard of outputting to the alsa driver and its default device if you install pulseaudio-alsa since the ALSA default device then is PulseAudio.


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 with:



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

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.

If the phonon-gstreamer backend is used for Phonon, GStreamer should also be configured to use PulseAudio by installing gstreamer0.10-good-plugins.

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.

If KMix/Veromix fail to connect to PulseAudio at boot you may need to edit /etc/pulse/client.conf to include autospawn = yes instead of autospawn = no.


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. See also MPD/Tips and Tricks#MPD and 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
Note: alsamixer will not tell you which output device is set as the default. One possible cause of no sound after install is that PulseAudio detects the wrong output device as a default. Install pavucontrol and check if there is any output on the pavucontrol panel when playing a .wav file.

Auto-Mute Mode

Auto-Mute Mode may be enabled. It can be disabled using alsamixer.

See http://superuser.com/questions/431079/how-to-disable-auto-mute-mode for more.

Bad configuration files

After starting PulseAudio, if 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 sound card.

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

Note: This may invariably crash the Flash plugin.

If you find audio from flash is being laggy, you might like to have flash access ALSA directly. This fixes this problem in some cases. To do this, configure PulseAudio to use dmix.

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:

# </home/yourusername/.asoundrc.asoundconf>

Maybe some program is monopolizing the audio device:

# fuser -v /dev/snd/*
                     USER       PID  ACCESS COMMAND
/dev/snd/controlC0:  root        931 F....  timidity
                     bob        1195 F....  panel-6-mixer
/dev/snd/controlC1:  bob        1195 F....  panel-6-mixer
                     bob        1215 F....  pulseaudio
/dev/snd/pcmC0D0p:   root        931 F...m  timidity
/dev/snd/seq:        root        931 F....  timidity
/dev/snd/timer:      root        931 f....  timidity

That means timidity blocks PulseAudio from accessing the audio devices. Just killing timidity will make the sound work again.

Another reason is FluidSynth conflicting with PulseAudio as discussed in this thread. The solution is to remove FluidSynth:

# pacman -Rnsc fluidsynth

No HDMI 5/7.1 Selection for Device

If you are unable to select 5/7.1 channel output for a working HDMI device, then turning off "stream device reading" in /etc/pulse/default.pa might help.

See #Fallback device is not respected.

KDE Plasma Workspaces

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.

To see your default audio device, you can run:

pactl stat

To see available audio devices:

pactl list

To set your default audio device, use "pacmd" or add to /etc/pulse/default.pa:

set-default-sink alsa_output.analog-stereo

Failed to create sink input: sink is suspended

If you do not have any output sound and receive dozens of errors related to a suspended sink in your journalctl -b log, then backup first and then delete your user-specific pulse folders:

$ rm -r ~/.pulse ~/.pulse-cookie

No HDMI sound output after some time with the monitor turned off

The monitor is connected via HDMI/DisplayPort, and the audio jack is plugged in the headphone jack of the monitor, but PulseAudio insists that it is unplugged:

pactl list sinks
hdmi-output-0: HDMI / DisplayPort (priority: 5900, not available)

This leads to no sound coming from HDMI output. A workaround for this is to switch to another TTY and back again. This problem has been reported by ATI/Nvidia/Intel users.

Can't update configuration of sound device in pavucontrol

pavucontrol is a handy GUI utility for configuring PulseAudio. Under its 'Configuration' tab, you can select different profiles for each of your sound devices e.g. analogue stereo, digital output (IEC958), HDMI 5.1 Surround etc.

However, you may run into an instance where selecting a different profile for a card results in the pulse daemon crashing and auto restarting without the new selection "sticking". If this occurs, use the other useful GUI tool, paprefs, to check under the "Simultaneous Output" tab for a virtual simultaneous device. If this setting is active (checked), it will prevent you changing any card's profile in pavucontrol. Uncheck this setting, then adjust your profile in pavucontrol prior to re-enabling simultaneous output in paprefs.

Simultaneous output to multiple sound cards / devices

Simultaneous output to two different devices can be very useful. For example, being able to send audio to your A/V receiver via your graphics card's HDMI output, while also sending the same audio through the analogue output of your motherboard's built-in audio. This is much less hassle than it used to be (in this example, we are using GNOME desktop).

Using paprefs, simply select "Add virtual output device for simultaneous output on all local sound cards" from under the "Simultaneous Output" tab. Then, under GNOME's "sound settings", select the simultaneous output you have just created.

If this doesn't work, try adding the following to ~/.asoundrc:

pcm.dsp {
   type plug
   slave.pcm "dmix"

Simultaneous output to multiple sinks on the same sound card not working

This can be useful for users who have multiple sound sources and want to play them on different sinks/outputs. An example use-case for this would be if you play music and also voice chat and want to output music to speakers (in this case Digital S/PDIF) and voice to headphones. (Analog)

This is sometimes auto detected by PulseAudio but not always. If you know that your sound card can output to both Analog and S/PDIF at the same time and PulseAudio does not have this option in it's profiles in pavucontrol, or veromix then you probably need to create a configuration file for your sound card.

More in detail you need to create a profile-set for your specific sound card. This is done in two steps mostly.

  • Create udev rule to make PulseAudio choose your PulseAudio configuration file specific to the sound card.
  • Create the actual configuration.

Create a pulseadio udev rule.

Note: This is only an example for Asus Xonar Essence STX. Read udev to find out the correct values.
Note: Your configuration file should have lower number than the original PulseAudio rule to take effect.
ACTION=="change", SUBSYSTEM=="sound", KERNEL=="card*", \
ATTRS{subsystem_vendor}=="0x1043", ATTRS{subsystem_device}=="0x835c", ENV{PULSE_PROFILE_SET}="asus-xonar-essence-stx.conf" 

Now, create a configuration file. If you bother, you can start from scratch and make it saucy. However you can also use the default configuration file, rename it, and then add your profile there that you know works. Less pretty but also faster.

To enable multiple sinks for Asus Xonar Essence STX you need only to add this in.

Note: asus-xonar-essence-stx.conf also includes all code/mappings from default.conf.
[Profile analog-stereo+iec958-stereo]
description = Analog Stereo Duplex + Digital Stereo Output
input-mappings = analog-stereo
output-mappings = analog-stereo iec958-stereo
skip-probe = yes

This will auto-profile your Asus Xonar Essence STX with default profiles and add your own profile so you can have multiple sinks.

You need to create another profile in the configuration file if you want to have the same functionality with AC3 Digital 5.1 output.

See PulseAudio article about profiles

Disable Bluetooth support

If you do not use Bluetooth, you may experience the following error in your journal:

bluez5-util.c: GetManagedObjects() failed: org.freedesktop.DBus.Error.ServiceUnknown: The name org.bluez was not provided by any .service files

To disable Bluetooth support in PulseAudio, make sure that the following lines are commented out in the configuration file in use (~/.config/pulse/default.pa or /etc/pulse/default.pa):

### Automatically load driver modules for Bluetooth hardware
#.ifexists module-bluetooth-policy.so
#load-module module-bluetooth-policy

#.ifexists module-bluetooth-discover.so
#load-module module-bluetooth-discover

Bluetooth headset replay problems

Some user reports huge delays or even no sound when the Bluetooth connection does not send any data. This is due to the module-suspend-on-idle module, which automatically suspends sinks/sources on idle. As this can cause problems with headset, the responsible module can be deactivated.

To disable loading of the module-suspend-on-idle module, comment out the following line in the configuration file in use (~/.config/pulse/default.pa or /etc/pulse/default.pa):

### Automatically suspend sinks/sources that become idle for too long
#load-module module-suspend-on-idle

Finally restart PulseAudio to apply the changes.

Automatically switch to Bluetooth or USB headset

Add the following:

# 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 has started. Add the following command to .xinitrc or .bash_profile 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 &

Prevent Pulse from restarting after being killed

Sometimes you may wish to temporarily disable Pulse. In order to do so you will have to prevent Pulse from restarting after being killed.

# Disable autospawning the PulseAudio daemon
autospawn = no

Daemon startup failed

Try resetting PulseAudio:

$ rm -rf /tmp/pulse* ~/.pulse*
$ pulseaudio -k
$ pulseaudio --start

If there is no server running but PulseAudio fails to start with an error message "User-configured server at ... refusing to start/autospawn", the issue may be with PulseAudio settings from a previous login. Check to see if there are any stale properties attached to the X11 root window with pax11publish -d, and if there are, remove them with pax11publish -r before trying to start the server. This manual cleanup is always required when using LXDM because it does not restart the X server on logout; see LXDM#PulseAudio.

inotify issue

If the previous fix doesn't work, see if you get an error like this:

$ pulseaudio -vvvv
E: [pulseaudio] module-udev-detect.c: You apparently ran out of inotify watches, probably because Tracker/Beagle took them all away. I wished people would do their homework first and fix inotify before using it for watching whole directory trees which is something the current inotify is certainly not useful for. Please make sure to drop the Tracker/Beagle guys a line complaining about their broken use of inotify.

In which case you have run out of inotify watches.

This can quickly be resolved by:

# echo 100000 > /proc/sys/fs/inotify/max_user_watches

To have it permanently changed, use:

# Increase inotify max watchs per user
fs.inotify.max_user_watches = 100000

Glitches, skips or crackling

The newer implementation of the PulseAudio sound server uses 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 add tsched=0 in /etc/pulse/default.pa:

load-module module-udev-detect tsched=0

Then restart the PulseAudio server:

$ pulseaudio -k
$ pulseaudio --start

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

Using Intel's IOMMU may also lead to these problems. If you are using IOMMU and experience glitches and/or skips add intel_iommu=igfx_off to your kernel command line.

Please report any such cards to PulseAudio Broken Sound Driver page

Setting the default fragment number and buffer size in PulseAudio

More Information

Finding out your audio device parameters (1/4)

To find your sound card buffering settings:

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

Calculate your fragment size in msecs and number of fragments (2/4)

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 have 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

Modify PulseAudio's configuration file (3/4)

; 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:

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

Restart the PulseAudio daemon (4/4)

$ pulseaudio -k
$ pulseaudio --start

For more information, see: Linux Mint topic

Laggy sound

This issue is due to incorrect buffer sizes.

Either disable any modifications (if any) to these entries, or, if issue still exists, uncomment:

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

Choppy, overdriven sound

Choppy sound in PulseAudio can result from wrong settings for the sample rate. Try:

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 0 dB 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. Note that ALSA may not correctly export the dB information to PulseAudio. Try:

load-module module-udev-detect ignore_dB=1

and restart the PulseAudio server. See also #No sound below a volume cutoff.

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+).

Per-application volumes change when the Master volume is adjusted

This is because PulseAudio uses flat volumes by default, instead of relative volumes, relative to an absolute master volume. If this is found to be inconvenient, asinine, or otherwise undesireable, relative volumes can be enabled by disabling flat volumes in the PulseAudio daemon's configuration file:

flat-volumes = no

and then restarting PulseAudio by executing

$ pulseaudio -k
$ pulseaudio --start

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 disabling flat volumes, as demonstrated in the previous section. 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 ~/.pulse/default.pa which can also cause maximized volume trouble. Comment that out as needed.

No microphone on ThinkPad T400/T500/T420


alsamixer -c 0

Unmute and maximize the volume of 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 microphone 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:

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 sample rate is incorrect. That is why there is static noise in Linux microphone recordings. To fix this, we need to set the sampling rate in /etc/pulse/daemon.conf for the sound hardware.

Determine sound cards in the system (1/5)

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

Sound card is hw:0,0.

Determine sampling rate of the sound card (2/5)

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 maximum sampling rate of our card.

Setting the sound card's sampling rate into PulseAudio configuration (3/5)

The default sampling rate in PulseAudio:

$ grep "default-sample-rate" /etc/pulse/daemon.conf
; default-sample-rate = 44100

44100 is disabled and needs to be changed to 96000:

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

Restart PulseAudio to apply the new settings (4/5)

$ pulseaudio -k
$ pulseaudio --start

Finally check by recording and playing it back (5/5)

Let us record some voice using a microphone for, say, 10 seconds. Make sure the microphone 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 use:


If you face problems with A2DP and PA 2.99 make sure you have sbc library.

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

Unable to select surround configuration other than "Surround 4.0"

If you're unable to set 5.1 surround output in pavucontrol because it only shows "Analog Surround 4.0 Output", open the ALSA mixer and change the output configuration there to 6 channels. Then restart pulseaudio, and pavucontrol will list many more options.

No sound below a volume cutoff

Known issue (won't fix): https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/223133

If sound does not play when PulseAudio's volume is set below a certain level, try setting ignore_dB=1 in /etc/pulse/default.pa:

load-module module-udev-detect ignore_dB=1

However, be aware that it may cause another bug preventing PulseAudio to unmute speakers when headphones or other audio devices are unplugged.

Low volume for internal microphone

If you experience low volume on internal notebook microphone, try setting:

set-source-volume 1 300000

Clients alter master output volume (a.k.a. volume jumps to 100% after running application)

If changing the volume in specific applications or simply running an application changes the master output volume this is likely due to flat volumes mode of pulseaudio. Before disabling it, KDE users should try lowering their system notifications volume in System Settings -> Application and System Notifications -> Manage Notifications under the Player Settings tab to something reasonable. Changing the Event Sounds volume in KMix or another volume mixer application will not help here. This should make the flat-volumes mode work out as intended, if it does not work, some other application is likely requesting 100% volume when its playing something. If all else fails, you can try to disable flat-volumes:

flat-volumes = no

Then restart PulseAudio daemon:

# pulseaudio -k
# pulseaudio --start

Realtime scheduling

If rtkit does not work, you can manually set up your system to run PulseAudio with real-time scheduling, which can help performance. To do this, add the following lines to /etc/security/limits.conf:

@pulse-rt - rtprio 9
@pulse-rt - nice -11

Afterwards, you need to add your user to the pulse-rt group:

# gpasswd -a <user> pulse-rt

No sound after resume from suspend

If audio generally works, but stops after resume from suspend, try "reloading" PulseAudio by executing:

$ /usr/bin/pasuspender /bin/true

This is better than completely killing and restarting it (pulseaudio -k followed by pulseaudio --start), because it doesn't break already running applications.

If the above fixes your problem, you may wish to automate it, by creating a systemd service file.

1. Create the template service file in /etc/systemd/system/resume-fix-pulseaudio@.service:

Description=Fix PulseAudio after resume from suspend

ExecStart=/usr/bin/pasuspender /bin/true


2. Enable it for your user account

# systemctl enable resume-fix-pulseaudio@YOUR_USERNAME_HERE.service

3. Reload systemd

# systemctl --system daemon-reload

ALSA channels mute when headphones are plugged/unplugged improperly

If when you unplug your headphones or plug them in the audio remains muted in alsamixer on the wrong channel due to it being set to 0%, you may be able to fix it by opening /etc/pulse/default.pa and commenting out the line:

load-module module-switch-on-port-available

pactl "invalid option" error with negative percentage arguments

pactl commands that take negative percentage arguments will fail with an 'invalid option' error. Use the standard shell '--' pseudo argument to disable argument parsing before the negative argument. e.g. pactl set-sink-volume 1 -- -5%.

Daemon already running

On some systems, PulseAudio may be started multiple times. journalctl will report:

[pulseaudio] pid.c: Daemon already running.

Make sure to use only one method of autostarting applications. pulseaudio includes these files:

  • /etc/X11/xinit/xinitrc.d/pulseaudio
  • /etc/xdg/autostart/pulseaudio.desktop
  • /etc/xdg/autostart/pulseaudio-kde.desktop

Also check user autostart files and directories, such as xinitrc, ~/.config/autostart/ etc.

Fallback device is not respected

PulseAudio does not have a true default device. Instead it uses a "fallback", which only applies to new sound streams. This means previously run applications are not affected by the newly set fallback device.

Only gnome-media and mate-media-pulseaudio handle this gracefully. Alternatively:

1. Move the old streams in pavucontrol manually to the new sound card.

2. Stop Pulse, erase the "stream-volumes" in ~/.pulse and restart Pulse. This also resets application volumes.

3. Disable stream device reading. This may be not wanted when using different soundcards with different applications.

load-module module-stream-restore restore_device=false

See also