PulseAudio
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.
Contents
- 1 Installation
- 2 Running
- 3 Equalizer
- 4 Back-end configuration
- 5 Desktop environments
- 6 Applications
- 7 Troubleshooting
- 7.1 No sound after install
- 7.2 No HDMI sound output after some time with the monitor turned off
- 7.3 Can't update configuration of sound device in pavucontrol
- 7.4 Simultaneous output to multiple sound cards / devices
- 7.5 Simultaneous output to multiple sinks on the same sound card not working
- 7.6 Disable Bluetooth support
- 7.7 Bluetooth headset replay problems
- 7.8 Automatically switch to Bluetooth or USB headset
- 7.9 Pulse overwrites ALSA settings
- 7.10 Prevent Pulse from restarting after being killed
- 7.11 Daemon startup failed
- 7.12 Glitches, skips or crackling
- 7.13 Setting the default fragment number and buffer size in PulseAudio
- 7.14 Laggy sound
- 7.15 Choppy, overdriven sound
- 7.16 Volume adjustment does not work properly
- 7.17 Per-application volumes change when the Master volume is adjusted
- 7.18 Volume gets louder every time a new application is started
- 7.19 No microphone on ThinkPad T400/T500/T420
- 7.20 No microphone input on Acer Aspire One
- 7.21 Sound output is only mono on M-Audio Audiophile 2496 sound card
- 7.22 Static noise in microphone recording
- 7.22.1 Determine sound cards in the system (1/5)
- 7.22.2 Determine sampling rate of the sound card (2/5)
- 7.22.3 Setting the sound card's sampling rate into PulseAudio configuration (3/5)
- 7.22.4 Restart PulseAudio to apply the new settings (4/5)
- 7.22.5 Finally check by recording and playing it back (5/5)
- 7.23 My Bluetooth device is paired but does not play any sound
- 7.24 Subwoofer stops working after end of every song
- 7.25 PulseAudio uses wrong microphone
- 7.26 Choppy sound with analog surround sound setup
- 7.27 No sound below a volume cutoff
- 7.28 Low volume for internal microphone
- 7.29 Clients alter master output volume (a.k.a. volume jumps to 100% after running application)
- 7.30 Realtime scheduling
- 7.31 No sound after resume from suspend
- 7.32 ALSA channels mute when headphones are plugged/unplugged improperly
- 7.33 pactl "invalid option" error with negative percentage arguments
- 7.34 Daemon already running
- 8 See also
Installation
- Required package: pulseaudio
- Optional GTK GUIs: paprefs and pavucontrol
- Optional volume control via mapped keyboard keys: pulseaudio-ctlAUR
- Optional console (CLI) mixers: ponymix and pamixer-gitAUR
- Optional web volume control: PaWebControl
- Optional system tray icon: pasystray-gitAUR
- Optional KDE plasma applet: kdemultimedia-kmix and kdeplasma-applets-veromixAUR
Running
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.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
Equalizer
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
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
ALSA
- Recommended package: pulseaudio-alsa
- Optional packages: lib32-libpulse and lib32-alsa-plugins
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
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.
OSS
There are multiple ways of making OSS-only programs play to PulseAudio:
ossp
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:
/usr/local/bin/OSSProgram
#!/bin/sh exec padsp /usr/bin/OSSprogram "$@"
Make sure /usr/local/bin
comes before /usr/bin
in your PATH.
GStreamer
To make GStreamer use PulseAudio, you need to install gst-plugins-good or gstreamer0.10-good-plugins.
OpenAL
OpenAL Soft should use PulseAudio by default, but can be explicitly configured to do so:/etc/openal/alsoft.conf
drivers=pulse,alsa
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.
ESD
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
/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:
~/.xinitrc
/usr/bin/start-pulseaudio-x11
GNOME
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
.
Xfce
Applications running under Xfce can take advantage of PulseAudio. To manage PulseAudio settings, you can use pavucontrol.
Applications
Audacious
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
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
Skype (x86_64 only)
Install lib32-libpulse, otherwise the following error will occur when trying to initiate a call: "Problem with Audio Playback".
Troubleshooting
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
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.
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:
.asoundrc
# </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 detection' in /etc/pulse/default.pa
might help.
load-module module-stream-restore restore_device=false
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.
/usr/lib/udev/rules.d/90-pulseaudio-Xonar-STX.rules
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.
asus-xonar-essence-stx.conf
also includes all code/mappings from default.conf
./usr/share/pulseaudio/alsa-mixer/profile-sets/asus-xonar-essence-stx.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
):
~/.config/pulse/default.pa
### Automatically load driver modules for Bluetooth hardware #.ifexists module-bluetooth-policy.so #load-module module-bluetooth-policy #.endif #.ifexists module-bluetooth-discover.so #load-module module-bluetooth-discover #.endif
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
):
~/.config/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:
/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 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 done 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.
~/.config/pulse/client.conf
# 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:
/etc/sysctl.d/99-sysctl.conf
# 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
:
/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.
Please report any such cards to PulseAudio Broken Sound Driver page
Setting the default fragment number and buffer size in PulseAudio
Finding out your audio device parameters (1/4)
To find your sound card buffering settings:
$ echo autospawn = no >> ~/.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' ~/.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)
/etc/pulse/daemon.conf
; 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
:
/etc/pulse/daemon.conf
; 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:
/etc/pulse/daemon.conf
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:
/etc/pulse/daemon.conf
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.
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:
/etc/pulse/daemon.conf
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.
~/.pulse/default.pa
which can also cause maximized volume trouble. Comment that out as needed.No microphone on ThinkPad T400/T500/T420
Run:
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:
/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 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:
/etc/bluetooth/audio.conf
[General] Enable=Socket
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
:
/etc/pulse/daemon.conf
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
:
/etc/pulse/daemon.conf
enable-lfe-remixing = yes
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
:
/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:
/etc/pulse/default.pa
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:
/etc/pulse/daemon.conf
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
:
[Unit] Description=Fix PulseAudio after resume from suspend After=suspend.target [Service] User=%I Type=oneshot Environment="XDG_RUNTIME_DIR=/run/user/%U" ExecStart=/usr/bin/pasuspender /bin/true [Install] WantedBy=suspend.target
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.
See also
- http://www.alsa-project.org/main/index.php/Asoundrc - ALSA wiki on .asoundrc
- http://www.pulseaudio.org/ - PulseAudio official site
- http://www.pulseaudio.org/wiki/FAQ - PulseAudio FAQ