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PulseAudio is a general purpose sound server. For a list of features, see Wikipedia:PulseAudio#Features.

Tip: If you want to play sound from multiple applications, ALSA already provides a basic software mixer named "dmix" which is enabled by default.



To install PulseAudio:

# pacman -S pulseaudio

Optionally you can install some GTK front-ends for PulseAudio:

# pacman -S paprefs pavucontrol


If the D-Bus system daemon is not already running, start it:

# /etc/rc.d/dbus start

PulseAudio can be started with:

$ pulseaudio --start

Or if you use X11:

$ start-pulseaudio-x11

If you use KDE 4:

$ start-pulseaudio-x11; start-pulseaudio-kde

PulseAudio can be stopped with:

$ pulseaudio --kill

Note that in the case of most X11 environments, PulseAudio will be automatically started together with the X11 session. See the section on Desktop Environments for details.

Backend Configuration


For the applications that do not support PulseAudio and support ALSA it is recommended to install the PulseAudio plugin for ALSA. The simplest way of doing this is installing the pulseaudio-alsa package:

# pacman -S pulseaudio-alsa

This package also contains the necessary /etc/asound.conf for configuring ALSA to use PulseAudio.

If you are on Arch x86_64 and want to have sound for 32 bit programs (like Wine), make sure to install lib32-libpulse and lib32-alsa-plugins as well.

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's currently loaded you can disable it by executing:

# rmmod snd_pcm_oss


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


This is the simplest method.

Install ossp and start oospd with:

/etc/rc.d/osspd start

Afterwards, add it to DAEMONS in rc.conf.

padsp wrapper

If you have a program that uses OSS you can make it work with PulseAudio by starting it with padsp:

$ padsp OSSprogram

A few examples:

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

If you prefer you can rename the program OSSprogram-bin 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, 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 can't 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:



The current binary of PortAudio in the community repository does not support PulseAudio and non-mmap audio devices. This can be remedied by building PortAudio from ABS and applying a patchTemplate:Linkrot to the sources. There is an AUR package for portaudio with pulse support.


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

Start Pulseaudio after the X session using:

$ start-pulseaudio-x11

This will start PulseAudio and load the X11 plugins.

Note that PulseAudio may already launch automatically. This is handled by either /etc/X11/xinit/xinitrc.d/pulseaudio or the files in /etc/xdg/autostart/.

X11 bell

To make PulseAudio play a sample when an X11 bell event happens (e.g. to make your terminal go 'Ping!' instead of 'Beep!'), add the following to /etc/pulse/default.pa:

# Prevent pulseaudio --start from failing when following commands fail (eg. when X is not active)
load-sample-lazy x11-bell /usr/share/sounds/freedesktop/stereo/dialog-error.oga
load-module module-x11-bell sample=x11-bell

You can also use another sample. dialog-error.oga is provided by sound-theme-freedesktop.


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


PulseAudio is not a drop-in replacement for aRts. If you use KDE 3, it is not currently possible to use PulseAudio.

KDE 4 and Qt4

If you've installed PulseAudio, it will be used by KDE4/Qt4 applications. For more information see the KDE pages in the PulseAudio wiki.

Additionally, a minimal KDE alternative to pavucontrol, the veromix plasmoid is available in the AUR.


Applications running under XFCE4 can take advantage of Pulseaudio. To manage Pulseaudio settings you can use pavucontrol.



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

Flashplugin (x86_64 only)

If you are using Adobe Flash Player browser plugin from the multilib repository you need to install lib32-alsa-plugins and lib32-libcanberra-pulse, otherwise you will not be able to use other audio applications while using flashplugin.

# pacman -S lib32-alsa-plugins lib32-libcanberra-pulse


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

Music Player Daemon (MPD)

You will need to configure MPD to use PulseAudio.

On a headless box, run PulseAudio as the mpd user. On a desktop, running mpd as yourself and not using the mpd user is preferred.


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 from multilib repository, otherwise you will get the Error "Problem with Audio Playback" when trying to initiate a call.

VLC media player

VLC supports PulseAudio through native plugin. In order to use it, in VLC you should open Tools -> Preferences (or Ctrl-P) and on Audio tab change Output Type to Pulseaudio audio output.

Also consider increasing Caching to 1000-1500 ms.: (This could solve some freezings/lags) change Show options to All and go to Input / Codecs -> Access Modules -> File

volume control

Adapt the variable 'device' to your sink-device listed in

pactl list

and you can change the volume using this script.

case "$1" in
	"up")    # increase volume by 1000
		pacmd dump | awk --non-decimal-data '$1~/set-sink-volume/{if ($2~/'${device}'/) {if ($3+1000 > 65535) {system ("pactl "$1" '${device}' "65535)} else {system ("pactl "$1" '${device}' "$3+1000)}}}'
	"down")  # decrease volume by 1000
		pacmd dump | awk --non-decimal-data '$1~/set-sink-volume/{if ($2~/'${device}'/) {if ($3-1000 < 0) {system ("pactl "$1" '${device}' "0)} else {system ("pactl "$1" '${device}' "$3-1000)}}}'
	"mute")  # toggle mute
		pacmd dump|awk --non-decimal-data '$1~/set-sink-mute/{if ($2~/'${device}'/) {system ("pactl "$1" '${device}' "($3=="yes"?"no":"yes"))}}'

There is also a amixer like commandline mixer avaible in AUR called pamixer

Alternative configurations

Surround sound systems

Many people have a surround card, but have speakers for just two channels, so PulseAudio cannot really default to a surround setup. To enable all the channels, edit /etc/pulse/daemon.conf: uncomment the default-sample-channels line (i.e. remove the semicolon from the beginning of the line) and set the value to 6 if you have a 5.1 setup, or 8 if you have 7.1 setup etc.

# Default
# For 5.1
# For 7.1

After doing the edit, you will need to restart Pulseaudio.

Splitting front/rear

You may want connect speakers to front analog output and headphones to rear output. It would be usefull to split front/rear to separate sinks. Add to default.pa:

 load-module module-remap-sink sink_name=speakers remix=no master=alsa_output.pci-0000_05_00.0.analog-surround-40 channels=2 master_channel_map=front-left,front-right channel_map=front-left,front-right
 load-module module-remap-sink sink_name=headphones remix=no master=alsa_output.pci-0000_05_00.0.analog-surround-40 channels=2 master_channel_map=rear-left,rear-right   channel_map=front-left,front-right

(replace alsa_output.pci-0000_05_00.0.analog-surround-40 to your sound card name that you got from 'pacmd list-sinks')

After that you can switch player between speakers and headphones.

Advanced ALSA Configuration

In order for ALSA to use PulseAudio it needs a special /etc/asound.conf (system wide settings) (recommended) or ~/.asoundrc (settings on a per user basis):

pcm.pulse {
    type pulse
ctl.pulse {
    type pulse
pcm.!default {
    type pulse
ctl.!default {
    type pulse

If you omit the last two groups, Pulseaudio will not be used by default. You will then need to change the ALSA device to "pulse" in the applications that you use to make it work.

ALSA Monitor source

To be able to record from a monitor source (a.k.a. "What-U-Hear", "Stereo Mix"), use pactl list to find out the name of the source in Pulseaudio (e.g. alsa_output.pci-0000_00_1b.0.analog-stereo.monitor). Then add lines like the following to /etc/asound.conf or ~/.asoundrc:

pcm.pulse_monitor {
  type pulse
  device alsa_output.pci-0000_00_1b.0.analog-stereo.monitor

ctl.pulse_monitor {
  type pulse
  device alsa_output.pci-0000_00_1b.0.analog-stereo.monitor

Now you can select pulse_monitor as a recording source.

Note: You can also simply install the package 'pulseaudio-alsa'

PulseAudio over network

One of PulseAudio's magnificent features is the possibility to stream audio from clients over TCP to the server running the PulseAudio daemon, allowing sound to be streamed through your LAN.

To accomplish this, one needs to enable module-native-protocol-tcp, and copy the pulse-cookie to the clients.

TCP support (networked sound)

To enable the TCP module, add this to (or uncomment, if already there) /etc/pulse/default.pa:

load-module module-native-protocol-tcp

Note: If you are having trouble connecting, use (on server)

pacmd>> list-modules

(you can even load modules from here!)

Zeroconf (Avahi) publishing

For the remote Pulseaudio server to appear in the PulseAudio Device Chooser (padevchooser) you need to load the appropriate zeroconf modules as well as to enable Avahi. On both machines add avahi-daemon to the DAEMONS in rc.conf. On the server, add load-module module-zeroconf-publish to /etc/pulse/default.pa, on the client, add load-module module-zeroconf-discover to /etc/pulse/default.pa. You can then redirect any stream or complete audio output to the remote pulseaudio server by selecting the appropriate sink.

Switching the PulseAudio server used by local X clients

To switch between servers on the client from within X, the pax11publish command can be used. For example, to switch from the default server to the server at hostname foo:

$ pax11publish -e -S foo

Or to switch back to the default:

$ pax11publish -e -r

Note that for the switch to become apparent, the programs using Pulse must be restarted.

When everything else seems to fail

The following is a quickfix and NOT a permanent solution

On the Server:

 $ paprefs 

Go to Network Access -> Enable access to local sound devices (Also check both 'Allow discover' and 'Don't require authentication').

On the Client:

 $ export PULSE_SERVER=server.ip && mplayer test.mp3

PulseAudio through JACK the new new way

This configuration only works with jackdbus (JACK2 compiled with D-Bus support).

load-module module-jackdbus-detect

As described on the Jack-DBUS Packaging page:

Server auto-launching is implemented as D-Bus call that auto-activates JACK D-Bus service, in case it is not already started, and starts the JACK server. Correct interaction with PulseAudio is done using a D-Bus based audio card "acquire/release" mechanism. When JACK server starts, it asks this D-Bus service to acquire the audio card and PulseAudio will unconditionally release it. When JACK server stops, it releases the audio card that can be grabbed again by PulseAudio.

module-jackdbus-detect.so dynamically loads and unloads module-jack-sink and module-jack-source when jackdbus is started and stopped.

PulseAudio through JACK the new way

The basic idea is that killing PulseAudio is bad idea, it may crash any apps using PulseAudio, and disrupt any audio playing

the flow of how this setup works:

  1. PulseAudio releases the sound card
  2. JACK grabs sound card and starts up
  3. script redirects PulseAudio to JACK
  4. manually send PulseAudio apps to JACK output (pavucontrol may come in helpful for this)
  5. use JACK programs etc
  6. via script, stop redirecting PulseAudio to JACK
  7. stop JACK and release soundcard
  8. PulseAudio grabs sound card and reroutes audio to it directly

with QJackCTL setup these scripts:

pulse-jack-pre-start.sh set it up as the execute script on startup script

pacmd suspend true

pulse-jack-post-start.sh set this one up as execute script after startup

pactl load-module module-jack-sink channels=2
pactl load-module module-jack-source channels=2
pacmd set-default-sink jack_out
pacmd set-default-source jack_in

pulse-jack-pre-stop.sh "execute script on shutdown"

SINKID=$(pactl list | grep -B 1 "Name: module-jack-sink" | grep Module | sed 's/[^0-9]//g')
SOURCEID=$(pactl list | grep -B 1 "Name: module-jack-source" | grep Module | sed 's/[^0-9]//g')
pactl unload-module $SINKID
pactl unload-module $SOURCEID
sleep 5

pulse-jack-post-stop.sh "execute script after shutdown"

pacmd suspend false

Pulseaudio through JACK the old way

The JACK-Audio-Connection-Kit is popular for audio work, and is widely supported by Linux audio applications. It fills a similar niche as Pulseaudio, but with more of an emphasis on professional audio work. In particular, audio applications such as Ardour and Audacity (recently) work well with Jack.

Pulseaudio provides module-jack-source and module-jack-sink which allow Pulseaudio to be run as a sound server above the JACK daemon. This allows the usage of per-volume adjustments and the like for the apps which need it, play-back apps for movies and audio, while allowing low-latency and inter-app connectivity for sound-processing apps which connect to JACK. However, this will prevent Pulseaudio from directly writing to the sound card buffers, which will increase overall CPU usage.

To just try PA on top of jack you can have PA load the necessary modules on start:

pulseaudio -L module-jack-sink -L module-jack-source

To use pulseaudio with JACK, JACK must be started up before Pulseaudio, using whichever method you prefer. Pulseaudio then needs to be started loading the 2 relevant modules. Edit /etc/pulse/default.pa, and change the following region:

### Load audio drivers statically (it is probably better to not load
### these drivers manually, but instead use module-hal-detect --
### see below -- for doing this automatically)
#load-module module-alsa-sink
#load-module module-alsa-source device=hw:1,0
#load-module module-oss device="/dev/dsp" sink_name=output source_name=input
#load-module module-oss-mmap device="/dev/dsp" sink_name=output source_name=input
#load-module module-null-sink
#load-module module-pipe-sink

### Automatically load driver modules depending on the hardware available
.ifexists module-udev-detect.so
load-module module-udev-detect
### Alternatively use the static hardware detection module (for systems that
### lack udev support)
load-module module-detect

to the following:

### Load audio drivers statically (it is probably better to not load
### these drivers manually, but instead use module-hal-detect --
### see below -- for doing this automatically)
#load-module module-alsa-sink
#load-module module-alsa-source device=hw:1,0
#load-module module-oss device="/dev/dsp" sink_name=output source_name=input
#load-module module-oss-mmap device="/dev/dsp" sink_name=output source_name=input
#load-module module-null-sink
#load-module module-pipe-sink
load-module module-jack-source
load-module module-jack-sink

### Automatically load driver modules depending on the hardware available
#.ifexists module-udev-detect.so
#load-module module-udev-detect
### Alternatively use the static hardware detection module (for systems that
### lack udev support)
#load-module module-detect

Basically, this prevents module-udev-detect from loading. module-udev-detect will always try to grab your sound-card (JACK has already done that, so this will cause an error). Also, the jack source and sink must be explicitly loaded.

QjackCtl with Startup/Shutdown Scripts

Using the settings listed above you can use QjackCtl to execute a script upon startup and shutdown to load/unload PulseAudio. Part of the reason you may wish to do this is that the above changes disable PulseAudio's automatic hardware detection modules. This particular setup is for using PulseAudio in an exclusive fashion with JACK, though the scripts could be modified to unload and load an alternate non-JACK setup, but killing and starting PulseAudio while programs might be using it would become problematic.

The following example could be used and modified as necessary as a startup script that daemonizes PulseAudio and loads the padevchooser program (optional, needs to be built from AUR) called jack_startup:

#Load PulseAudio and PulseAudio Device Chooser

pulseaudio -D

as well as a shutdown script to kill PulseAudio and the Pulse Audio Device Chooser, as another example called jack_shutdown also in the home directory:

#Kill PulseAudio and PulseAudio Device Chooser

pulseaudio --kill
killall padevchooser

Both scripts need to be made executable:

chmod +x jack_startup jack_shutdown

then with QjackCtl loaded, click on the Setup button and then the Options tab and tick both "Execute Script after Startup:" And "Execute Script on Shutdown:" and put either use the ... button or type the path to the scripts (assuming the scripts are in the home directory) ~/jack_startup and ~/jack_shutdown making sure to save the changes you have made.

Pulseaudio through OSS

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

 load-module module-oss

Then start Pulseaudio as usual. You should have sinks and sources for your OSS devices.

Pulseaudio from within a chroot (ex. 32-bit chroot in 64-bit install)

Since a chroot sets up an alternative root for the running/jailing of applications, pulseaudio must be installed within the chroot itself (pacman -S pulseaudio within the chroot environment).

Pulseaudio, if not set up to connect to any specific server (this can be done in /etc/pulse/client.conf, through the PULSE_SERVER environment variable, or through publishing to the local X11 properties using module-x11-publish), will attempt to connect to the local pulse server, failing which it will spawn a new pulse server. Each pulse server has a unique ID based on the machine-id value in /var/lib/dbus. To allow for chrooted apps to access the pulse server, the following directories must be mounted within the chroot:-


/dev/shm should also be mounted for efficiency and good performance. Note that mounting /home would normally also allow sharing of the ~/.pulse folder.

For specific direction on accomplishing the appropriate mounts, please refer to the wiki on installing a bundled 32-bit system, especially the additional section specific to Pulseaudio.

Simultaneous Digital and Analog Output

As an example situation: you want to listen to music at your desktop, using the regular 3.5mm analog outputs into your desktop PC speakers. You also want to feed the same music to your hi-fi system in another room via your sound card's digital output. You would think this would be a simple matter, but for many sound chipsets, it requires some editing to get it set up since the default Pulseaudio profiles allow you to use either analog or digital, but not both simultaneously.

The solution is as follows (with thanks for some useful pointers from: http://superuser.com/questions/267442/how-to-get-simultaneous-sound-from-pulseadio-on-two-outputs-analog-and-digital). Amarok is used in this example.

  • Install "pavucontrol" (pacman -S pavucontrol). You won't be needing the other useful pulse configuration utility "paprefs" for this task.
  • In a terminal, launch "pacmd" - a command line utility for loading pulse modules
  • Enter the following commands in pacmd in this order (where your numbers for "hw:0,0" are determined by entering "aplay -l" in terminal. One number is the hardware ID for the analog device, the other for the digital):
 load-module module-alsa-sink device="hw:0,0" sink_name=analog_output
 load-module module-alsa-sink device="hw:0,1" sink_name=digital_output
 load-module module-combine sink_name=analog_digital slaves=digital_output,analog_output
  • Open "pavucontrol" from a terminal
  • Under pavucontrol's "Playback Tab", for the running app Amarok, select "Simultaneous output to Internal Audio, Internal Audio" (you will need to have Amarok playing something to test this)
  • Under the "Configuration Tab" Select "Digital Stereo Duplex (IEC958)" for "Internal Audio" (your exact names may vary here, according to how ALSA has named your hardware). You should now be able to hear Amarok playing on both your analog and digital outputs!

The above solution only holds to long as you have pacmd running, and is specific for the running app (in the above example, Amarok).

To set the the sinks up every reboot and for all audio apps, add the above load-module commands to your /etc/pulse/default.pa file. You may need to add them at the start of the file, just above line "### Automatically restore the volume of streams and devices", or it may not work as expected.

Finally, use the Gnome 3 sound preferences to set your system's default "Output" to "Simultaneous output to Internal Audio, Internal Audio".


No sound after install

Flash Content

If after installing pulseaudio you get no sound from flash content, you can try installing libflashsupport-pulse from the AUR. You can also try lib32-libpulse if on a 64-bit Arch and using [multilib]'s flashplugin:

# pacman -S lib32-libcanberra-pulse lib32-alsa-plugins

No cards

If PulseAudio starts, run pacmd list. If no cards are reported, make sure that your 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 your home 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 you run it. You can prevent the effects of .asoundrc altogether by commenting the last line like this:



It may be that you have another output device set as preferred in phonon. Make sure that every setting has your 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.

Muted audio device

If you experience no audio output via any means while using ALSA as your default device, you may have to unmute your sound card. To do this, you will want to launch alsamixer and make sure each column has a green 00 under it (this can be toggled by pressing 'm')

$ alsamixer -c 0

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

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 and adjust the sleep, depending how long pulse needs to start on your system.

(sleep 30 && alsactl -f /var/lib/alsa/asound.state restore) &

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

$ /etc/rc.d/avahi-daemon restart

Glitches, skips or crackling

The 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. 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.

Choppy 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 you experience choppy sound in applications using openAL, you should also change the sample rate in /etc/openal/alsoft.conf:

frequency = 48000

Volume adjustment doesn't work properly

You might want to check


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". If you encounter this issue, you can fix it by uncommenting

flat-volumes = no



and then restarting PulseAudio by executing

pulseaudio -k

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

No mic on ThinkPad T400/T500


alsamixer -c 0

and maximize the volume of/unmute the "Internal Mic". Please improve this documentation if you know the cause of this issue.

No mic input on Acer Aspire One

Install pavucontrol, unlink the microphone channels and turn down the left one to 0. You should now be able to use your microphone. 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's 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's verify the changes to deamon.conf

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


  default-sample-rate = 96000

and it's done.

4. Restart pulseaudio to apply the new settings

pulseaudio --kill
pulseaudio --start

5. Finally check by recording and playing it back

Let's 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's play the recording...

aplay test-mic.wav

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

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