Linux как узнать версию pulseaudio

/.config/pulse/default.pa to insert/alter the set-default-sink command using the sink’s name as the numbering cannot be guaranteed repeatable.

client.conf

This is the configuration file read by every PulseAudio client application. It is used to configure runtime options for individual clients. It can be used to set and configure the default sink and source statically as well as allowing (or disallowing) clients to automatically start the server if not currently running. If autospawn is enabled, clients will automatically start PulseAudio if it is not already running when a client attempts to connect to it. This can be useful if you do not want PulseAudio to always be running to conserve system resources. Otherwise, you really should have it start with your X11 session.

Configuration command

The main command to configure a server during runtime is pacmd . Run pacmd —help for a list options, or just run pacmd to enter the shell interactive mode and Ctrl+d to exit. All modifications will immediately be applied.

Once your new settings have been tested and meet your needs, edit the default.pa accordingly to make the change persistent. See PulseAudio/Examples for some basic settings.

It is important to understand that the «sources» (processes, capture devices) and «sinks» (sound cards, servers, other processes) accessible and selectable through PulseAudio depend upon the current hardware «Profile» selected. These «Profiles» are those ALSA «pcms» listed by the command aplay -L , and more specifically by the command pacmd list-cards , which will include a line «index:», a list beginning «profiles:», and a line «active profile: » in the output, among other things. «Profiles» correspond to different card input/output configurations, notably the number of available input/output channels.

The «active profile» can be set with the command pacmd set-card-profile INDEX PROFILE , with no comma separating INDEX and PROFILE, where INDEX is just the number on the line «index:» and a PROFILE name is everything shown from the beginning of any line under «profile:» to just before the colon and first space, as shown by the command pacmd list-cards . For instance, pacmd set-card-profile 0 output:analog-stereo+input:analog-stereo .

It may be easier to select a «Profile» with a graphical tool like pavucontrol , under the «Configuration» tab, or KDE System Settings, «Multimedia/Audio and Video Settings», under the «Audio Hardware Setup» tab. Each audio «Card», which are those devices listed by the command aplay -l , or again by the command pacmd list-cards , will have its own selectable «Profile». When a «Profile» has been selected, the then available «sources» and «sinks» can be seen by using the commands pacmd list-sources and pacmd list-sinks . Note that the «index» of the available sources and sinks will change each time a card profile is changed.

The selected «Profile» can be an issue for some applications, especially the Adobe Flash players, typically /usr/lib/mozilla/plugins/libflashplayer.so and /usr/lib/PepperFlash/libpepflashplayer.so . Often, these Flash players will only work when one of the Stereo profiles is selected, and otherwise, will play video with no sound, or will simply «crash». When all else fails, you might try selecting a different profile.

Of course, when configuring some variation of Surround Sound in PulseAudio, the appropriate Surround profile will have to be selected, before Surround Sound will work, or in order to do things like remap the speaker channels.

Connection and authentication

Since PulseAudio runs as a daemon as the current user, clients needs to know where to find the daemon socket to connect to it as well as a shared random cookie file clients use to authenticate with it. By default, clients should be able to locate the daemon without problem using environment variables, X11 root window properties and finally by trying the default location ( unix:/run/user/$ID/pulse/native ). However, if you have clients that needs to access PulseAudio outside of your X11 session like mpd running as a different user, you will need to tell it how to connect to your PulseAudio instance. See PulseAudio/Examples#Allowing multiple users to share a PulseAudio daemon for a complete example. An authentication cookie containing random bytes is enabled by default to ensure audio does not leak from one user to another on a multi-user system. If you already control who can access the server using user/group permissions, you can disable the cookie by passing auth-cookie-enabled=0 to module-native-protocol-unix .

Environment variables

These two variables are the important ones in order for libpulse clients to locate PulseAudio if you moved its socket to somewhere else. See pulseaudio(1) for more details and other useful environment variables clients will read.

X11 properties

PulseAudio also uses window properties on the root window of the X11 server to help find the daemon. Since environment variables cannot be modified after child processes are started, X11 properties are more flexible because they are more easily changed because they are globally shared. As long as applications receive a DISPLAY= environment variable, it can read the most up-to-date values. X11 properties can be queried using xprop -root , or with pax11publish -d to read pulse-specific properties. pax11publish can also be used to update the properties from environment variables ( pax11publish -e , or pax11publish -r to remove them entirely). If possible, it is recommended to let PulseAudio do it by itself using the module-x11-publish module or the start-pulseaudio-x11 command. The following table is there only for completeness, you should not ever need to manually set these variables by hand.

Running

PulseAudio on Arch has pulseaudio.socket enabled by default for the systemd/User instance. This means that PulseAudio will automatically start when needed.

For more information, see PulseAudio: Running.

Stopping

Stop the pulseaudio.socket and pulseaudio.service user units.

Back-end configuration

If you have applications that do not support PulseAudio explicitly but rely on ALSA, these applications will try to access the sound card directly via ALSA and will therefore bypass PulseAudio. PulseAudio will thus not have access to the sound card any more. As a result, all applications relying on PulseAudio will not be working any more, leading to this issue. To prevent this, you will need to install the pulseaudio-alsa package. It contains the necessary /etc/alsa/conf.d/99-pulseaudio-default.conf for configuring ALSA to use PulseAudio. Also make sure that

/.asoundrc does not exist, as it would override the /etc/asound.conf file.

Also install lib32-libpulse and lib32-alsa-plugins if you run a x86_64 system and want to have sound for 32-bit multilib programs like Wine and Steam.

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:

Enable DTS via ALSA

To enable PulseAudio DTS (Digital Theater System) via ALSA install dcaenc AUR package and enable it:

Finally restart PulseAudio. If experience volume issues with your DTS device and/or PulseAudio, you may fix it by looking for more setting option at dcaenc’s Github.

Expose PulseAudio sources, sinks and mixers to ALSA

Although pulseaudio-alsa contains the necessary configuration file to allow ALSA applications to use PulseAudio’s default device, ALSA’s pulse plugin is more versatile than that:

The source code can be read to know all available options.

ALSA/dmix without grabbing hardware device

You may want to use ALSA directly in most of your applications while still being able to use applications which 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.

• Create a configuration file in /etc/pulse/default.pa.d/ to unload the autodetection modules and load back-end drivers statically. Add device parameters as follows:

• Optional: If you use kmix you may want to control ALSA volume instead of PulseAudio volume:

• 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 output to PulseAudio:

Install ossp package and start osspd.service .

padsp wrapper

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

You can also add a custom wrapper script like this:

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

GStreamer

Install gst-plugins-good , or gstreamer0.10-good-plugins AUR if your intended program has a legacy GStreamer implementation.

OpenAL

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

By default, OpenAL does not allow pulseaudio to move audio streams to a different device. To change this, add the allow-moves option:

libao

Edit the libao configuration file:

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

Audio post-processing

PulseEffects

PulseEffects is a GTK advanced utility for applying several audio effects (e.g. Noise reduction, Equalizer etc.) to audio input and output.

If PulseEffects plugins are greyed out after installing plugins, trying to start the daemon produces an error, or no devices are shown in the Settings > PulseAudio tab, consider clearing the cache as shown in [3].

A collection of PulseEffects presets can be found in community presets.

Equalization

If you want to use a different equalizer rather that the one integrated in #PulseEffects, there are the following options.

LADSPA module

Install pulseaudio-equalizer-ladspa , an equalizer based on LADSPA swh-plugins . Launch pulseaudio-equalizer-gtk GUI and tweak the parameters to match your expectations.

Integrated module

PulseAudio has an integrated 10-band equalizer system. In order to use it, install pulseaudio-equalizer and read the following instructions.

Load the equalizer sink and dbus-protocol module

To start the GUI, run qpaeq .

To load the equalizer and dbus module on every boot, create a .pa file in /etc/pulse/default.pa.d/ or edit

/.config/pulse/default.pa and add the following lines:

Dynamic Range Compression

Dynamic range compression can be done with #PulseEffects. Anyway PulseEffects might introduce much overhead and latency to audio stream, so if you only need a compression effect and a minor load on the system, other options are available using a module-ladspa-sink.

Steve Harris plugin

Steve Harris LADSPA is a set of plugins containing various compression modules. Install swh-plugins and edit the configuration as the following

You have to specify your card sink name, get it from pacmd list-sinks . In order to apply the changes, stop and restart Pulseaudio. The above configuration has empty control options using the default values.

To tweak the module with custom control parameters, fill them respecting the right order.

Calf plugin

For a more professional compressor, you can use the one developed by Calf Studio Gear. Install calf-ladspa AUR and edit the configuration as the following

The plugin has 11 control options. If you want to insert custom values, read the following table and do not forget to specify them in the right order.

Microphone echo/noise cancellation

Arch does not load the PulseAudio echo-cancellation module by default, therefore, we have to add it in /etc/pulse/default.pa.d/ . First you can test if the module is present with pacmd and entering list-modules . If you cannot find a line showing name: you have to create:

then restart Pulseaudio:

and check if the module is activated by starting pavucontrol . Under Recording the input device should show Echo-Cancel Source Stream from» .

Turning on beamforming=1 in the aec_args can also significantly reduce background noise if you have more than one microphone (which is common on many new laptops). However, beamforming requires specifying your mic_geometry (see below).

If you want existing streams to be automatically moved to the new sink and source, you have to load the module-switch-on-connect with ignore_virtual=no before.

Possible ‘aec_args’ for ‘aec_method=webrtc’

Here is a list of possible ‘aec_args’ for ‘aec_method=webrtc’ with their default values [5][6]:

• analog_gain_control=1 — Analog AGC — ‘Automatic Gain Control’ done over changing the volume directly — Will most likely lead to distortions.
• digital_gain_control=0 — Digital AGC — ‘Automatic Gain Control’ done in post processing (higher CPU load).
• experimental_agc=0 — Allow enabling of the webrtc experimental AGC mechanism.
• agc_start_volume=85 — Initial volume when using AGC — Possible values 0-255 — A too low initial volume may prevent the AGC algorithm from ever raising the volume high enough [7].
• high_pass_filter=1 — ?
• noise_suppression=1 — Noise suppression.
• voice_detection=1 — VAD — Voice activity detection.
• extended_filter=0 — The extended filter is more complex and less sensitive to incorrect delay reporting from the hardware than the regular filter. The extended filter mode is disabled by default, because it seemed produce worse results during double-talk [8]. Enable this option if your microphone or speaker has a larger latency, for example, if you use a wireless microphone or some HDMI TVs as speaker.
• intelligibility_enhancer=0 — Some bits for webrtc intelligibility enhancer.
• drift_compensation=0 — Drift compensation to allow echo cancellation between different devices (such as speakers on your laptop and the microphone on your USB webcam). — only possible with «mobile=0».
• beamforming=0 — This can significantly reduce background noise. See [9][10]
  • mic_geometry=x1,y1,z1,x2,y2,z2 — Only with «beamforming=1».
  • target_direction=a,e,r — Only with «beamforming=1». Note: If the module does not want to load with this argument, set azimuth (a) to the desired value, but set elevation (e) and radius (r) to 0.
• mobile=0 — ?
  • routing_mode=speakerphone — Possible Values «quiet-earpiece-or-headset,earpiece,loud-earpiece,speakerphone,loud-speakerphone» — only valid with «mobile=1».
  • comfort_noise=1 — ? — only valid with «mobile=1».

Disable audio post processing in certain applications

If you are using the module-echo-cancel, you probably do not want other applications to do additional audio post processing. Here is a list for disabling audio post processing in following applications:

• Mumble:
  1. Configure -> Settings -> Check ‘Advanced’ check box -> Audio Input
  2. Echo: Select ‘Disabled’
  3. Noise Suppression: Set slider to ‘Off’
  4. Max. Aplification: Set slider to ‘1.0’
• TeamSpeak:
  1. Tools -> Options -> Capture
  2. Uncheck: ‘Typing attenuation’, ‘Remove background noise’, ‘Echo cancellation’ and ‘Echo reduction (Ducking)’
• Firefox: see Firefox tweaks#Disable WebRTC audio post processing
• Steam:
  1. In window «Friends List» -> Manage friends list settings (gear symbol) -> VOICE -> Show Advanced Settings
  2. Set the following sliders to «OFF»: «Echo cancellation», «Noise cancellation», «Automatic volume/gain control»
• Skype:
  1. Tools -> Settings. -> Audio & Video -> Microphone -> Automatically adjust microphone settings -> off

Script for reloading module-echo-cancel

Since the module-echo-cancel is not always needed, or must be reloaded if the source_master or sink_master has changed, it is nice to have a easy way to load or reload the module-echo-cancel.

Create the following script and make it executable:

To run the script easily from the graphical environment, you can create a desktop launcher for it.

Recurrent neural network noise suppression (RNNoise)

Installing the package noise-suppression-for-voice AUR will allow real-time noise suppression based on RNNoise: Learning Noise Suppression [11]. Configuration details can be found on the projects Github site [12]. One can install Cadmus ( cadmus-deb AUR or cadmus-appimage AUR ) which is a GUI frontend for @werman’s Pulse Audio real-time noise suppression plugin.

Another alternative is noisetorch AUR which is also build on top of RNNoise. There is not only input noise cancellation but also an output.

Applications

Refer to QEMU#Host for a detailed guide on how to configure pulseaudio within QEMU.

AlsaMixer.app

Here is a two examples where the first one is for ALSA and the other one is for pulseaudio. You can run multiple instances of it. Use the -w option to choose which of the control buttons to bind to the mouse wheel.

XMMS2

Make it switch to pulseaudio output:

To make xmms2 use a different output sink, e.g.:

See also the official guide [13].

KDE Plasma Workspaces and Qt4

PulseAudio will automatically be used by KDE/Qt4 applications. It is supported by default in the KDE sound mixer. For more information see the KDE page in the PulseAudio wiki.

One useful tidbit from that page is that load-module module-device-manager should be loaded. This usually happens automatically at login through the script /usr/bin/start-pulseaudio-x11 ; if you find that the module is not loaded automatically you can consider adding it manually to a configuration file in /etc/pulse/default.pa.d/ . See #Switch on connect for possible conflicts with the module-switch-on-connect .

If the phonon-gstreamer backend is used for Phonon, GStreamer should also be configured as described in #GStreamer.

Audacious

In order to use PulseAudio, set Edit → Preferences… → Devices → Playback → Device in Audacious to “default” or “pulse”. These devices are added to the drop-down list by pulseaudio-alsa .

Music Player Daemon (MPD)

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:

mpv supports PulseAudio same as written for #MPlayer. Configuration in

/.config/mpv/mpv.conf per-user, or /etc/mpv/mpv.conf system-wide.

guvcview

guvcview when using the PulseAudio input from a Webcam may have the audio input suspended resulting in no audio being recorded. You can check this by executing:

If the audio source is «suspended» then create the folowing .pa file:

And then either restarting PulseAudio or your computer will only idle the input source instead of suspending it. guvcview will then correctly record audio from the device.

Networked audio

This article or section is a candidate for merging with PulseAudio/Examples#PulseAudio over network.

One of PulseAudio’s unique features is its ability to stream audio from clients over TCP to a server running the PulseAudio daemon reliably within a LAN. Ensure that client and server systems agree on the time (i.e., use NTP), or audio streams may be choppy or may not work at all. For a more detailed guide visit the Official PulseAudio Documentation

Enable the TCP module on the server(the computer that actually outputs sound), create the folowing .pa file:

Or you can use the paprefs gui application (root is not required). Go to Network Server > Enable network access to local sound devices.

To make sure module-native-protocol-tcp is loaded on the server, you can use:

It is a requirement that both the client and server share the same cookie. Ensure that the clients and server share the same cookie file found under

/.config/pulse/cookie . It does not matter whose cookie file you use (the server or a client’s), just that the server and client(s) share the same one.

If it is undesirable to copy the cookie file from clients, anonymous clients can access the server by passing auth-anonymous to module-native-protocol-tcp on the server (again in /etc/pulse/default.pa.d/ ):

It is also possible to authenticate based on client IP address:

Change the LAN IP subnet to match that of those clients you wish to have access to the server.

Starting system-wide on boot

The pulseaudio daemon normally starts as a user service when a user logs in and attempts to play some sort of audio. For running a dedicated pulseaudio server accepting client connections over TCP, the daemon must be started on boot as a system service. To do this, create pulseaudio.service in /etc/systemd/system containing the following:

Then enable pulseaudio.service at the system level.

You will also need to disable the user-level pulseaudio service across the whole system:

This is necessary even if you are accessing the system over SSH, to make sure the user-level pulseaudio service will never start.

Selecting the Server

For a single shell or command you can set the environment variable $PULSE_SERVER to the host name or IP address of the desired PulseAudio server.

Alternatively you can create or modify

/.pulse/client.conf or /etc/pulse/client.conf to set a default-server persistently.

Tips and tricks

This article or section is a candidate for merging with PulseAudio/Examples.

Keyboard volume control

See Keyboard shortcuts#Xorg to bind the following commands to your volume keys: XF86AudioRaiseVolume , XF86AudioLowerVolume and XF86AudioMute .

First find out which sink corresponds to the audio output you would like to control. To list available sinks:

Suppose sink 0 is to be used, to raise the volume:

To lower the volume:

To mute/unmute the volume:

To mute/unmute the microphone:

Play sound from a non-interactive shell (systemd service, cron)

Set XDG_RUNTIME_DIR before the command (replace user_id with the ID of the user running PulseAudio):

Or use machinectl :

X11 Bell Events

To get pulseaudio to handle X11 bell events, run the following commands after the X11 session has been started:

Or use configuration files /etc/pulse/default.pa.d/ or

To adjust the volume of the X11 bell, run the following command:

100 is a percentage. This requires the xorg-xset package. See Autostarting for a way to run these commands automatically when the X11 session is started.

Switch on connect

This is a default enabled module used to switch the output sound to the newly connected device. For example, if you plug in a USB headset, the output will be switched to that. If you unplug it, the output will be set back to the last device. This used to be quite buggy but got a lot of attention in PulseAudio 8.0 and should work quite well now.

Script for switching analog outputs

Some sound cards present the option of multiple analog outputs, being switchable through using Pulseaudio profiles. But switching manually can become a chore, so you can use the following commands to switch it:

This will set the default output to whatever port you chose. Example:

The values can be easily obtained using:

Current output can be obtained through:

This process can be automated through a simple script. This script then can be given a shortcut by the user:

This script is intended to swap between two profiles. First checking the current profile then swapping it. Users are required to change the field ‘active profile’ according to the language pactl reports. Users might need to change the number of the card and the output to fit their machine.

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