PipeWire
PipeWire is a new low-level multimedia framework. It aims to offer capture and playback for both audio and video with minimal latency and support for PulseAudio, JACK, ALSA and GStreamer-based applications.
The daemon based on the framework can be configured to be both an audio server (with PulseAudio and JACK features) and a video capture server.
PipeWire also supports containers like Flatpak and does not rely on the audio and video user groups. Instead, it uses a Polkit-like security model, asking Flatpak or Wayland for permission to record screen or audio.
Installation
Install the pipewire package from the official repositories. There is also lib32-pipewire for multilib support.
Pipewire uses systemd/User for management of the server and automatic socket activation.
Optionally, install pipewire-docs to review the documentation.
Pipewire can work as drop-in replacement for other audio servers. See #Audio for details.
Session manager
Like JACK, PipeWire implements no connection logic internally. The burden of watching for new streams and connect them to the appropriate output device or application is left to an external component known as a session manager.
WirePlumber
WirePlumber is the recommended session manager. It is based on a modular design, with Lua plugins that implement the actual management functionality.
The stock configuration files are stored in /usr/share/wireplumber. The recommended way to customize Wireplumber is adding snippets overriding specific settings in /etc/wireplumber or ~/.config/wireplumber. [1].
WirePlumber changed its configuration format in version 0.5 from .lua to .conf. See https://pipewire.pages.freedesktop.org/wireplumber/daemon/configuration/migration.html#config-migration for migration instructions.
PipeWire Media Session
pipewire-media-session is deprecated and no longer recommended. It was mostly implemented for testing and as an example for building new session managers.
GUI
- coppwr — Low level control and diagnostic GUI for PipeWire.
- Helvum — GTK-based patchbay for PipeWire, inspired by the JACK tool catia. Does not save wire sets.
- qpwgraph — Qt-based Graph/Patchbay for PipeWire, inspired by the JACK tool QjackCtl. Saves wire sets.
- pwvucontrol — Pipewire Volume Control. Alternative to pavucontrol.
- sonusmix — Pipewire audio routing tool
Configuration
The PipeWire package provides an initial set of configuration files in /usr/share/pipewire. You should not edit these files directly, as package updates will overwrite your changes. To configure PipeWire, you can copy files from /usr/share/pipewire to the alternate system-wide location /etc/pipewire, or to the user location ~/.config/pipewire. An equally named file in a directory with a higher precedence makes the analogous files ignored. [2]
Profiles
Pipewire brings a custom "Pro Audio" profile in addition to the PulseAudio profiles, selectable through pavucontrol. The effect of which is described in the PipeWire wiki.
Usage
Audio
PipeWire can be used as an audio server, similar to PulseAudio and JACK. It aims to replace both PulseAudio and JACK, by providing a PulseAudio-compatible server implementation and ABI-compatible libraries for JACK clients. See the blog post PipeWire Late Summer Update 2020 for more information.
First, install pipewire-audio. Depending on the type of audio clients, you may also need to take some extra steps. You may need to install additional firmware for your audio device, see Advanced Linux Sound Architecture#ALSA firmware.
ALSA clients
Install pipewire-alsa (and remove pulseaudio-alsa if it was installed) to route all applications using the ALSA API through PipeWire.
PulseAudio clients
Install pipewire-pulse. It will replace pulseaudio and pulseaudio-bluetooth. Reboot, re-login or stop pulseaudio.service and start the pipewire-pulse.service user unit to see the effect.
Normally, no further action is needed as the user service pipewire-pulse.socket should be enabled automatically by the package. To check if the replacement is working, run the following command for the Server Name and default input/output:
$ pactl info
...
Server Name: PulseAudio (on PipeWire x.y.z)
...
Default Sink: alsa_output.{bus}-{device}.{profile}
Default Source: alsa_input.{bus}-{device}.{profile}
...
pactl(1) is provided by PulseAudio client library package (libpulse), which is installed with pipewire-pulse as a dependency.
Setting overall or individual channel volume
To adjust output channel volume, the sink needs to be specified using pactl get-sink-volume {sink} using the value of Default Sink: (above) or Name: (below), default sink device (@DEFAULT_SINK@), or Sink # (e.g. 1 below):
$ pactl list sinks | grep -B1 -A9 State:
Sink #1
State: RUNNING
Name: alsa_output.pci-0000_2d_00.4.analog-surround-51
...
Driver: PipeWire
...
Mute: no
Volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB, rear-left: 65536 / 100% / 0.00 dB, rear-right: 65536 / 100% / 0.00 dB, front-center: 65536 / 100% / 0.00 dB, lfe: 65536 / 100% / 0.00 dB
balance 0.00
Hint: if audio is playing, grep(1) for RUNNING as other devices will be SUSPENDED.
The balance ratio is calculated automatically. To set the overall volume of the default device use:
pactl set-sink-volume @DEFAULT_SINK@ 75%
To set individual channels, provide each channel volume separately:
pactl set-sink-volume @DEFAULT_SINK@ 100% 75% 100% 75% 100% 100%
Source inputs are handled similarly. For further configuration (e.g. regarding modules) see the official upstream Wiki about Migration from PulseAudio and Pipewire-Pulse Configuration.
JACK clients
Install pipewire-jack for JACK support. There is also lib32-pipewire-jack for multilib support.
pw-jack(1) may be used to start JACK clients, but it is technically not required, as it only serves as a wrapper around the PIPEWIRE_REMOTE, PIPEWIRE_DEBUG and PIPEWIRE_LATENCY environment variables.
It is possible to request a custom buffer size by setting a quotient of buffersize/samplerate (which equals the block latency in seconds):
PIPEWIRE_LATENCY="128/48000" application
Bluetooth devices
PipeWire handles Bluetooth audio devices if the pipewire-audio package is installed.
Automatic profile selection
WirePlumber has profile auto-switching enabled by default. It can automatically switch between HSP/HFP and A2DP profiles whenever an input stream is detected. You can disable it with the following command:
$ wpctl settings --save bluetooth.autoswitch-to-headset-profile false
pipewire-media-session has it disabled by default. You can set bluez5.autoswitch-profile property to true to enable it:
/etc/pipewire/media-session.d/bluez-monitor.conf (or ~/.config/pipewire/media-session.d/bluez-monitor.conf)
...
rules = [
{
...
actions = {
update-props = {
...
bluez5.autoswitch-profile = true
...
PipeWire patch sets for command line
qpwgraph can be used to visualize and create connections, and also save and load patch sets.
For non-GUI needs, the following are bash scripts to save wiresets, load wiresets, and dewire all connections. For saving and loading, use a command-line parameter for the filename.
pw-savewires
#!/bin/bash if [[ "$#" -ne 1 ]]; then echo echo 'usage: pw-savewires filename' echo exit 0 fi rm $1 &> /dev/null while IFS= read -r line; do link_on=`echo $line | cut -f 4 -d '"'` link_op=`echo $line | cut -f 6 -d '"'` link_in=`echo $line | cut -f 8 -d '"'` link_ip=`echo $line | cut -f 10 -d '"'` echo "Saving: " "'"$link_on:$link_op"','"$link_in:$link_ip"'" echo "'"$link_on:$link_op"','"$link_in:$link_ip"'" >> $1 done < <(pw-cli dump short link)
pw-loadwires
#!/bin/python
import sys
import csv
import os
if len(sys.argv) < 2:
print('\n usage: pw-loadwires filename\n')
quit()
with open(sys.argv[1], newline='') as csvfile:
pwwreader = csv.reader(csvfile, delimiter=',', quotechar='"')
for row in pwwreader:
print('Loading: ' + row[0] + ' --> ' + row[1])
process = os.popen('pw-link ' + row[0] + ' ' + row[1])
pw-dewire
#!/bin/bash
while read -r line; do
echo 'Dewiring: ' $line '...'
pw-link -d $line
done < <(pw-cli dump short link {{!}} grep -Eo '^[0-9]+')
Sharing audio devices with computers on the network
While PipeWire itself is not network transparent, its pulse implementation supports network streaming. An easy way to share audio between computers on the network is to use the Avahi daemon for discovery. To enable this functionality, install the pipewire-zeroconf package.
Make sure that the avahi-daemon.service is running (and UDP port 5353 is open if using a firewall) on all computers that will be sharing audio.
To share the local audio devices load the appropriate modules on the host (make sure to use the local IP address):
$ pactl load-module module-native-protocol-tcp listen=192.168.1.10 $ pactl load-module module-zeroconf-publish
Then load the discovery module on the clients:
$ pactl load-module module-zeroconf-discover
It is also possible to load the modules automatically by creating a dedicated configuration file:
/etc/pipewire/pipewire-pulse.conf.d/50-network-party.conf (or ~/.config/pipewire/pipewire-pulse.conf.d/50-network-party.conf)
pulse.cmd = [
{ cmd = "load-module" args = "module-native-protocol-tcp listen=192.168.1.10" }
{ cmd = "load-module" args = "module-zeroconf-discover" }
{ cmd = "load-module" args = "module-zeroconf-publish" }
]
Streaming audio to an AirPlay receiver
It is possible to stream audio to a device that is posing as an AirPlay Receiver. To enable this functionality, load the RAOP Discover module:
$ pactl load-module module-raop-discover
It is also possible to load this module automatically by creating a dedicated configuration file:
/etc/pipewire/pipewire.conf.d/raop-discover.conf (or ~/.config/pipewire/pipewire.conf.d/raop-discover.conf)
context.modules = [
{
name = libpipewire-module-raop-discover
args = { }
}
]
Some speakers' AirPlay implementations (like Sonos AirPlay 2 speakers) may require opening up ports 6001 and 6002 for incoming UDP traffic on your source device.
Run PipeWire on top of native JACK
PipeWire can also run as a JACK client on top of the native JACK daemon if desired.
See JACK and PipeWire (PipeWire wiki) and JACK Bridge (PipeWire wiki) for more information and additional configuration (like available channels for example).
To use it install the pipewire-jack-client and start JACK. Pipewire should be bridged automatically.
It can manually be loaded (as explained by pactl(1)) like a PulseAudio module: pactl load-module module-jackdbus-detect before starting jack.
Use ALSA dmix devices as PipeWire sinks
It is possible to have a PipeWire server (or multiple, for each user) output to ALSA via ALSA dmix devices. This allows you to use ALSA as the primary audio output system while being able to use non-ALSA devices such as Bluetooth headphones.
ALSA dmix setup
Suppose you have two cards, PCH and HDMI:
/proc/asound/cards
0 [PCH ]: HDA-Intel - HDA Intel PCH
HDA Intel PCH at 0xdff40000 irq 146
1 [HDMI ]: HDA-Intel - HDA ATI HDMI
HDA ATI HDMI at 0xdfe60000 irq 147
and your PCMs look like:
/proc/asound/pcm
00-00: ALC1220 Analog : ALC1220 Analog : playback 1 : capture 1 00-02: ALC1220 Alt Analog : ALC1220 Alt Analog : capture 1 01-03: HDMI 0 : HDMI 0 : playback 1 01-07: HDMI 1 : HDMI 1 : playback 1 01-08: HDMI 2 : HDMI 2 : playback 1 01-09: HDMI 3 : HDMI 3 : playback 1 01-10: HDMI 4 : HDMI 4 : playback 1 01-11: HDMI 5 : HDMI 5 : playback 1
and suppose your ALSA configuration looks something like this:
/etc/asound.conf
ctl.!default {
type hw
card PCH
}
pcm.!default {
type plug
slave.pcm "dmix:PCH,0"
}
pcm.dhdmi {
type plug
slave.pcm "dmix:HDMI,9"
}
In this particular example, the dmix devices would be dmix:PCH,0 and dmix:HDMI,9.
PipeWire dmix setup
First of all, stop WirePlumber from monitoring and adding hardware ALSA devices by disabling the monitor.alsa feature:
/etc/wireplumber/wireplumber.conf.d/10-disable-alsa-monitor.conf (or ~/.config/wireplumber/wireplumber.conf.d/10-disable-alsa-monitor.conf)
wireplumber.profiles = {
main = {
monitor.alsa = disabled
}
}
Now, configure PipeWire to use dmix devices. The default configuration file (/usr/share/pipewire/pipewire.conf) contains a commented out example which you can use as a basis.
Add your own element to the context.objects array:
/etc/pipewire/pipewire.conf.d/alsa-dmix.conf (or ~/.config/pipewire/pipewire.conf.d/alsa-dmix.conf)
context.objects = [
# We do not start with dmix, but with an input device.
# Do not forget to add an input device.
# On a friend's Laptop, I saw Zoom having a nervous
# breakdown and endlessly crying because no input device
# was configured! You have been warned.
{ factory = adapter
args = {
factory.name = api.alsa.pcm.source
node.name = "alsa-mic-internal" # name of pulse device (mpv)
node.description = "Mic Internal" # name of pulse device (pavucontrol)
media.class = "Audio/Source"
api.alsa.path = "hw:PCH,0"
}
}
# Okay, now we add our dmix PCMs
{ factory = adapter
args = {
factory.name = api.alsa.pcm.sink # sink for dmix
node.name = "alsa-dmix-internal" # name of pulse device (mpv)
node.description = "PCM Internal" # name of pulse device (pavucontrol)
media.class = "Audio/Sink" # Sink for dmix
api.alsa.path = "dmix:PCH,0"
}
}
{ factory = adapter
args = {
factory.name = api.alsa.pcm.sink # sink for dmix
node.name = "alsa-dmix-hdmi" # name of pulse device (mpv)
node.description = "PCM HDMI" # name of pulse device (pavucontrol)
media.class = "Audio/Sink" # Sink for dmix
# remember this is a non-default dmix from /etc/asound.conf
api.alsa.path = "dmix:HDMI,9"
}
}
]
As a user (non-root), check out the output of wpctl status, and set the default input(source) and output(sink) devices to your liking with wpctl set-default ID. ID is the number before sink/source names.
Now, you can fully test your configuration.
Switching between device profiles
Some hardware audio devices, like snd_hda_intel, function differently depending on which profile the device is running in. In the case of snd_hda_intel, there are separate profiles for HDMI and analog output.
Switching to HDMI with WirePlumber:
$ wpctl set-profile <device-ID> 3 $ wpctl status
... ├─ Sinks: │ * 53. Built-in Audio Digital Stereo (HDMI) [vol: 1.00] ...
Switching to analog with WirePlumber:
$ wpctl set-profile <device-ID> 1 $ wpctl status
... ├─ Sinks: │ * 51. Built-in Audio Analog Stereo [vol: 0.60] ...
WebRTC screen sharing
Most applications used to rely on X11 for capturing the desktop (or individual applications), for example when using WebRTC in web browsers (e.g. on Google Meet). On Wayland, the screen sharing mechanism is handled through the XDG Desktop Portal and PipeWire, which enables sharing content under Wayland with fine-grained access controls.
Firefox (84+) and Chromium (110+) support this method by default, while on older versions of Chromium (73+), one needs to enable WebRTC PipeWire support by setting the corresponding (experimental) flag at the URL chrome://flags/#enable-webrtc-pipewire-capturer or via CLI argument --enable-features=WebRTCPipeWireCapturer.
obs-studio (27+) supports this method by using the new PipeWire capture source.
Video
Although the software is not yet production-ready, it is safe to play around with. Most applications that rely on GStreamer to handle e.g. video streams should work out-of-the-box using the PipeWire GStreamer plugin, see GStreamer#PipeWire. Applications like e.g. cheese are therefore already able to share video input using it.
Using pipewire-v4l2, it should also be possible to use the pw-v4l2 script to preload a library (/lib/pipewire-0.3/v4l2/libpw-v4l2.so) that intercepts v4l2 calls and routes video through pipewire.
Audio post-processing
Pipewire module-filter-chain
Pipewire has an internal module called filter-chain that can create nodes to process audio input and output. See /usr/share/pipewire/filter-chain/ for examples including equalization, virtual surround sound, LADSPA plugins and channel mixing.
Systemwide parametric equalization
Copy /usr/share/pipewire/filter-chain/sink-eq6.conf to /etc/pipewire/pipewire.conf.d/ (or ~/.config/pipewire/pipewire.conf.d/).
Then edit sink-eq6.conf to incorporate the desired parameters. For headphones, these can be obtained from Oratory1990's database or, if not listed there, the AutoEQ project.
If you require a pre-amp, modify eq_band_1 to apply a bq_highshelf filter at frequency 0Hz with a negative gain (gains from -120 to +20dB supported):
label = bq_highshelf
control = { "Freq" = 0 "Q" = 1.0 "Gain" = -7.5 }
For more than 6 bands, add more entries to the nodes list and corresponding links connecting one filter ":Out" to the next filter ":In", for instance to increase to 11 bands (preamp + 10):
{ output = "eq_band_6:Out" input = "eq_band_7:In" }
{ output = "eq_band_7:Out" input = "eq_band_8:In" }
{ output = "eq_band_8:Out" input = "eq_band_9:In" }
{ output = "eq_band_9:Out" input = "eq_band_10:In" }
{ output = "eq_band_10:Out" input = "eq_band_11:In" }
Restart Pipewire, select "Equalizer Sink" as your default sound output device; this should then apply to all applications.
EasyEffects
EasyEffects (former PulseEffects) is a GTK utility which provides a large array of audio effects and filters to individual application output streams and microphone input streams. Notable effects include an input/output equalizer, output loudness equalization and bass enhancement, input de-esser and noise reduction plug-in. See the GitHub page for a full list of effects.
In order to use EasyEffects, install easyeffects. See Community Presets for a collection of preset configurations. See AutoEq for collection of algorithmically generated EQ presets for headphones.
NoiseTorch
NoiseTorch is an alternative way for noise suppression, packaged with noisetorchAUR. There also exists noisetorch-gitAUR.
After starting it the module can be loaded for the selected microphone. It is possible to adjust the voice activation threshold, which should be set to the highest level, not filtering out any actual voice.
You can start audio processing with systemd automatically, see [4]. Note that the noisetorch binary path is different if installed from AUR.
Noise suppression for voice
Install the noise-suppression-for-voice package.
Then simply follow the instructions given on GitHub.
JamesDSP
JamesDSP for Linux (available as jamesdspAUR) provides open-source sound effects for PipeWire and PulseAudio. It uses its own effects engine and without depending on LADSPA, Calf, etc. JamesDSP was initially published as an audio effects processor for Android devices.
Using LADSPA, LV2 and VST plugins
If you want to choose between the full list of available LADSPA, LV2 and VST plugins, you can apply them using carla with pipewire-jack.
Start Carla and go to Settings > Configure Carla > Engine. Make sure Audio driver is set to JACK and choose a process mode depending on your needs. You can also choose the process mode by running Carla with a specific command, for example carla-rack for the Continuous Rack mode.
You can connect application outputs to Carla manually, but if you want to pass multiple applications through Carla, it might be more convenient to create a single virtual device between applications and Carla and optionally use it as a default device. At the begin, create a new null sink named default_null_sink.
/etc/pipewire/pipewire.conf.d/10-default-null-sink.conf (or ~/.config/pipewire/pipewire.conf.d/10-default-null-sink.conf)
context.objects = [
{
factory = adapter
args = {
factory.name = support.null-audio-sink
node.name = "default_null_sink"
media.class = Audio/Sink
audio.position = [ FL FR ]
monitor.channel-volumes = true
monitor.passthrough = true
}
}
]
Restart PipeWire to apply changes.
Alternatively, you can create a temporary virtual device with pw-cli(1) or, if pipewire-pulse is installed, with pactl(1). See the PipeWire wiki for details.
In the Rack tab, add whichever plugin you want. Make sure they are stereo type. You can change their order. In the Continuous Rack process mode, the one on top of the list will be the first to receive the audio stream, just like in EasyEffects. Afterwards go to the Patchbay tab and connect the default_null_sink L/R monitors to Carla inputs, then Carla outputs to the playbacks of your desired device (speakers, earphones, HDMI, etc). Save the configuration to a local file, for example ~/Documents/carla_sink_effects.carxp. Carla will automatically restore the connections after opening this file.
You can test the effects while a multimedia application is reproducing audio, i.e. watching a video on a website through Firefox. There are two methods to do it. The first one, inside Carla Patchbay tab, disconnecting all Firefox connections and linking its L/R outputs to default_null_sink playbacks. The second through pavucontrol, locating Firefox audio stream and redirecting it to default_null_sink (this should remember the connection to automatically redirect the application to the same sink on the next instance).
To run Carla with the Continuous Rack process mode and load the saved file at startup, create a systemd user service:
~/.config/systemd/user/jack-carla-rack.service
[Unit] Description=Load Carla Rack JACK host [Service] Environment=PIPEWIRE_LINK_PASSIVE=true Type=exec ExecStart=/usr/bin/carla-rack --no-gui %h/Documents/carla_sink_effects.carxp [Install] WantedBy=default.target
Then enable the jack-carla-rack.service user unit.
Note that if you set the default_null_sink as the default device in system settings, all applications will be redirected to it and the volume keys will change its level, not the one on the speakers. If you want to control volume speakers, leave them as the default in system settings and redirect your desired application to default_null_sink inside pavucontrol (Pipewire compatibility layer will remember the connection on the next instance of the same application).
Troubleshooting
Audio
Microphone is not detected by PipeWire
PipeWire's alsa-monitor module uses alsa-card-profiles to detect devices by default. If this is not working for you, try to turn off api.alsa.use-acp, or optionally turn on api.alsa.use-ucm in wireplumber:
/etc/wireplumber/wireplumber.conf.d/50-alsa-config.conf (or ~/.config/wireplumber/wireplumber.conf.d/50-alsa-config.conf)
monitor.alsa.properties = {
# Use ALSA-Card-Profile devices. They use UCM or the profile
# configuration to configure the device and mixer settings.
# alsa.use-acp = true
# Use UCM instead of profile when available. Can be disabled
# to skip trying to use the UCM profile.
alsa.use-ucm = true
}
With pipewire-media-session:
/etc/pipewire/media-session.d/alsa-monitor.conf (or ~/.config/pipewire/media-session.d/alsa-monitor.conf)
...
rules = [
{
...
actions = {
update-props = {
...
api.alsa.use-acp = false
...
Then, restart WirePlumber and check available devices:
$ pw-record --list-targets
Available targets ("*" denotes default): 62
58: description="Built-in Audio" prio=1872
60: description="Built-in Audio" prio=2000
* 62: description="Built-in Audio (Loopback PCM)" prio=1984
An alternative solution suggested in this PipeWire issue is to add the microphone manually. First of all, make sure the microphone is detected by ALSA.
$ arecord -l
**** List of CAPTURE Hardware Devices **** card card_number: card_name, device device_number: device_name ...
Choose your microphone from the list, and to further test the microphone, run the following commands.
$ arecord --duration=5 --format=dat --device=hw:card_number,device_number test-mic.wav # record from the mic $ aplay test-mic.wav # play it
If the microphone is working with arecord, but not detected by PipeWire, try to add a config file to manually add this device.
/etc/pipewire/pipewire.conf.d/microphone.conf (or ~/.config/pipewire/pipewire.conf.d/microphone.conf)
context.objects = [
{ factory = adapter
args = {
factory.name = api.alsa.pcm.source
node.name = "microphone"
node.description = "Undetected Microphone"
media.class = "Audio/Source"
api.alsa.path = "hw:card_number,device_number"
}
}
]
And then restart PipeWire to reload the config.
Sound does not automatically switch when connecting a new device
To automatically switch to newly connected devices, create this file:
/etc/pipewire/pipewire-pulse.conf.d/switch-on-connect.conf (or ~/.config/pipewire/pipewire-pulse.conf.d/switch-on-connect.conf)
pulse.cmd = [
{ cmd = "load-module" args = "module-switch-on-connect" }
]
Then restart the pipewire-pulse.service with systemctl --user and check that module-switch-on-connect is loaded.
No sound after connecting to Bluetooth device
As of 2020-12-07, if there is no sound after connecting a Bluetooth device, you might need to switch the default sink and/or move a sink input to the correct sink. Use pactl list sinks to list the available sinks and pactl set-default-sink to switch the default sink to the Bluetooth device. This can be automated via udev using a script similar to this one.
See this Reddit thread for a discussion of the issue. According to author of the script, the headset profile (HSP) might still have problems.
No sound in mpv, vlc, totem, but sound works in web browser and GNOME speaker test
Condition Description
The best tool to verify the condition of this issue is to use mpv on a file that's expected to work with installed codecs:
mpv --ao=alsa test_file.mpv mpv --ao=pcm test_file.mpv mpv --ao=jack test_file.mpv mpv --ao=pulse test_file.mpv mpv --ao=openal test_file.mpv
This recipie applies if some or all of the above tests produce sound and the same test with pipewire option does not produce sound:
mpv --ao=pipewire test_file.mpv
Gnome desktop speaker test and web browser 'youtube' produce valid sound outcomes.
Switching inputs, muting, unmuting, changing volume in Gnome does not resolve the issue.
Sink status reported by pactl list sinks as 'SUSPENDED' is of no concern, because status properly changes when running video through a web browser.
Use of pactl info does not point to any obvious issues.
Inspection of relevant systemd unit logs does not point to any obvious issues.
Reason for the issue
It seems that a path from pipewire to hardware got muted somehow. The original author does not know how to identify and point out at the issue using command line tooling.
Solution
Install the pavucontrol package. Run pavucontrol, select appropriate source and then use Mute button to mute and unmute the source while mpv --ao=pipewire test.mp4 video is running.
Low volume
After replacing PulseAudio with Pipewire, sound may work fine, but after a reboot, the volume becomes intolerably low.
Open alsamixer, use F6 to select the proper soundcard, and make sure the ALSA volumes are at 100%. alsactl should maintain this setting after reboot.
Increasing RLIMIT_MEMLOCK
Dec 13 11:11:11 HOST pipewire-pulse[99999]: Failed to mlock memory 0x7f4f659d8000 32832: This is not a problem but for best performance, consider increasing RLIMIT_MEMLOCK
Install realtime-privileges and add your own user to the realtime group.
Alternatively, increasing memlock from 64kB to 128kB seems enough to fix this. If you are running pipewire-pulse under systemd/User, add:
username soft memlock 64 username hard memlock 128
to /etc/security/limits.d/username.conf
Changing the default sample rate
By default PipeWire sets a fixed global sample rate of 48kHz. If you need to change it (e.g. you own a DAC supporting a higher value), you can set a new default:
/etc/pipewire/pipewire.conf (or ~/.config/pipewire/pipewire.conf)
...
context.properties = {
...
default.clock.rate = sample_rate
...
Changing the allowed sample rate(s)
PipeWire can also change dynamically the output sample rates supported by your DAC. The sample rate follows the sample rate of the audio stream being played.
~/.config/pipewire/pipewire.conf
...
context.properties = {
...
default.clock.allowed-rates = [ sample_rate_1 sample_rate_2 sample_rate_3 ... ]
...
for example, [ 44100 48000 88200 96000 ].
According to the developer: "PipeWire allows up to 16 different sample rates and will switch when possible". That means, with configuration above, no resampling is done when supported. Since PipeWire 0.3.61 up to 32 different sample rates can be configured.
Consult your hardware manual for supported values of your DAC. Supported rates by the kernel driver codec are listed with the following command.
$ grep -E 'Codec|Audio Output|rates' /proc/asound/card*/codec#*
To check out which output sample rate are configured for a card run:
$ grep rate: /proc/asound/card?/pcm??/sub?/hw_params /proc/asound/card1/pcm0p/sub0/hw_params:rate: 96000 (96000/1)
In pcm0p or pcm0c c is short for "capture" and p is for "playback".
$ pw-top
also shows sample rate for each card and audio stream.
Sound quality (resampling quality)
If you used PulseAudio with resample-method = speex-float-10 or soxr-vhq, then you might consider setting resample.quality to 10 or the maximum 14:
/etc/pipewire/pipewire-client.conf.d/resample.conf (or ~/.config/pipewire/pipewire-client.conf.d/resample.conf)
stream.properties = {
resample.quality = 10
}
Do not forget to restart the pipewire.service and pipewire-pulse.socket user units (never forget pipewire-pulse.socket if you want your configuration changes to be applied).
There is a very little quality difference between 10 and 14, but the CPU load difference is 2-3x. And the latency difference between 4, 10, 14 is yet to be investigated by anybody. resample.quality = 14 on 44100→48000 Hz on Ryzen 2600 causes pipewire or pipewire-pulse processes to cause 4.0% one CPU core load.
You can compare resamplers here: https://src.infinitewave.ca/ (do not pay attention to anything above 18 KHz and over 120 dB). speex is listed as "Xiph.org Speex".
PipeWire uses its own resampling algorithm called Spa. Like with SoX's sox, Speex's speexenc, PipeWire includes its standalone version: spa-resample. Usage:
$ spa-resample -q 14 -f s24 -r 48000 input16bit44100orAnythingElse.wav output24bit48000hz.wav
It is probably somehow possible to use other resamplers by creating your own sink. Or just use a plugin in your music player (e.g., Qmmp has SoX plugin).
External sound card not activated after reconnect
Check ~/.config/pipewire/media-session.d/default-profile if there is any entry with default profile "off" and remove it. If that does not help, remove all files from ~/.config/pipewire/media-session.d/ and restart the pipewire.service user unit.
No Sound or pactl info shows Failure: Connection refused
It means applications are unable to connect to the PipeWire-Pulse service check if the pipewire-pulse.service user unit is running.
If that does not fix it, run strace -f -o /tmp/pipe.txt pactl info and pastebin /tmp/pipe.txt while seeking help on IRC (#pipewire on OFTC) or the mailing-lists.
Low audio quality on Bluetooth
In case Bluetooth playback stutters, check the unit status of the pipewire.service user unit for errors similar as below:
Feb 17 18:23:01 HOST pipewire[249297]: (bluez_input.18:54:CF:04:00:56.a2dp-sink-60) client too slow! rate:512/48000 pos:370688 status:triggered
If they appear, check the currently selected codec using pactl list sinks and try changing it by setting bluez5.codecs to one of sbc aac ldac aptx aptx_hd. You can also try mSBC support (fixes mic on Sony 1000XM3, i.e. Headphones WH-1000XM3 and Earbuds WF-1000XM3), and the SBC-XQ codec.
With wireplumber:
/etc/wireplumber/wireplumber.conf.d/51-bluez-config.conf (or ~/.config/wireplumber/wireplumber.conf.d/51-bluez-config.conf)
monitor.bluez.properties = {
bluez5.enable-sbc-xq = true
bluez5.enable-msbc = true
bluez5.codecs = [ sbc sbc_xq ]
}
With pipewire-media-session:
/etc/pipewire/media-session.d/bluez-monitor.conf (or ~/.config/pipewire/media-session.d/bluez-monitor.conf)
...
properties = {
...
bluez5.enable-msbc = true
bluez5.enable-sbc-xq = true
bluez5.codecs = [sbc sbc_xq]
...
Restart PipeWire by restarting the pipewire.service user unit for the changes to take effect.
Noticeable audio delay or audible pop/crack when starting playback
This is caused by node suspension when inactive.
With wireplumber, create a new file to overwrite the default configuration:
/etc/wireplumber/wireplumber.conf.d/51-disable-suspension.conf (or ~/.config/wireplumber/wireplumber.conf.d/51-disable-suspension.conf)
monitor.alsa.rules = [
{
matches = [
{
# Matches all sources
node.name = "~alsa_input.*"
},
{
# Matches all sinks
node.name = "~alsa_output.*"
}
]
actions = {
update-props = {
session.suspend-timeout-seconds = 0
}
}
}
]
# bluetooth devices
monitor.bluez.rules = [
{
matches = [
{
# Matches all sources
node.name = "~bluez_input.*"
},
{
# Matches all sinks
node.name = "~bluez_output.*"
}
]
actions = {
update-props = {
session.suspend-timeout-seconds = 0
}
}
}
]
Restart pipewire.service and wireplumber.service to apply changes.
Instead of disabling suspension entirely, you can also change the timeout value to the desired number of seconds of delay before source suspension.
Some devices implement their own detection of silence and suspension. For them disabling node suspention alone won't work. It's possible to work around them by adding a small amount of noise, making it so the output never goes fully silent:
.../51-disable-suspension.conf
...
session.suspend-timeout-seconds = 0, # 0 disables suspend
dither.method = "wannamaker3", # add dither of desired shape
dither.noise = 2, # add additional bits of noise
...
It may be necessary to play with dither.noise and dither.method parameters to make it so the noise is sufficiently silent and simultaneously loud enough to prevent detection of silence. See PipeWire documentation.
With pipewire-media-session:
Disable this by editing /etc/pipewire/media-session.d/*-monitor.conf depending on where the delay occurs and changing property session.suspend-timeout-seconds to 0 to disable or experiment with other values and see what works.
Alternatively you can comment out the line suspend-node in /etc/pipewire/media-session.d/media-session.conf.
Restart both pipewire.service and pipewire-pulse.service to apply these changes, or alternatively reboot.
Audio cutting out when multiple streams start playing
This problem can typically be diagnosed by reading the journal of the pipewire-pulse.service user unit and finding lines similar to:
pipewire-pulse[21740]: pulse-server 0x56009b9d5de0: [Nightly] UNDERFLOW channel:0 offset:370676 underrun:940
According to the official PipeWire troubleshooting guide, to solve this problem for wireplumber:
/etc/wireplumber/wireplumber.conf.d/50-alsa-config.conf (or ~/.config/wireplumber/wireplumber.conf.d/50-alsa-config.conf)
monitor.alsa.rules = [
{
matches = [
{
node.name = "~alsa_output.*"
}
]
actions = {
update-props = {
api.alsa.period-size = 1024
api.alsa.headroom = 8192
}
}
}
]
With pipewire-media-session:
/etc/pipewire/media-session.d/alsa-monitor.conf (or ~/.config/pipewire/media-session.d/alsa-monitor.conf
api.alsa.headroom = 1024
If you experience audio stuttering because of kernel page locking or late scheduling see Gaming#Tweaking kernel parameters for response time consistency.
Audio is distorted
- For microphones, try navigating to the card that is having issues after running
alsamixerand use the arrow keys to reduce any "Mic Boost" or "Internal Mic Boost" options. - Follow #Changing the default sample rate, reducing the sample rate to
44100(44.1 kHz).
Audio problems after standby
If the sound is missing or otherwise garbled after waking the machine up from sleep, it might help to reinitialize ALSA:
# alsactl init
High latency with USB DACs (e.g. Schiit DACs)
Changing sample rates or formats might help reduce latency with some DACs such as Schiit Hel 2.[5] Using matching rules in pipewire-media-session we can set properties for devices.[6][dead link 2024-07-30 ⓘ]
Copy the default configuration file /usr/share/pipewire/media-session.d/alsa-monitor.conf into /etc/pipewire/media-session.d/ (or ~/.config/pipewire/media-session.d/).
Then append a new rule-block similar to the following one:
/etc/pipewire/media-session.d/alsa-monitor.conf (or ~/.config/pipewire/media-session.d/alsa-monitor.conf)
...
rules = {
...
{
matches = [
{
node.name = "alsa_output.name-of-node"
}
]
actions = {
update-props = {
audio.format = "S24_3LE"
audio.rate = 96000
# Following value should be doubled until audio does not cut out or other issues stop occurring
api.alsa.period-size = 128
...
alsa_output.name-of-node node can be obtained using pw-top.
Your DAC might support a different format or sample rate. You can check what your DAC supports by querying ALSA:
First get the card number of your DAC:
$ aplay -l
... card 3: S2 [Schiit Hel 2], device 0: USB Audio [USB Audio] Subdevices: 0/1 Subdevice #0: subdevice #0 ...
So in this example it would be card 3. Get all supported sample rates and formats:
$ cat /proc/asound/cardX/streamX
...
Playback:
...
Interface 1
Altset 1
Format: S16_LE
Channels: 2
Endpoint: 0x05 (5 OUT) (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000
Data packet interval: 125 us
Bits: 16
...
Interface 1
Altset 2
Format: S24_3LE
Channels: 2
Endpoint: 0x05 (5 OUT) (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000
Data packet interval: 125 us
Bits: 24
...
Interface 1
Altset 3
Format: S32_LE
Channels: 2
Endpoint: 0x05 (5 OUT) (ASYNC)
Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000
Data packet interval: 125 us
Bits: 32
...
...
In this case S16_LE, S24_3LE, S32_LE are the supported formats and 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 are the supported sample rates across all formats.
No sound from USB DAC until 30% volume
Some USB DACs will have no sound output until a certain level of volume is reached [7]. Typically this is around 25% - 30% which then leads to an uncomfortably loud initial volume and the inability to maintain a low volume. The solution is to ignore hardware mixer volume control by setting api.alsa.soft-mixer to true.
To achieve this with wireplumber use:
/etc/wireplumber/wireplumber.conf.d/alsa-soft-mixer.conf (or ~/.config/wireplumber/wireplumber.conf.d/alsa-soft-mixer.conf)
monitor.alsa.rules = [
{
matches = [
{
device.name = "~alsa_card.*"
}
]
actions = {
update-props = {
# Do not use the hardware mixer for volume control. It
# will only use software volume. The mixer is still used
# to mute unused paths based on the selected port.
api.alsa.soft-mixer = true
}
}
}
]
Then, restart pipewire. Set your master volume in alsamixer and then save the settings by running alsactl store as root. You should now be able to use your volume mixer as normal.
Realtime audio does not work
If RTKit error: org.freedesktop.DBus.Error.AccessDenied shows up in the status of the pipewire.service user unit, then the priority of the pipewire daemon was not changed to realtime. See [8] for this issue.
Simultaneous output to multiple sinks on the same sound card
Create a copy of /usr/share/alsa-card-profile/mixer/profile-sets/default.conf so that changes persist across updates. Here we define a profile joining the two default mappings for Analog and HDMI.
/usr/share/alsa-card-profile/mixer/profile-sets/multiple.conf
[General] auto-profiles = no [Mapping analog-stereo] device-strings = front:%f channel-map = left,right paths-output = analog-output analog-output-lineout analog-output-speaker analog-output-headphones analog-output-headphones-2 paths-input = analog-input-front-mic analog-input-rear-mic analog-input-internal-mic analog-input-dock-mic analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line analog-input-headphone-mic analog-input-headset-mic priority = 15 [Mapping hdmi-stereo] description = Digital Stereo (HDMI) device-strings = hdmi:%f paths-output = hdmi-output-0 channel-map = left,right priority = 9 direction = output [Profile multiple] description = Analog Stereo Duplex + Digital Stereo (HDMI) Output output-mappings = analog-stereo hdmi-stereo input-mappings = analog-stereo
Then configure your session manager to use the new card-profile for matching devices. Identifying information can be found using pw-dump or wpctl.
For wireplumber:
/etc/wireplumber/wireplumber.conf.d/51-alsa-custom.conf (or ~/.config/wireplumber/wireplumber.conf.d/51-alsa-custom.conf)
monitor.alsa.rules = [
{
matches = [
{
device.nick = "HDA Intel PCH"
}
]
actions = {
update-props = {
api.alsa.use-acp = true
api.acp.auto-profile = false
api.acp.auto-port = false
device.profile-set = "multiple.conf"
device.profile = "multiple"
}
}
}
]
/etc/pipewire/media-session.d/alsa-monitor.conf (or ~/.config/pipewire/media-session.d/alsa-monitor.conf)
rules = [
{
matches = [ { alsa.card_name = "HDA Intel PCH" } ]
actions = {
update-props = {
api.alsa.use-acp = true
device.profile-set = "multiple.conf"
device.profile = "multiple"
api.acp.auto-profile = false
api.acp.auto-port = false
}
}
}
]
No notification sounds from Discord
This might cause by having the min.quantum too low, try setting it to more than 700. You can make an override for Discord specifically by appending the following rule to the pulse.rules section of pipewire-pulse.conf.
/etc/pipewire/pipewire-pulse.conf (or ~/.config/pipewire/pipewire-pulse.conf)
...
pulse.rules = [
...
{
# Discord notification sounds fix
matches = [ { application.process.binary = "Discord" } ]
actions = {
update-props = {
pulse.min.quantum = 1024/48000 # 21ms
}
}
}
...
FMOD games crashing under PipeWire
Some games that use an old version of the FMOD audio engine, like Pillars of Eternity, invoke pulseaudio --check and crash if the PulseAudio binary is not present. A workaround is to symlink /bin/pulseaudio to /bin/true.[9]
# ln -s /bin/true /bin/pulseaudio
Note that if you wish to reinstall PulseAudio, you need to remove the symlink.
Auto-switching is not working
If auto-switching is not working it may be an issue with WirePlumber state. As suggested by this comment you can delete WirePlumber's local state and restart the daemon to see if that helps:
$ rm -r ~/.local/state/wireplumber/
Then restart the wireplumber.service user unit.
Missing realtime priority/crackling under load after suspend
Due to a bug from 2011 in rtkit, suspend events cause PipeWire's realtime priority to be revoked and not restored. To disable the protection which causes this, edit rtkit-daemon.service:
/etc/systemd/system/rtkit-daemon.service.d/override.conf
[Service] ExecStart= ExecStart=/usr/lib/rtkit-daemon --no-canary
Then restart the rtkit-daemon.service unit and pipewire.service user unit, along with the media session service.
No sound during streaming to RAOP devices (Sonos etc.)
Set up mDNS hostname resolution using either Avahi or systemd-resolved.
No sound devices show up in KDE Plasma
PipeWire clients (including the desktop environment) may rely on the XDG_RUNTIME_DIR environment variable to connect to the PipeWire daemon. [10] If you experience no sound devices immediately after login, it may be because this variable has manually been set to the wrong path.
Although this be resolved by manually restarting PipeWire, other issues can still occur such as being unable to screen share in Chromium (with pipewire context failed). XDG_RUNTIME_DIR is automatically set by pam_systemd(8), so you should remove any instances of it being set in your initialization files.
Device volume for SDDM users is not restored on login
If you use SDDM and notice that your audio volume level is not properly restored after logging in, mask Pipewire for SDDM, since Wireplumber running under SDDM can interfere with your user's Wireplumber session.
# systemctl --user -M sddm@ mask pipewire.socket
For more details, see this Debian Wiki article.
Terminal bell not working
From PipeWire's perspective, one must have the module x11.bell loaded. This shall be the configuration default (see also in config files mentioned above). Check if you have package pipewire-x11-bell installed. Also, your window manager might influence the terminal bell, e.g., for xfwm, check in the xfwm-terminal settings that "Audible bell" is activated. Now, restart pipewire service:
$ systemctl --user restart pipewire
You can try if the termimal bell works with:
$ echo $'\a'
Video
OBS (etc.) display nothing, even if they ask for a window/screen
If you are sure that you have xdg-desktop-portal installed as well as either xdg-desktop-portal-gtk or xdg-desktop-portal-kde, check the running state of the daemons.
In OBS, if everything is working, you should see this in stdout:
... info: [pipewire] desktop selected, setting up screencast info: [pipewire] created stream 0x5632d7456850 info: [pipewire] playing stream…
For multi-monitor setups the slurp package will allow to capture of all the screens.
See also
- Wiki — PipeWire Wiki on Freedesktop GitLab
- Pipewire Update Blog Post — Blog post from January 2018 outlining the state of PipeWire at the time
- PipeWire Late Summer Update 2020 — Blog post from September 2020