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This article describes the setup of a playing sound with JACK (JACK Audio Connection Kit).

JACK is a sound server for professional audio production, that provides low-latency communication for applications that implement the JACK API (like mpv).

JACK is designed to work with only one active audio interface (sound card), because of technical reasons needed to guarantee audio quality, latency, and synchronization. Remember, JACK is designed for studio quality work, not for ease of use. If more, or different, inputs and outputs are needed, use a different audio interface. It may be possible to get multiple interfaces working at the same time with JACK, but it is not trivial, and results may vary.
When the JACK daemon is running, usual system sound may no longer be output. To restore sound there are several options[1], such as configuring applications to output to JACK, using a second soundcard, or bridging.


JACK uses ALSA for accessing the sound card hardware, be sure ALSA is working.

There are two implementations of the JACK API, both implementations are considered equivalent. JACK2 is usually the one to go for, JACK 1 is no longer under active development.

JACK is the original implementation, it uses a C API and has built-in Linux MIDI integration. JACK2 is a re-implementation in C++ that has support for multiprocessing and D-Bus, while MIDI support is handled by ALSA. Visit what are the differences between JACK 1 and JACK2 for an in-depth comparison.

USE flags

USE flags for media-sound/jack2 Jackdmp jack implemention for multi-processor machine

alsa Add support for media-libs/alsa-lib (Advanced Linux Sound Architecture)
classic Enable building of jackd
dbus Enable dbus support for anything that needs it (gpsd, gnomemeeting, etc)
doc Add extra documentation (API, Javadoc, etc). It is recommended to enable per package instead of globally
ieee1394 Enable FireWire/iLink IEEE1394 support (dv, camera, ...)
libsamplerate Build with support for converting sample rates using libsamplerate
metadata Enable metadata API
opus Enable Opus audio codec support
pam Add basic realtime configuration via sys-auth/realtime-base
systemd Enable use of systemd-specific libraries and features like socket activation or session tracking
tools Pull basic tools (e.g. jack_lsp/connect) from media-sound/jack-example-tools


Emerge the sound server:

root #emerge --ask media-sound/jack2
JACK2 will successfully compile with LTO enabled, but incoming connections will suffer a segmentation fault, fail to see ports, or fail to connect to open ports. Be sure to compile without LTO for a functional experience.

The JACK 1 package is media-sound/jack-audio-connection-kit (link provides a list of JACK 1 USE flags).

Global JACK USE flag

The global jack USE flag enables support for JACK in other packages, so they can submit sound to a JACK server:

root #euse -E jack

The euse command is part of app-portage/gentoolkit.

After setting this, be sure to update the system so the changes take effect:

root #emerge --ask --changed-use --deep @world

Additional software

There is a JACK configuration and control software package media-sound/cadence:

root #emerge --ask media-sound/cadence

Another alternative is media-sound/qjackctl:

root #emerge --ask media-sound/qjackctl

See the article on music production for software to use with JACK.

The JACK website has a page on sofware that uses JACK.


Sound server

All users that need JACK should be in the audio group:

root #usermod -a -G audio $USERNAME

JACK can be configured using the jack_control utility. A basic configuration script for JACK could look like:

CODE A basic configuration
jack_control start
jack_control ds alsa
jack_control dps device hw:2,0
jack_control dps rate 48000
jack_control dps nperiods 2
jack_control dps period 64

To determine the appropriate playback device (instead of hw:2,0):

user $aplay -l


JACK can route sound which was sent to other sound systems like PulseAudio[2] and ALSA through itself.


To use the ALSA only applications with JACK, you will need 2 things: the snd_aloop module and the jack ALSA plugin.

The snd-aloop is the Loopback virtual sound card. This will be used as the default ALSA sound card. That allow the ALSA applications to use it without the need to change their default configuration, the jack ALSA plugin will then route these applications to the JACK graph.

ALSA application <=> Loopback <=> ALSA jack plugin <=> JACK

Loopback device
Kernel configuration

Activate the following kernel options:

Device Drivers --->
    <M> Sound card support
        <M> Advanced Linux Sound Architecture --->
            <*> Generic Sound Devices --->
                <M> Generic loopback driver (PCM)

To configure the Loopback as the default ALSA sound card, we need to create the file /etc/modprobe.d/alsa.conf. Example with several sound cards, it's probably an overkill due to the evolution of the default ALSA configuration, but it has been working from years on several computers with very minor changes. It can also be used in other cases when the computer have several sound cards and a constancy order between reboot is a must.

Sound card 0 is the Loopback virtual device, sound card 1 and 2 are the built-in audio card which have 2 devices, one being the HDMI, sound card 3 is for an USB card.

FILE /etc/modprobe.d/alsa.conf
# Alsa kernel modules' configuration file.

# ALSA portion
alias char-major-116 snd
# OSS/Free portion
alias char-major-14 soundcore

# ALSA portion
alias snd-card-0 snd-aloop
alias snd-card-1 snd-intel
alias snd-card-2 snd-intel
alias snd-card-3 snd-usb-audio
# OSS/Free portion
alias sound-slot-0 snd-card-0
alias sound-slot-1 snd-card-1
alias sound-slot-2 snd-card-2
alias sound-slot-3 snd-card-3

# For the model option, see the output of 'lspci -s ... -n'
# Needed to differentiate the DMI part of the hda card from its other part.
options snd-aloop index=0
options snd-hda-intel index=1 model=8086:1c20
options snd-hda-intel index=2 model=10de:0bea
options snd-usb-audio index=3

# OSS/Free portion - card #1
alias sound-service-0-0 snd-mixer-oss
alias sound-service-0-1 snd-seq-oss
alias sound-service-0-3 snd-pcm-oss
alias sound-service-0-8 snd-seq-oss
alias sound-service-0-12 snd-pcm-oss
#  OSS/Free portion - card #2
alias sound-service-1-0 snd-mixer-oss
alias sound-service-1-3 snd-pcm-oss
alias sound-service-1-12 snd-pcm-oss
#  OSS/Free portion - card #3
alias sound-service-2-0 snd-mixer-oss
alias sound-service-2-3 snd-pcm-oss
alias sound-service-2-12 snd-pcm-oss
#  OSS/Free portion - card #4
alias sound-service-3-0 snd-mixer-oss
alias sound-service-3-3 snd-pcm-oss
alias sound-service-3-12 snd-pcm-oss
# OSS aliases
alias /dev/mixer snd-mixer-oss
alias /dev/dsp snd-pcm-oss
alias /dev/midi snd-seq-oss
# Correct number of cards. Start with zero,
# so it is number of cards minus one.
options snd cards_limit=3
Jack ALSA plugin

The jack ALSA plugin provide a better and more elegant solution than the previous ones like the zita-ajbridge (see Using an ALSA Loopback device and zita-ajbridge on the ProAudio Gentoo overlay wiki - the author of zita-ajbridge will maybe disagree on that...).


USE flags for media-plugins/alsa-plugins ALSA extra plugins

arcam_av Enables Arcam AV control plugin
debug Enable extra debug codepaths, like asserts and extra output. If you want to get meaningful backtraces see https://wiki.gentoo.org/wiki/Project:Quality_Assurance/Backtraces
ffmpeg Enable ffmpeg/libav-based audio/video codec support
jack Add support for the JACK Audio Connection Kit
libsamplerate Build with support for converting sample rates using libsamplerate
mix Enables upmix and vdownmix plugin
oss Add support for OSS (Open Sound System)
pulseaudio Add sound server support via media-libs/libpulse (may be PulseAudio or PipeWire)
speex Add support for the speex audio codec (used for speech)
usb_stream Enables usb_stream plugin

We need at least the jack USE flag.

root #emerge --ask media-plugins/alsa-plugins


FILE ~/.asoundrc
pcm.!default {
    type plug
    slave { pcm "jack" }

pcm.jack {
    type jack
    playback_ports {
        0 system:playback_1
        1 system:playback_2
    capture_ports {
        0 system:capture_1
        1 system:capture_2

ctl.mixer0 {
    type hw
    card 0

That file is necessary when using qjackctl or the command line to manage the JACK server. Cadence will manage it for You: On its System tab, select Bridge Type: ALSA -> Jack (Plugin).

Now, You can configure JACK to use the wanted and real sound card as usual, and all the ALSA applications will be available as per magic into the JACK graph.


To use PulseAudio's JACK module media-sound/pulseaudio needs to have the jack USE flag enabled.

In order to route all audio from PulseAudio to JACK, the JACK sink needs to be configured:

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

PulseAudio will recognize that JACK started and will bridge its audio to JACK.


"Cannot use real-time scheduling" error

Some applications may show a permission error when trying to connect to Jack, for example:

CODE Error message
ERROR: JACK: Cannot create thread res = 1
ERROR: JACK: JackClient::AcquireSelfRealTime error
ERROR: JACK: Cannot use real-time scheduling (RR/5) (1: Operation not permitted)

Adding the current user to the realtime group should solve this:

root #usermod -a -G realtime $USERNAME

Log out and log back in to apply group change.

See also

  • ALSA — describes the setup of a sound card with ALSA (Advanced Linux Sound Architecture).
  • PipeWire — low-latency, graph-based, processing engine and server, for interfacing with audio and video devices.
  • PulseAudio — a multi-platform, open source, sound server that provides a number of features on top of the low-level audio interface ALSA