pipewire-props - PipeWire object property reference.
PipeWire describes and configures audio and video elements with
objects of the following main types:
Node
Audio or video sink/source endpoint
Device
Sound cards, bluetooth devices, cameras, etc. May have
multiple nodes.
Monitor
Finding devices and handling hotplugging
Port
Audio/video endpoint in a node
Link
Connection between ports, that transporting audio/video
between them.
Client
Application connected to PipeWire.
All objects have properties ('props'), most of which can be
set in configuration files or at runtime when the object is created.
Some of the properties are 'common properties' (for example
node.description) and can be set on all objects of the given type. Other
properties control settings of a specific kinds of device or node (ALSA,
Bluetooth, ...), and have meaning only for those objects.
Usually, all the properties are configured in the session manager
configuration. For how to configure them, see the session manager
documentation. In minimal PipeWire setups without a session manager, they
can be configured via context.objects in pipewire.conf(5).
See also
WirePlumber configuration
These are common properties for devices.
device.name # string
A (unique) name for the device. It can be used by
command-line and other tools to identify the device.
device.param.PARAM = { ... } # JSON
Set value of a device Param to a JSON value when
the device is loaded. This works similarly as pw-cli(1) set-param
command. The PARAM should be replaced with the name of the Param to set,
ie. for example device.Param.Props = { ... } to set Props.
device.plugged # integer
when the device was created.
As a uint64 in nanoseconds.
device.nick # string
device.description # string
localized human readable device one-line description.
Ex. 'Foobar USB Headset'
device.serial # string
Serial number if applicable.
device.vendor.id # integer
device.vendor.name # string
vendor name if applicable
device.product.id # integer
product name if applicable
device.product.name # string
device.class # string
device.form-factor # string
form factor if applicable.
One of 'internal', 'speaker', 'handset', 'tv', 'webcam',
'microphone', 'headset', 'headphone', 'hands-free', 'car', 'hifi',
'computer', 'portable'
device.icon # string
icon for the device.
A base64 blob containing PNG image data
device.icon-name # string
an XDG icon name for the device.
Ex. 'sound-card-speakers-usb'
device.intended-roles # string
intended use.
A space separated list of roles (see media.role) this device is
particularly well suited for, due to latency, quality or form factor.
device.disabled = false # boolean
Disable the creation of this device in session
manager.
There are other common device.* properties for technical
purposes and not usually user-configurable.
See also
Key Names in the API documentation for a full
list.
The properties here apply to general audio or video input/output
streams, and other nodes such as sinks or sources corresponding to real or
virtual devices.
These contain properties to identify the node or to display the
node in a GUI application.
node.name
A (unique) name for the node. This is usually set on sink
and sources to identify them as targets for linking by the session
manager.
node.description
A human readable description of the node or stream.
media.name
A user readable media name, usually the artist and title.
These are usually shown in user facing applications to inform the user about
the current playing media.
media.title
A user readable stream title.
media.artist
A user readable stream artist
media.copyright
User readable stream copyright information
media.software
User readable stream generator software information
media.language
Stream language in POSIX format. Ex: en_GB
media.filename
File name for the stream
media.icon
Icon for the media, a base64 blob with PNG image
data
media.icon-name
An XDG icon name for the media. Ex:
audio-x-mp3
media.comment
Extra stream comment
media.date
Date of the media
media.format
User readable stream format information
object.linger = false
If the object should outlive its creator.
device.id
ID of the device the node belongs to.
The classifying properties of a node are use for routing the
signal to its destination and for configuring the settings.
media.type
The media type contains a broad category of the media
that is being processed by the node. Possible values include 'Audio', 'Video',
'Midi'
media.category
What kind of processing is done with the media. Possible
values include:
- Playback: media playback.
- Capture: media capture.
- Duplex: media capture and playback or media processing in general.
- Monitor: a media monitor application. Does not actively change media data
but monitors activity.
- Manager: Will manage the media graph.
media.role
The Use case of the media. Possible values include:
- Movie: Movie playback with audio and video.
- Music: Music listening.
- Camera: Recording video from a camera.
- Screen: Recording or sharing the desktop screen.
- Communication: VOIP or other video chat application.
- Game: Game.
- Notification: System notification sounds.
- DSP: Audio or Video filters and effect processing.
- Production: Professional audio processing and production.
- Accessibility: Audio and Visual aid for accessibility.
- Test: Test program.
media.class
The media class is to classify the stream function.
Possible values include:
- Video/Source: a producer of video, like a webcam.
- Video/Sink: a consumer of video, like a display window.
- Audio/Source: a source of audio samples like a microphone.
- Audio/Sink: a sink for audio samples, like an audio card.
- Audio/Duplex: a node that is both a sink and a source.
- Stream/Output/Audio: a playback stream.
- Stream/Input/Audio: a capture stream.
The session manager assigns special meaning to the nodes based on
the media.class. Sink or Source classes are used as targets for Stream
classes, etc..
node.latency = 1024/48000
Sets a suggested latency on the node as a fraction. This
is just a suggestion, the graph will try to configure this latency or less for
the graph. It is however possible that the graph is forced to a higher
latency.
node.lock-quantum = false
When this node is active, the quantum of the graph is
locked and not allowed to change automatically. It can still be changed
forcibly with metadata or when a node forces a quantum.
JACK clients use this property to avoid unexpected quantum
changes.
node.force-quantum = INTEGER
While the node is active, force a quantum in the graph.
The last node to be activated with this property wins.
A value of 0 unforces the quantum.
node.rate = RATE
Suggest a rate (samplerate) for the graph. The suggested
rate will only be applied when doing so would not cause interruptions (devices
are idle) and when the rate is in the list of allowed rates in the
server.
node.lock-rate = false
When the node is active, the rate of the graph will not
change automatically. It is still possible to force a rate change with
metadata or with a node property.
node.force-rate = RATE
When the node is active, force a specific sample rate on
the graph. The last node to activate with this property wins.
A RATE of 0 means to force the rate in node.rate
denominator.
node.always-process = false
When the node is active, it will always be joined with a
driver node, even when nothing is linked to the node. Setting this property to
true also implies node.want-driver = true.
This is the default for JACK nodes, that always need their process
callback called.
node.want-driver = true
The node wants to be linked to a driver so that it can
start processing. This is the default for streams and filters since 0.3.51.
Nodes that are not linked to anything will still be set to the idle state,
unless node.always-process is set to true.
node.pause-on-idle = false
node.suspend-on-idle = false
When the node is not linked anymore, it becomes idle.
Normally idle nodes keep processing and are suspended by the session manager
after some timeout. It is possible to immediately pause a node when idle with
this property.
When the session manager does not suspend nodes (or when there is
no session manager), the node.suspend-on-idle property can be used
instead.
node.loop.name = null
node.loop.class = data.rt
Add the node to a specific loop name or loop class. By
default the node is added to the data.rt loop class. You can make more
specific data loops and then assign the nodes to those.
Other well known names are main-loop.0 and the main
node.loop.class which runs the node data processing in the main loop.
priority.driver # integer
The priority of choosing this device as the driver in the
graph. The driver is selected from all linked devices by selecting the device
with the highest priority.
Normally, the session manager assigns higher priority to sources
so that they become the driver in the graph. The reason for this is that
adaptive resampling should be done on the sinks rather than the source to
avoid signal distortion when capturing audio.
clock.name # string
The name of the clock. This name is auto generated from
the card index and stream direction. Devices with the same clock name will not
use a resampler to align the clocks. This can be used to link devices together
with a shared word clock.
In Pro Audio mode, nodes from the same device are assumed to have
the same clock and no resampling will happen when linked together. So,
linking a capture port to a playback port will not use any adaptive
resampling in Pro Audio mode.
In Non Pro Audio profile, no such assumption is made and adaptive
resampling is done in all cases by default. This can also be disabled by
setting the same clock.name on the nodes.
node.autoconnect = true
Instructs the session manager to automatically connect
this node to some other node, usually a sink or source.
node.exclusive = false
If this node wants to be linked exclusively to the
sink/source.
node.target = <node.name|object.id>
Where this node should be linked to. This can be a
node.name or an object.id of a node. This property is deprecated, the
target.object property should be used instead, which uses the more unique
object.serial as a possible target.
target.object = <node.name|object.serial>
Where the node should link to, this can be a node.name or
an object.serial.
node.dont-reconnect = false
When the node has a target configured and the target is
destroyed, destroy the node as well. This property also inhibits that the node
is moved to another sink/source.
Note that if a stream should appear/disappear in sync with the
target, a session manager (WirePlumber) script should be written
instead.
node.passive = false
This is a passive node and so it should not keep
sinks/sources busy. This property makes the session manager create passive
links to the sink/sources. If the node is not otherwise linked (via a
non-passive link), the node and the sink it is linked to are idle (and
eventually suspended).
This is used for filter nodes that sit in front of sinks/sources
and need to suspend together with the sink/source.
node.link-group = ID
Add the node to a certain link group. Nodes from the same
link group are not automatically linked to each other by the session manager.
And example is a coupled stream where you don't want the output to link to the
input streams, making a useless loop.
stream.dont-remix = false
Instruct the session manager to not remix the channels of
a stream. Normally the stream channel configuration is changed to match the
sink/source it is connected to. With this property set to true, the stream
will keep its original channel layout and the session manager will link
matching channels with the sink.
priority.session # integer
The priority for selecting this node as the default
source or sink.
Streams and also most device nodes can be configured in a certain
format with properties.
audio.rate = RATE
Forces a samplerate on the node.
audio.channels = INTEGER
The number of audio channels to use. Must be a value
between 1 and 64.
audio.format = FORMAT
Forces an audio format on the node. This is the format
used internally in the node because the graph processing format is always
float 32.
Valid formats include: S16, S32, F32, F64, S16LE, S16BE, ...
audio.allowed-rates
An array of allowed samplerates for the node. ex. '[
44100 48000 ]'
node.param.PARAM = { ... } # JSON
Set value of a node Param to a JSON value when the
device is loaded. This works similarly as pw-cli(1) set-param
command. The PARAM should be replaced with the name of the Param to set,
ie. for example node.param.Props = { ... } to set Props.
node.disabled = false # boolean
Disable the creation of this node in session
manager.
Most audio nodes (ALSA, Bluetooth, audio streams from
applications, ...) have common properties for the audio adapter. The adapter
performs sample format, sample rate and channel mixing operations.
All properties listed below are node properties.
The merger is used as the input for a sink device node or a
capture stream. It takes the various channels and merges them into a single
stream for further processing.
The merger will also provide the monitor ports of the input
channels and can apply a software volume on the monitor signal.
monitor.channel-volumes = false
The volume of the input channels is applied to the volume
of the monitor ports. Normally the monitor ports expose the raw unmodified
signal on the input ports.
Source, sinks, capture and playback streams contain a high quality
adaptive resampler. It uses sinc based resampling with linear
interpolation of filter banks to perform arbitrary resample factors. The
resampler is activated in the following cases:
- The hardware of a device node does not support the graph samplerate.
Resampling will occur from the graph samplerate to the hardware
samplerate.
- The hardware clock of a device does not run at the same speed as the graph
clock and adaptive resampling is required to match the clocks.
- A stream does not have the same samplerate as the graph and needs to be
resampled.
- An application wants to activate adaptive resampling in a stream to make
it match some other clock.
PipeWire performs most of the sample conversions and resampling in
the client (Or in the case of the PulseAudio server, in the pipewire-pulse
server that creates the streams). This ensures all the conversions are
offloaded to the clients and the server can deal with one single format for
performance reasons.
Below is an explanation of the options that can be tuned in the
sample converter.
resample.quality = 4
The quality of the resampler. from 0 to 14, the default
is 4.
Increasing the quality will result in better cutoff and less
aliasing at the expense of (much) more CPU consumption. The default quality
of 4 has been selected as a good compromise between quality and performance
with no artifacts that are well below the audible range.
See Infinite Wave for a comparison of the performance.
resample.disable = false
Disable the resampler entirely. The node will only be
able to negotiate with the graph when the samplerates are compatible.
Source, sinks, capture and playback streams can apply channel
mixing on the incoming signal.
Normally the channel mixer is not used for devices, the device
channels are usually exposed as they are. This policy is usually enforced by
the session manager, so we refer to its documentation there.
Playback and capture streams are usually configured to the channel
layout of the sink/source they connect to and will thus perform channel
mixing.
The channel mixer also implements a software volume. This volume
adjustment is performed on the original channel layout. ex: A stereo
playback stream that is up-mixed to 5.1 has 2 a left an right volume
control.
channelmix.disable = false
Disables the channel mixer completely. The stream will
only be able to link to compatible sources/sinks with the exact same channel
layout.
channelmix.min-volume = 0.0
channelmix.max-volume = 10.0
Gives the min and max volume values allowed. Any volume
that is set will be clamped to these values.
channelmix.normalize = false
Makes sure that during such mixing & resampling
original 0 dB level is preserved, so nothing sounds wildly quieter/louder.
While this options prevents clipping, it can in some cases produce
too low volume. Increase the volume in that case or disable
normalization.
channelmix.lock-volumes = false
Completely disable volume or mute changes. Defaults to
false.
channelmix.mix-lfe = true
Mixes the low frequency effect channel into the front
center or stereo pair. This might enhance the dynamic range of the signal if
there is no subwoofer and the speakers can reproduce the low frequency
signal.
channelmix.upmix = true
Enables up-mixing of the front center (FC) when the
target has a FC channel. The sum of the stereo channels is used and an
optional lowpass filter can be used (see channelmix.fc-cutoff
).
Also enabled up-mixing of LFE when channelmix.lfe-cutoff is set
to something else than 0 and the target has an LFE channel. The LFE channel
is produced by adding the stereo channels.
If channelmix.upmix is true, the up-mixing of the rear channels
is also enabled and controlled with the channelmix-upmix-method
property.
channelmix.upmix-method = psd
3 methods are provided to produce the rear channels in a
surround sound:
- 1.
- none. No rear channels are produced.
- 2.
- simple. Front channels are copied to the rear. This is fast but can
produce phasing effects.
- 3.
- psd. The rear channels as produced from the front left and right ambient
sound (the difference between the channels). A delay and optional phase
shift are added to the rear signal to make the sound bigger.
channelmix.lfe-cutoff = 150
Apply a lowpass filter to the low frequency effects. The
value is expressed in Hz. Typical subwoofers have a cutoff at around 150 and
200. The default value of 0 disables the feature.
channelmix.fc-cutoff = 12000
Apply a lowpass filter to the front center frequency. The
value is expressed in Hz.
Since the front center contains the dialogs, a typical cutoff
frequency is 12000 Hz.
This option is only active when the up-mix is enabled.
channelmix.rear-delay = 12.0
Apply a delay in milliseconds when up-mixing the rear
channels. This improves specialization of the sound. A typical delay of 12
milliseconds is the default.
This is only active when the psd up-mix method is used.
channelmix.stereo-widen = 0.0
Subtracts some of the front center signal from the stereo
channels. This moves the dialogs more to the center speaker and leaves the
ambient sound in the stereo channels.
This is only active when up-mix is enabled and a Front Center
channel is mixed.
channelmix.hilbert-taps = 0
This option will apply a 90 degree phase shift to the
rear channels to improve specialization. Taps needs to be between 15 and 255
with more accurate results (and more CPU consumption) for higher values.
This is only active when the psd up-mix method is used.
dither.noise = 0
This option will add N bits of random data to the signal.
When no dither.method is specified, the random data will flip between
[-(1<<(N-1)), 0] every 1024 samples. With a dither.method, the dither
noise is amplified with 1<<(N-1) bits.
This can be used to keep some amplifiers alive during silent
periods. One or two bits of noise is usually enough, otherwise the noise
will become audible. This is usually used together with
session.suspend-timeout-seconds to disable suspend in the session
manager.
Note that PipeWire uses floating point operations with 24 bits
precission for all of the audio processing. Conversion to 24 bits integer
sample formats is lossless and conversion to 32 bits integer sample formats
are simply padded with 0 bits at the end. This means that the dither noise
is always only in the 24 most significant bits.
dither.method = none
Optional dithering
can be done on the quantized output
signal.
There are 6 modes available:
- 1.
- none No dithering is done.
- 2.
- rectangular Dithering with a rectangular noise distribution. This adds
random bits in the [-0.5, 0.5] range to the signal with even
distribution.
- 3.
- triangular Dithering with a triangular noise distribution. This add random
bits in the [-1.0, 1.0] range to the signal with triangular distribution
around 0.0.
- 4.
- triangular-hf Dithering with a sloped triangular noise distribution.
- 5.
- wannamaker3 Additional noise shaping is performed on the sloped triangular
dithering to move the noise to the more inaudible range. This is using the
'F-Weighted' noise filter described by Wannamaker.
- 6.
- shaped5 Additional noise shaping is performed on the triangular dithering
to move the noise to the more inaudible range. This is using the Lipshitz
filter.
Dithering is only useful for conversion to a format with less than
24 bits and will be disabled otherwise.
debug.wav-path = ''
Make the stream to also write the raw samples to a WAV
file for debugging purposes.
These control low-level technical features:
clock.quantum-limit
See pipewire.conf(5)
resample.peaks = false # boolean
Instead of actually resampling, produce peak amplitude
values as output. This is used for volume monitoring, where it is set as a
property of the 'recording' stream.
resample.prefill = false # boolean
Prefill resampler buffers with silence. This affects the
initial samples produced by the resampler.
adapter.auto-port-config = null # JSON
If specified, configure the ports of the node when it is
created, instead of leaving that to the session manager to do. This is useful
(only) for minimal configurations without a session manager.
Value is SPA JSON of the form:
{
mode = "none", # "none", "passthrough", "convert", "dsp"
monitor = false, # boolean
control = false, # boolean
position = "preserve" # "unknown", "aux", "preserve"
}
See spa_param_port_config for the meaning.
Monitor properties
alsa.use-acp # boolean
Use ALSA Card Profiles (ACP) for device
configuration.
alsa.udev.expose-busy # boolean
Expose the ALSA card even if it is busy/in use. Default
false. This can be useful when some of the PCMs are in use by other
applications but the other free PCMs should still be exposed.
Device properties
api.alsa.path # string
ALSA device path as can be used in snd_pcm_open() and
snd_ctl_open().
api.alsa.use-ucm = true # boolean
When ACP is enabled and a UCM configuration is available
for a device, by default it is used instead of the ACP profiles. This option
allows you to disable this and use the ACP profiles instead.
This option does nothing if api.alsa.use-acp is set to
false.
api.alsa.soft-mixer = false # boolean
Setting this option to true will disable the hardware
mixer for volume control and mute. All volume handling will then use software
volume and mute, leaving the hardware mixer untouched. The hardware mixer will
still be used to mute unused audio paths in the device.
api.alsa.ignore-dB = false # boolean
Setting this option to true will ignore the decibel
setting configured by the driver. Use this when the driver reports wrong
settings.
device.profile-set # string
This option can be used to select a custom ACP
profile-set name for the device. This can be configured in UDev rules, but it
can also be specified here. The default is to use 'default.conf' unless there
is a matching udev rule.
device.profile # string
The initial active profile name. The default is to start
from the 'Off' profile and then let session manager select the best profile
based on its policy.
api.acp.auto-profile = true # boolean
Automatically select the best profile for the device. The
session manager usually disables this, as it handles this task instead. This
can be enabled in custom configurations without the session manager handling
this.
api.acp.auto-port = true # boolean
Automatically select the highest priority port that is
available ('port' is a PulseAudio/ACP term, the equivalent of a 'Route' in
PipeWire). The session manager usually disables this, as it handles this task
instead. This can be enabled in custom configurations without the session
manager handling this.
api.acp.probe-rate # integer
Sets the samplerate used for probing the ALSA devices and
collecting the profiles and ports.
api.acp.pro-channels # integer
Sets the number of channels to use when probing the 'Pro
Audio' profile. Normally, the maximum amount of channels will be used but with
this setting this can be reduced, which can make it possible to use other
samplerates on some devices.
Node properties
audio.channels # integer
The number of audio channels to open the device with.
Defaults depends on the profile of the device.
audio.rate # integer
The audio rate to open the device with. Default is 0,
which means to open the device with a rate as close to the graph rate as
possible.
audio.format # string
The audio format to open the device in. By default this
is 'UNKNOWN', which will open the device in the best possible bits
(32/24/16/8..). You can force a format like S16_LE or S32_LE.
audio.position # JSON array of strings
The audio position of the channels in the device. This is
auto detected based on the profile. You can configure an array of channel
positions, like '[ FL, FR ]'.
audio.allowed-rates # JSON array of integers
The allowed audio rates to open the device with. Default
is '[ ]', which means the device can be opened in any supported rate.
Only rates from the array will be used to open the device. When
the graph is running with a rate not listed in the allowed-rates, the
resampler will be used to resample to the nearest allowed rate.
api.alsa.period-size # integer
The period size to open the device in. By default this is
0, which will open the device in the default period size to minimize
latency.
api.alsa.period-num # integer
The amount of periods to use in the device. By default
this is 0, which means to use as many as possible.
api.alsa.headroom # integer
The amount of extra space to keep in the ringbuffer. The
default is 0. Higher values can be configured when the device read and write
pointers are not accurately reported.
api.alsa.start-delay = 0 # integer
Some devices need some time before they can report
accurate hardware pointer positions. In those cases, an extra start delay can
be added to compensate for this startup delay. This sets the startup delay in
samples.
api.alsa.disable-mmap = false # boolean
Disable mmap operation of the device and use the ALSA
read/write API instead. Default is false, mmap is preferred.
api.alsa.disable-batch # boolean
Ignore the ALSA batch flag. If the batch flag is set,
ALSA will need an extra period to update the read/write pointers. Ignore this
flag from ALSA can reduce the latency. Default is false.
api.alsa.use-chmap # boolean
Use the driver provided channel map. Default is true when
using UCM, false otherwise because many driver don't report this
correctly.
api.alsa.multi-rate # boolean
Allow devices from the same card to be opened in multiple
sample rates. Default is true. Some older drivers did not properly advertise
the capabilities of the device and only really supported opening the device in
one rate.
api.alsa.htimestamp = false # boolean
Use ALSA htimestamps in scheduling, instead of the system
clock. Some ALSA drivers produce bad timestamps, so this is not enabled by
default and will be disabled at runtime if it looks like the ALSA timestamps
are bad.
api.alsa.htimestamp.max-errors # integer
Specify the number of consecutive errors before
htimestamp is disabled. Setting this to 0 makes htimestamp never get
disabled.
api.alsa.disable-tsched = false # boolean
Disable timer-based scheduling, and use IRQ for
scheduling instead. The 'Pro Audio' profile will usually enable this setting,
if it is expected it works on the hardware.
api.alsa.auto-link = false # boolean
Link follower PCM devices to the driver PCM device when
using IRQ-based scheduling. The 'Pro Audio' profile will usually enable this
setting, if it is expected it works on the hardware.
latency.internal.rate # integer
Static set the device systemic latency, in samples at
playback rate.
latency.internal.ns # integer
Static set the device systemic latency, in
nanoseconds.
api.alsa.path # string
UNDOCUMENTED
api.alsa.open.ucm # boolean
Open device using UCM.
api.alsa.bind-ctls # boolean
UNDOCUMENTED
iec958.codecs # JSON array of string
Enable only specific IEC958 codecs. This can be used to
disable some codecs the hardware supports. Available values: PCM, AC3, DTS,
MPEG, MPEG2-AAC, EAC3, TRUEHD, DTSHD
Monitor properties
The following are settings for the Bluetooth device monitor, not
device or node properties:
bluez5.roles = [ a2dp_sink a2dp_source bap_sink bap_source
bap_bcast_sink bap_bcast_source hfp_hf hfp_ag ] # JSON array of
string
Enabled roles.
Currently some headsets (Sony WH-1000XM3) are not working with
both hsp_ag and hfp_ag enabled, so by default we enable only HFP.
Supported roles:
- hsp_hs (HSP Headset),
- hsp_ag (HSP Audio Gateway),
- hfp_hf (HFP Hands-Free),
- hfp_ag (HFP Audio Gateway)
- a2dp_sink (A2DP Audio Sink)
- a2dp_source (A2DP Audio Source)
- bap_sink (LE Audio Basic Audio Profile Sink)
- bap_source (LE Audio Basic Audio Profile Source)
- bap_bcast_sink (LE Audio Basic Audio Profile Broadcast Sink)
- bap_bcast_source (LE Audio Basic Audio Profile Broadcast
Source)
bluez5.codecs # JSON array of string
Enabled A2DP codecs (default: all). Possible values:
sbc, sbc_xq, aac, aac_eld, aptx, aptx_hd, aptx_ll,
aptx_ll_duplex, faststream, faststream_duplex, lc3plus_h3,
ldac, opus_05, opus_05_51, opus_05_71, opus_05_duplex,
opus_05_pro, opus_g, lc3.
bluez5.default.rate # integer
Default audio rate.
bluez5.default.channels # integer
Default audio channels.
bluez5.hfphsp-backend # integer
HFP/HSP backend (default: native). Available values: any,
none, hsphfpd, ofono, native
bluez5.hfphsp-backend-native-modem # string
bluez5.dummy-avrcp player # boolean
Register dummy AVRCP player. Some devices have wrongly
functioning volume or playback controls if this is not enabled. Default:
false
bluez5.enable-sbc-xq # boolean
Override device quirk list and enable SBC-XQ for devices
for which it is disabled.
bluez5.enable-msbc # boolean
Override device quirk list and enable MSBC for devices
for which it is disabled.
bluez5.enable-hw-volume # boolean
Override device quirk list and enable hardware volume fo
devices for which it is disabled.
bluez5.hw-offload-sco # boolean
HFP/HSP hardware offload SCO support (default: false).
This feature requires a custom configuration that routes SCO audio
to ALSA nodes, in a platform-specific way. See
tests/examples/bt-pinephone.lua in WirePlumber for an example. Do not
enable this setting if you don't know what all this means, as it won't
work.
bluez5.a2dp.opus.pro.channels = 3 # integer
PipeWire Opus Pro audio profile channel count.
bluez5.a2dp.opus.pro.coupled-streams = 1 # integer
PipeWire Opus Pro audio profile coupled stream
count.
bluez5.a2dp.opus.pro.locations = 'FL,FR,LFE' # string
PipeWire Opus Pro audio profile audio channel
locations.
bluez5.a2dp.opus.pro.max-bitrate = 600000 # integer
PipeWire Opus Pro audio profile max bitrate.
bluez5.a2dp.opus.pro.frame-dms = 50 # integer
PipeWire Opus Pro audio profile frame duration (1/10
ms).
bluez5.a2dp.opus.pro.bidi.channels = 1 # integer
PipeWire Opus Pro audio profile duplex channels.
bluez5.a2dp.opus.pro.bidi.coupled-streams = 0 # integer
PipeWire Opus Pro audio profile duplex coupled stream
count.
bluez5.a2dp.opus.pro.bidi.locations = 'FC' # string
PipeWire Opus Pro audio profile duplex coupled channel
locations.
bluez5.a2dp.opus.pro.bidi.max-bitrate = 160000 #
integer
PipeWire Opus Pro audio profile duplex max bitrate.
bluez5.a2dp.opus.pro.bidi.frame-dms = 400 # integer
PipeWire Opus Pro audio profile duplex frame duration
(1/10 ms).
bluez5.bcast_source.config = [] # JSON
Example:
bluez5.bcast_source.config = [
{
"broadcast_code": "Børne House",
"encryption: false,
"bis": [
{ # BIS configuration
"qos_preset": "16_2_1", # QOS preset name from table Table 6.4 from BAP_v1.0.1.
"audio_channel_allocation": 1, # audio channel allocation configuration for the BIS
"metadata": [ # metadata configurations for the BIS
{ "type": 1, "value": [ 1, 1 ] }
]
}
]
}
]
Device properties
bluez5.auto-connect # boolean
Auto-connect devices on start up. Disabled by default if
the property is not specified.
bluez5.hw-volume = [ hfp_ag hsp_ag a2dp_source ] # JSON array
of string
Profiles for which to enable hardware volume
control.
bluez5.profile # string
Initial device profile. This usually has no effect as the
session manager overrides it.
bluez5.a2dp.ldac.quality = 'auto' # string
LDAC encoding quality Available values:
- auto (Adaptive Bitrate, default)
- hq (High Quality, 990/909kbps)
- sq (Standard Quality, 660/606kbps)
- mq (Mobile use Quality, 330/303kbps)
bluez5.a2dp.aac.bitratemode = 0 # integer
AAC variable bitrate mode. Available values: 0 (cbr,
default), 1-5 (quality level)
bluez5.a2dp.opus.pro.application = 'audio' # string
PipeWire Opus Pro Audio encoding mode: audio, voip,
lowdelay
bluez5.a2dp.opus.pro.bidi.application = 'audio' #
string
PipeWire Opus Pro Audio duplex encoding mode: audio,
voip, lowdelay
bluez5.bap.cig = 'auto' # integer, or 'auto'
Set CIG ID for BAP unicast streams of the device.
Node properties
bluez5.media-source-role # string
Media source role for Bluetooth clients connecting to
this instance. Available values:
- playback: playing stream to speakers
- input: appear as source node.
Port properties are usually not directly configurable via PipeWire
configuration files, as they are determined by applications creating them.
Below are some port properties may interesting for users:
port.name # string
port.alias # string
See also
Key Names in the API documentation for a full
list.
Link properties are usually not directly configurable via PipeWire
configuration files, as they are determined by applications creating
them.
See also
Key Names in the API documentation for a full
list.
Client properties are usually not directly configurable via
PipeWire configuration files, as they are determined by the application
connecting to PipeWire. Clients are however affected by the settings in
pipewire.conf(5) and session manager settings.
Note
Only the properties pipewire.* are safe to use for
security purposes such as identifying applications and their capabilities, as
clients can set and change other properties freely.
Below are some client properties may interesting for users.
application.name # string
application keys
application name. Ex: 'Totem Music Player'
application.process.id # integer
pipewire.sec.pid # integer
Client pid, set by protocol.
Note that for PulseAudio applications, this is the PID of the
pipewire-pulse process.
See also
Key Names in the API documentation for a full
list.
Objects such as devices and nodes also have parameters that
can be modified after the object has been created. For example, the active
device profile, channel volumes, and so on.
For some objects, the parameters also allow changing some
of the properties. The settings of most ALSA and virtual device
parameters can be configured also at runtime.
These settings are available in device parameter called
Props in its params field. They can be seen e.g. using pw-dump
<id> for an ALSA device:
{
...
"Props": [
{
...
"params": [
"audio.channels",
2,
"audio.rate",
0,
"audio.format",
"UNKNOWN",
"audio.position",
"[ FL, FR ]",
"audio.allowed-rates",
"[ ]",
"api.alsa.period-size",
0,
"api.alsa.period-num",
0,
"api.alsa.headroom",
0,
"api.alsa.start-delay",
0,
"api.alsa.disable-mmap",
false,
"api.alsa.disable-batch",
false,
"api.alsa.use-chmap",
false,
"api.alsa.multi-rate",
true,
"latency.internal.rate",
0,
"latency.internal.ns",
0,
"clock.name",
"api.alsa.c-1"
]
}
...
They generally have the same names and meaning as the
corresponding properties.
One or more params can be changed using
pw-cli(1):
pw-cli s <id> Props '{ params = [ "api.alsa.headroom" 1024 ] }'
These settings are not saved and need to be reapplied for each session
manager restart.
The sound card profiles ('Analog stereo', 'Analog stereo duplex',
...) except 'Pro Audio' come from two sources:
- UCM: ALSA Use Case Manager: the profile configuration system from ALSA.
See https://github.com/alsa-project/alsa-ucm-conf/
- ACP ('Alsa Card Profiles'): Pulseaudio's profile system ported to
PipeWire. See
https://www.freedesktop.org/wiki/Software/PulseAudio/Backends/ALSA/Profiles/
See the above links on how to configure these systems.
For ACP, PipeWire looks for the profile configuration files
under
- ~/.config/alsa-card-profile
- /etc/alsa-card-profile
- /usr/share/alsa-card-profile/mixer`.
The path and profile-set files are in subdirectories
paths and profile-sets of these directories. It is possible to
override individual files locally by putting a modified copy into the ACP
directories under ~/.config or /etc.
Technically, PipeWire objects is what are manipulated by
applications using the PipeWire API.
The list of object types that are usually 'exported' (i.e. appear
in pw-dump(1) output) is larger than considered above:
- Node
- Device
- Port
- Link
- Client
- Metadata
- Module
- Profiler
- SecurityContext
Monitors do not appear in this list; they are not usually
exported, and technically also Device objects. They are considered above as
a separate object type because they have configurable properties.
Metadata objects are what is manipulated with
pw-metadata(1)
Modules can be loaded in configuration files, or by PipeWire
applications.
The Profiler and SecurityContext objects only provide
corresponding PipeWire APIs.
The PipeWire Developers
<https://gitlab.freedesktop.org/pipewire/pipewire/issues>; PipeWire is
available from <https://pipewire.org>