pipewire-props(7)

NAME
pipewire-props - pipewire-props
 PipeWire object property reference.

DESCRIPTION

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

COMMON DEVICE PROPERTIES

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
    a short device nickname

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
    vendor ID if applicable

device.vendor.name # string
    vendor name if applicable

device.product.id # integer
    product name if applicable

device.product.name # string
    product ID if applicable

device.class # string
    device class

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.

COMMON NODE PROPERTIES

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.

  Identifying Properties
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.

  Classifying Properties
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..

  Scheduling Properties
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.

  Session Manager Properties
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.

  Format Properties

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 ]'

  Other Properties

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.

AUDIO ADAPTER PROPERTIES

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.

  Merger Parameters

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.

  Resampler Parameters

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.

  Channel Mixer Parameters

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 Parameters

debug.wav-path = ''
    Make the stream to also write the raw samples to a WAV file for debugging purposes.

  Other Parameters

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.

ALSA PROPERTIES

  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. This can be interesting to work around bugs in the mixer detection or decibel reporting. The hardware mixer will still be used to mute unused audio paths in the device. Use api.alsa.disable-mixer-path to also disable mixer path selection.

api.alsa.disable-mixer-path = false # boolean
    Setting this option to true will disable the hardware mixer path selection. The hardware mixer path is the configuration of the mixer depending on the jacks that are inserted in the card. If this is disabled, you will have to manually enable and disable mixer controls but it can be used to work around bugs in the mixer. The hardware mixer will still be used for volume and mute. Use api.alsa.soft-mixer to also disable hardware volume and mute.

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.

api.alsa.split-enable # boolean
    For UCM devices with split PCMs, don't split to multiple PCMs using alsa-lib plugins, but instead add api.alsa.split properties to emitted nodes with PCM splitting information.

  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

api.alsa.split.parent # boolean
    PCM is UCM SplitPCM parent PCM, to be opened with SplitPCM set.

api.alsa.split.position # JSON
    (SPA JSON list) If present, this is a virtual device corresponding to a subset of channels in an underlying PCM, listed in this property. Informative property.

api.alsa.split.hw-position # JSON
    (SPA JSON list) Channel map of the underlying split PCM. Informative property.

BLUETOOTH PROPERTIES

  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,
        "sync_factor": 2,
        "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 ] }
            ]
          }
        ]
      }
    ]

bluez5.bap-server-capabilities.rates # Array of integers
    Supported sampling frequencies for the LC3 codec (default: all). Possible values: 8000, 16000, 24000, 32000, 44100, 48000

bluez5.bap-server-capabilities.durations # Array of doubles
    Supported frame durations for the LC3 codec (default: all). Possible values: 7.5, 10

bluez5.bap-server-capabilities.channels # Array of integers
    Supported audio channel counts for the LC3 codec (default: [1, 2]). Possible values: 1, 2, 3, 4, 5, 6, 7, 8

bluez5.bap-server-capabilities.framelen_min # integer
    Minimum number of octets supported per codec frame for the LC3 codec (default: 20).

bluez5.bap-server-capabilities.framelen_max # integer
    Maximum number of octets supported per codec frame for the LC3 codec (default: 400).

bluez5.bap-server-capabilities.max_frames # integer
    Maximum number of codec frames supported per SDU for the LC3 codec (default: 2).

  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

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 name

port.alias # string
    port alias

See also
    Key Names in the API documentation for a full list.

LINK PROPERTIES

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

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
    process id (pid)

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.

RUNTIME SETTINGS

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.

ALSA CARD PROFILES

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.

OTHER OBJECT TYPES

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.

INDEX

  Monitor properties

•   alsa.udev.expose-busy

•   alsa.use-acp

•   bluez5.a2dp.opus.pro.bidi.channels

•   bluez5.a2dp.opus.pro.bidi.coupled-streams

•   bluez5.a2dp.opus.pro.bidi.frame-dms

•   bluez5.a2dp.opus.pro.bidi.locations

•   bluez5.a2dp.opus.pro.bidi.max-bitrate

•   bluez5.a2dp.opus.pro.channels

•   bluez5.a2dp.opus.pro.coupled-streams

•   bluez5.a2dp.opus.pro.frame-dms

•   bluez5.a2dp.opus.pro.locations

•   bluez5.a2dp.opus.pro.max-bitrate

•   bluez5.bap-server-capabilities.channels

•   bluez5.bap-server-capabilities.durations

•   bluez5.bap-server-capabilities.framelen_max

•   bluez5.bap-server-capabilities.framelen_min

•   bluez5.bap-server-capabilities.max_frames

•   bluez5.bap-server-capabilities.rates

•   bluez5.bcast_source.config

•   bluez5.codecs

•   bluez5.default.channels

•   bluez5.default.rate

•   bluez5.dummy-avrcp

•   bluez5.enable-hw-volume

•   bluez5.enable-msbc

•   bluez5.enable-sbc-xq

•   bluez5.hfphsp-backend

•   bluez5.hfphsp-backend-native-modem

•   bluez5.hw-offload-sco

•   bluez5.roles

  Device properties

•   api.acp.auto-port

•   api.acp.auto-profile

•   api.acp.pro-channels

•   api.acp.probe-rate

•   api.alsa.disable-mixer-path

•   api.alsa.ignore-dB

•   api.alsa.path

•   api.alsa.soft-mixer

•   api.alsa.split-enable

•   api.alsa.split.parent

•   api.alsa.use-ucm

•   bluez5.a2dp.aac.bitratemode

•   bluez5.a2dp.ldac.quality

•   bluez5.a2dp.opus.pro.application

•   bluez5.a2dp.opus.pro.bidi.application

•   bluez5.auto-connect

•   bluez5.bap.cig

•   bluez5.hw-volume

•   bluez5.profile

•   device.class

•   device.description

•   device.disabled

•   device.form-factor

•   device.icon

•   device.icon-name

•   device.intended-roles

•   device.name

•   device.nick

•   device.param.PARAM

•   device.plugged

•   device.product.id

•   device.product.name

•   device.profile

•   device.profile-set

•   device.serial

•   device.vendor.id

•   device.vendor.name

  Node properties

•   adapter.auto-port-config

•   api.alsa.auto-link

•   api.alsa.bind-ctls

•   api.alsa.disable-batch

•   api.alsa.disable-mmap

•   api.alsa.disable-tsched

•   api.alsa.headroom

•   api.alsa.htimestamp

•   api.alsa.htimestamp.max-errors

•   api.alsa.multi-rate

•   api.alsa.open.ucm

•   api.alsa.path

•   api.alsa.period-num

•   api.alsa.period-size

•   api.alsa.split.hw-position

•   api.alsa.split.position

•   api.alsa.start-delay

•   api.alsa.use-chmap

•   audio.allowed-rates

•   audio.channels

•   audio.format

•   audio.position

•   audio.rate

•   bluez5.media-source-role

•   channelmix.disable

•   channelmix.fc-cutoff

•   channelmix.hilbert-taps

•   channelmix.lfe-cutoff

•   channelmix.lock-volumes

•   channelmix.max-volume

•   channelmix.min-volume

•   channelmix.mix-lfe

•   channelmix.normalize

•   channelmix.rear-delay

•   channelmix.stereo-widen

•   channelmix.upmix

•   channelmix.upmix-method

•   clock.name

•   clock.quantum-limit

•   debug.wav-path

•   device.id

•   dither.method

•   dither.noise

•   iec958.codecs

•   latency.internal.ns

•   latency.internal.rate

•   media.artist

•   media.category

•   media.class

•   media.comment

•   media.copyright

•   media.date

•   media.filename

•   media.format

•   media.icon

•   media.icon-name

•   media.language

•   media.name

•   media.role

•   media.software

•   media.title

•   media.type

•   monitor.channel-volumes

•   node.always-process

•   node.autoconnect

•   node.description

•   node.disabled

•   node.dont-reconnect

•   node.exclusive

•   node.force-quantum

•   node.force-rate

•   node.latency

•   node.link-group

•   node.lock-quantum

•   node.lock-rate

•   node.loop.class

•   node.loop.name

•   node.name

•   node.param.PARAM

•   node.passive

•   node.pause-on-idle

•   node.rate

•   node.suspend-on-idle

•   node.target

•   node.want-driver

•   object.linger

•   priority.driver

•   priority.session

•   resample.disable

•   resample.peaks

•   resample.prefill

•   resample.quality

•   stream.dont-remix

•   target.object

  Port properties

•   port.alias

•   port.name

  Client properties

•   application.name

•   application.process.id

•   pipewire.sec.pid

AUTHORS

The PipeWire Developers <https://gitlab.freedesktop.org/pipewire/pipewire/issues>; PipeWire is available from <https://pipewire.org>

SEE ALSO

pipewire.conf(5)