Pipewire Pro Audio...A sorta guide

So, You have a nice shiny Audio Interface or maybe just want to take full control of your integrated sound. Well this will be a “sorta” guide as to how you go about that and enable Pro Audio in Pipewire.

In Pipewire all audio devices now have what is known as a “Pro Audio” Profile. This disables all the mixer options but separates every channel into several AUX# channels depending on the number of ins and outs. This allows much finer control of inputs on outputs and allows you to use loopback devices and such for specific audio routing. You can use loopback without this but mapping things when theyre all separate in pro audio is easier.

In Example, when in pro audio mode my motu M4 looks like this to pipewire

object.path = "alsa:pcm:0:hw:0,0:playback"
		api.alsa.path = "hw:0,0"
		api.alsa.pcm.card = "0"
		api.alsa.pcm.stream = "playback"
		audio.channels = "4"
		audio.position = "AUX0,AUX1,AUX2,AUX3"
		device.routes = "0"
		alsa.resolution_bits = "32"
		device.api = "alsa"
		device.class = "sound"
		alsa.class = "generic"
		alsa.subclass = "generic-mix"
		alsa.name = "USB Audio"
		alsa.id = "USB Audio"
		alsa.subdevice = "0"
		alsa.subdevice_name = "subdevice #0"
		alsa.device = "0"
		alsa.card = "0"
		alsa.card_name = "M4"
		alsa.long_card_name = "MOTU M4 at usb-0000:08:00.0-2, high speed"
		alsa.driver_name = "snd_usb_audio"
		device.profile.name = "pro-output-0"
		device.profile.description = "Pro"
		card.profile.device = "0"
		device.id = "56"
		factory.name = "api.alsa.pcm.sink"
		priority.driver = "1000"
		priority.session = "1000"
		media.class = "Audio/Sink"
		node.nick = "M4"
		node.name = "alsa_output.usb-MOTU_M4_M40000066088-00.pro-output-0"
		node.description = "M Series Pro"
		device.icon-name = "audio-card"

Notice the audio.position = “AUX0,AUX1,AUX2,AUX3” which normally would be setup as a 4.0 channel surround with AUX2/3 as rear channels. This in my case isnt desirable as those arent rear channels but L/R on my line level outputs.

to switch your device to “Pro Audio” youll need to get your cards ID. This can be done with

pw-cli ls Device

here is my output

id 56, type PipeWire:Interface:Device/3
 		object.serial = "56"
 		factory.id = "14"
 		client.id = "35"
 		device.api = "alsa"
 		device.description = "M Series"
 		device.name = "alsa_card.usb-MOTU_M4_M40000066088-00"
 		device.nick = "M4"
 		media.class = "Audio/Device"

in this case mine is ID 56 which is the very first id listed there.

Next you need the card # from Alsa which you can get with

arecord -l

mine is

**** List of CAPTURE Hardware Devices ****
card 0: M4 [M4], device 0: USB Audio [USB Audio]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

There are other ways to get that # but this ones easy and simple. So now that we have the card # lets get the profile index # with

spa-acp-tool -c "card #" lv

example output

spa-acp-tool -c 0 lv

card 0: profiles:5 devices:4 ports:2
    properties: (8)
        use-ucm = "true"
        verbose = "true"
        alsa.card = "0"
        alsa.card_name = "M4"
        alsa.long_card_name = "MOTU M4 at usb-0000:08:00.0-2, high speed"
        alsa.driver_name = "snd_usb_audio"
        device.string = "0"
        device.description = "M4"
    profile 0: name:"off" prio:0 (available: yes)
        description:"Off"
        devices: (0)
    profile 1: name:"output:analog-surround-40+input:analog-surround-40" prio:1212 (available: unknown)
        description:"Analog Surround 4.0 Output + Analog Surround 4.0 Input"
        devices: (2)
            device 3: direction:playback name:"analog-surround-40" prio:12 flags:00000000 devices: "surround40:%f" 
            device 2: direction:capture name:"analog-surround-40" prio:12 flags:00000000 devices: "surround40:%f" 
    profile 2: name:"output:analog-surround-40" prio:1200 (available: unknown)
        description:"Analog Surround 4.0 Output"
        devices: (1)
            device 3: direction:playback name:"analog-surround-40" prio:12 flags:00000000 devices: "surround40:%f" 
    profile 3: name:"input:analog-surround-40" prio:12 (available: unknown)
        description:"Analog Surround 4.0 Input"
        devices: (1)
            device 2: direction:capture name:"analog-surround-40" prio:12 flags:00000000 devices: "surround40:%f" 
  * profile 4: name:"pro-audio" prio:1 (available: yes)
        description:"Pro Audio"
        devices: (2)
          * device 0: direction:playback name:"pro-output-0" prio:0 flags:00000001 devices: "hw:0,0" 
          * device 1: direction:capture name:"pro-input-0" prio:0 flags:00000001 devices: "hw:0,0" 

This isnt the entire output just the important part, here we see Profile 4 is the “Pro Audio” profile so for my card that is index # 4.

Now you need to assign the pro audio profile. You can use Pavucontrol to do this but i prefer the native pipewire way to set it permanently

remember that card ID we got earlier? well now you need it, you can set “Pro Audio” as the default for your device with

pw-cli s "card-id" Profile '{ index: "Profile-Index", save: true }'

for me thats

pw-cli s 56 Profile '{ index: 4, save: true }'

You are now permanently using pro audio profile.

So the question is, what can you do with this? Well in my case it allows me to properly map my channels on my Motu M4 with this

in ~/.config/pipewire/media-session.d/media-session.conf in the context.modules = [ section i added


{   name = libpipewire-module-loopback
        args = {
            node.name = "M4_Line_Out"
            node.description = "M4 Line Out"
            capture.props = {
                media.class = "Audio/Sink"
                audio.position = [ FL FR ]
            }
            playback.props = {
                audio.position = [ AUX2 AUX3 ]
                node.target = "alsa_output.usb-MOTU_M4_M40000066088-00.pro-output-0"
                stream.dont-remix = true
                node.passive = true
            }
        }
    }
    
    {   name = libpipewire-module-loopback
        args = {
            node.name = "M4_Monitor_Out"
            node.description = "M4 Monitor Out"
            capture.props = {
                media.class = "Audio/Sink"
                audio.position = [ FL FR ]
            }
            playback.props = {
                audio.position = [ AUX0 AUX1 ]
                node.target = "alsa_output.usb-MOTU_M4_M40000066088-00.pro-output-0"
                stream.dont-remix = true
                node.passive = true
            }
        }
    }
    
    {   name = libpipewire-module-loopback
        args = {
            node.name = "M4_Input_L"
            node.description = "M4 Input L"
            capture.props = {
                audio.position = [ AUX0 AUX1 ]
                stream.dont-remix = true
                node.target = "alsa_input.usb-MOTU_M4_M40000066088-00.pro-input-0"
                node.passive = true
            }
            playback.props = {
                media.class = "Audio/Source"
                audio.position = [ MONO ]
            }
        }
    }
    
    {   name = libpipewire-module-loopback
        args = {
            node.name = "M4_Input_R"
            node.description = "M4 Input R"
            capture.props = {
                audio.position = [ AUX2 AUX3 ]
                stream.dont-remix = true
                node.target = "alsa_input.usb-MOTU_M4_M40000066088-00.pro-input-0"
                node.passive = true
            }
            playback.props = {
                media.class = "Audio/Source"
                audio.position = [ MONO ]
            }
        }
    }

What this does is map my Line Out, Monitor Out, and L/R Mic/Instrument inputs to their proper ports on my physical interface which Pipewire didnt have a proper default setup for. You can also use this to merge multiple outputs so you can have the same audio output to multiple devices from a single source and much much more. The pipewire wiki has a decent guide on how to set this and other things up here

The on thing you may have a hard time with on that wiki is figuring out your node target, you can find that using

pw-cli dump node

part of my output in example

id: 44
	permissions: rwxm
	type: PipeWire:Interface:Node/3
	input ports: 4/64
	output ports: 4/0
	state: "suspended"
	properties:
		object.path = "alsa:pcm:0:hw:0,0:playback"
		api.alsa.path = "hw:0,0"
		api.alsa.pcm.card = "0"
		api.alsa.pcm.stream = "playback"
		audio.channels = "4"
		audio.position = "AUX0,AUX1,AUX2,AUX3"
		device.routes = "0"
		alsa.resolution_bits = "32"
		device.api = "alsa"
		device.class = "sound"
		alsa.class = "generic"
		alsa.subclass = "generic-mix"
		alsa.name = "USB Audio"
		alsa.id = "USB Audio"
		alsa.subdevice = "0"
		alsa.subdevice_name = "subdevice #0"
		alsa.device = "0"
		alsa.card = "0"
		alsa.card_name = "M4"
		alsa.long_card_name = "MOTU M4 at usb-0000:08:00.0-2, high speed"
		alsa.driver_name = "snd_usb_audio"
		device.profile.name = "pro-output-0"
		device.profile.description = "Pro"
		card.profile.device = "0"
		device.id = "56"
		factory.name = "api.alsa.pcm.sink"
		priority.driver = "1000"
		priority.session = "1000"
		media.class = "Audio/Sink"
		node.nick = "M4"
		node.name = "alsa_output.usb-MOTU_M4_M40000066088-00.pro-output-0"
		node.description = "M Series Pro"
		device.icon-name = "audio-card"
		device.bus = "usb"
		device.bus-path = "pci-0000:08:00.0-usb-0:2:1.0"
		node.pause-on-idle = "false"
		resample.quality = "5"
		channelmix.normalize = "true"
		channelmix.mix-lfe = "true"
		audio.rate = "44100"
		factory.id = "18"
		client.id = "36"
		clock.quantum-limit = "5248"
		node.driver = "true"
		factory.mode = "merge"
		audio.adapt.follower = ""
		library.name = "audioconvert/libspa-audioconvert"
		object.id = "44"
		object.serial = "58"
		node.max-latency = "10496/44100"

in this case its part of my output node, the target for my output loopback would be node.name = “alsa_output.usb-MOTU_M4_M40000066088-00.pro-output-0” .

If you have any questions or need help setting up something specific feel free to ask and we can try and figure it out. Pipewire is both decently and poorly documented so sometimes specific setups can be a pain.

Good luck and hope this helps

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Now lets add to this

Here is a quick run down of setting latency and sample rate for Pipewire

All your pipewire settings can be put in the ~/.config/pipewire folder in your home directory. The settings files can be copied from /usr/share/pipewire. Unless you plan on setting up filter chains you can delete the copied filter-chain folder and the minimal.conf file isnt needed.

First, lets look at the alsa-monitor.conf in the ~/.config/pipewire/media-session.d/ folder

at the bottom of the alsa conf file youll find

actions = {
            update-props = {
                #node.nick                       = "My Node"
                #node.nick                       = null
                #priority.driver                 = 100
                #priority.session                = 100
                node.pause-on-idle               = false
                #resample.quality                = 4
                #channelmix.normalize            = false
                #channelmix.mix-lfe              = false
                #audio.channels                  = 2
                #audio.format                    = "S16LE"
                #audio.rate                      = 44100
                #audio.position                  = "FL,FR"
                #session.suspend-timeout-seconds = 5            # 0 disables suspend
                #monitor.channel-volumes         = false

                #latency.internal.rate = 0            # internal latency in samples
                #latency.internal.ns   = 0            # internal latency in nanoseconds

                #api.alsa.period-size   = 1024
                #api.alsa.headroom      = 0
                #api.alsa.start-delay   = 0
                #api.alsa.disable-mmap  = false
                #api.alsa.disable-batch = false
                #api.alsa.use-chmap     = false
                #iec958.codecs          = [ PCM DTS AC3 MPEG MPEG2-AAC EAC3 TrueHD DTS-HD ]
            }

I wont explain all these options but the important ones are

                node.pause-on-idle               = false
                resample.quality                = 5
                channelmix.normalize            = true
                channelmix.mix-lfe              = true
                #audio.channels                  = 2
                #audio.format                    = "S16LE"
                audio.rate                      = 44100
                #audio.position                  = "FL,FR"

You can set the alsa sample rate with audio.rate

node.pause-on-idle being false can prevent nodes being shut down and causing audible pop when started up again

resample.quality, the higher this is the better the sound quality when audio is resampled but it uses more CPU. 4-5 is good for 99% of people and you wont hear the difference going higher.

channelmix.normalize this will normalize the audio to prevent wild swings in volume between sources

channelmix.mix-lfe will mix the sub channel into your #.0 setup instead of having it in the #.1 channel

The rest of the important ones are explained here https://gitlab.freedesktop.org/pipewire/pipewire/-/wikis/Config-Devices


now the next important file for latency and sample rate is ~/.config/pipewire/pipewire.conf

in this file you will see near the top (this ones mine yours will be different)


 ## Properties for the DSP configuration.
    default.clock.rate          = 44100
    default.clock.allowed-rates = [ 44100 ]
    default.clock.quantum       = 896
    default.clock.min-quantum   = 128
    default.clock.max-quantum   = 4096
    default.clock.quantum-limit = 5248
    #default.video.width         = 640
    #default.video.height        = 480
    #default.video.rate.num      = 25
    #default.video.rate.denom    = 1

default.clock.rate and default.clock.allowed-rates are the default sample rate and the allowed alternative sample rates the device can switch to. If you set them the same anything not the default or allowed will be resampled. You can have up to 16 alternate sample rates.

default.clock.quantum is the default latency for your nodes. The simplest way to figure out what this number represents for latency is “(default.clock.quantum/clock.rate)x1000 = latency in ms”. In my case the default latency is 20ms. The exact latency will change based on the sample rate/clock rate

default.clock.min-quantum and default.clock.max-quantum are you default minimum and maximum latencies. Pipewire will try to use the lowest possible latency for any node but increase it is needed to prevent Xruns, set the minimum to the lowest you want/can use and max to the maximum you consider acceptable for latency. You will have to test this out to make sure you dont have any issues. You can force a specific latency all the time setting these as the same #.

default.clock.quantum-limit is the absolute limit for the quantum, pipewire cant push latency beyond this when attempting to prevent Xruns. This should be the absolute peak latency you will accept to prevent your audio crapping the bed.

more information here https://gitlab.freedesktop.org/pipewire/pipewire/-/wikis/Config-PipeWire

One last thing to note, if you are restricted on CPU resources you should use a 44.1khz sample rate aka 44100. This is easier for the system to process, i use to because i dont feel i need higher and it is easier/faster with lower latency.

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