I use Arch for Linux desktop and I'm overwhelmed by all the options to set up sound. Right now I have pipewire-pulse installed and that seems to just work, but I'm wondering if there's anything more I should install or setup to take advantage of my audio DAC and Sennheiser HD6XX headphones in terms of sound quality (I'm not running a complex audio setup--my only concern is maximum audio fidelity though I'm not a musician or even an audiophile who owns only a lossless collection).

The documentation for my audio DAC mentions native ALSA is preferred over Pulseaudio performance-wise, but AFAIK with just native ALSA it's not possible to have per-volume volume control (so is not really a realistic setup for desktop use where I'm using apps like mpv, Firefox, and Discord, etc.) and also Pipewire seems to be the solution for everything.

Besides pipewire-pulse package, do I want pipewire-alsa and/or pipewire-jack? Would the former provide any additional benefit like in the case of above? As for JACK, it's suppose to be low-latency but I"m not sure if that would offer benefits for my usecase or if typical Linux desktop apps can even take advantage of it (I assume most people only need to install pipewire-pulse and without further configuration for a suitable sound setup). Since these are just APIs and Pipewire used internally, wouldn't it make sense for most people to install all 3 packages as a set-and-forget where I assume applications would simply use the one they need transparently and no further configuration is required?

Any tips are much appreciated.

Hi everyone! I've recently discovered that my scarlet solo sounds better on a windows machine with official asio drivers, than on linux. I'm talking about instrument input which I use for my guitar. It sounds muddy and has less gain. I use pipewire + alsa combination, latest kernel and my system recognises it as a Focusrite interface. My distro is NixOS and I use musnix to get the real-time audio settings. I'm willing to provide DI track from both systems, or any needed additional information.

Does anyone know what is wrong? Thank you sooooo much! (⁠◕⁠ᴗ⁠◕⁠✿⁠)

Hi,

I am really interested in getting into audio development on linux and was trying to get started by practicing Rust development by using this crate.

I have been getting some stuff working, but I find it really hard to understand what the various structures do and how to use them. The registry makes sense for listening on updates to the links and devices, but creating objects seems so cryptic. I am not even sure how I would go about receiving and outputting audio. It feels like I am missing almost all the relevant knowledge about PipeWire development.

The docs leave a lot to be desired, so I was hoping that some of you might know some more in-depth documentation and examples somewhere. The stuff that I know that I don't know how to do includes:

• How to control port links
• How to send and receive audio

The stuff that I don't know that I don't know includes:

• I don't know what I don't know

Any help would be greatly appreciated!

https://www.rncbc.org/drupal/node/2592

​

https://preview.redd.it/l39pg89fztxc1.png?width=580&format=png&auto=webp&s=f293c96b6145f45ed6c3789de6cd1287cd15dc86

Hi,

I have recently installed an up to date version of the pipewire-audio meta package on Debian Sid from scratch following the Sound guide on wiki.debian.org . My audio device is a RME Babyface Pro running CC mode. I have removed any previous remains of the pulseaudio service that was responsible for sound prior pipewire installation. Adjusting levels, inputs and devices with pavucontrol.

In fact, I am running pipewire, wireplumber and pipewire-pulse and -alsa which is working (mostly) fine:

~ LANG=C pactl info | grep '^Server Name'
Server Name: PulseAudio (on PipeWire 1.0.5)

➜  ~ LANG=C aplay -L | grep -A 1 default
sysdefault
    Default Audio Device
--
default
    Default ALSA Output (currently PipeWire Media Server)
--
sysdefault:CARD=Pro71969184
    Babyface Pro (71969184), USB Audio
--
sysdefault:CARD=Generic_1
    HD-Audio Generic, ALC897 Analog

apt list -i 'pipewire*' 'wireplumber*'
pipewire-alsa/unstable,now 1.0.5-1+b3 amd64 [installed]
pipewire-bin/unstable,now 1.0.5-1+b3 amd64 [installed,automatic]
pipewire-pulse/unstable,now 1.0.5-1+b3 amd64 [installed]
pipewire/unstable,now 1.0.5-1+b3 amd64 [installed,automatic]
wireplumber/unstable,now 0.4.17-1+b2 amd64 [installed]

While I am also a gamer and happen to play DayZ through Steam (Proton 8.05) I noticed a weird issue with missing sound output from this game when using the Pro Audio configuration instead of what is called 'Analog Stereo Duplex'.

When running the Pro Audio mode in pavucontrol there is no Playback output available while playing i.e. DayZ/ Stumble Guys. Using the Analog Stereo Duplex configuration a 'DayZ Stream #1' Playback output is available that produces actual sound that I can hear.

As it is only with these two games so far that no audio is generated it surely is very limited. However, I am wondering what could be the reason for this. I have copied the default pipewire.conf to ~/.config/pipewire/. I would like to understand what causes the missing output while using the Pro Audio profile.

Any help is gladly appreciated and I am happy to elaborate on specifics.

OS: Linux Mint 21.3 x86_64 Kernel: 5.15.0-105-generic

​

https://build.opensuse.org/package/show/multimedia:proaudio/vaporizer2

Ratatouille is a Neural Model loader and mixer for Linux/Windows.

https://preview.redd.it/492qp61vklxc1.png?width=495&format=png&auto=webp&s=6ff188b6d120eec830741632f29570ba34b47ef2

Ratatouille allow to load up to two neural model files and mix there
output. Those models could be [*.nam files](https://tonehunt.org/all) or
[*.json or .aidax files](https://cloud.aida-x.cc/all). So you could
blend from clean to crunch for example, or, go wild and mix different
amp models, or mix a amp with a pedal simulation.

To round up the sound it allow to load up to two Impulse Response files
and mix there output as well. You could try the wildest combinations,
or, be conservative and load just your single preferred IR-File.

Each neural model may have a different expected Sample Rate, Ratatouille
will resample the buffer to match that.

On the Release Page been ready to use binaries, note please that
Ratatouille.lv2-v3-v0.2-linux-x86_64.tar.xz is a fully optimized version
using the x86-64-v3 optimisation, you could check if your system
supports that by running the command
`/usr/lib64/ld-linux-x86-64.so.2 --help 2>/dev/null | grep 'x86-64-v3
(supported'`

if that return nothing, your system can't use this, in that case you

should choose Ratatouille.lv2-v0.2-linux-x86_64.tar.xz

Impulse Response Files will be resampled on the fly to match the session
Sample Rate.
To build from source, please use Ratatouille.lv2-v0.2-src.tar.xz as only
that contain the needed submodules.

Release Page:
https://github.com/brummer10/Ratatouille.lv2/releases/tag/v0.2

Project Page:
https://github.com/brummer10/Ratatouille.lv2

​

​

ive been wanting to move from windows to linux for ages but always had one limiting factor in that I use a beach mix create for my audio sources to have different mixes and a routing table and there was no real way to use that in linux so I just wanna ask if there is anything out there that would work similar to those choices

Granted, I'm assuming any thing like WAVES won't work because of waveshell / pluginserver?

Kontakt should work or do I need native access on linux as well?

I use reaper (primarily, next is Ableton / FL) and want to slowly migrate my work to linux. It would be nice if I could just back up reaper settings, import them on linux as well as use YABRIDGE to also view projects made on windows.

I'm looking to get a synthesizer that can do Arpeggio stuff and I'd LOVE to be able to use software with it. One in particular I'm looking at is the Korg Wavestate MKII. It's a nice little sequencer but it supports Windows and Mac (not sure about Linux but probably not). I'd like to find software that will take full advantage of that sequencer. I have stuff like Reaper, Ardour, and a couple of others. What would be good software that could take advantage of most or all of the features with the Wavestate?

If there ISN'T any good software for Linux... I hate to say it but I think I'd be more apt to buying a Mac rather than using a Windows based machine again.

I guess I'm looking for something that will communicate with the sequencer itself. Not just record it. I can do that with Audacity I believe. I'm looking for something that will work WITH the sequencer itself.

If anyone here uses a sequencer (or better yet, a Wavestate) I'd love to hear from you!

I have my UMC1820 set up so that it works in my DAW, all the inputs are recognized in the DAW and everything is working as expected. In the system I can direct audio through and use my good headphones on calls I would like to be able to use a mic for calls but for some reason there is no input recognized over USB despite there being signal indicators on the device. It is in multichannel duplex configuration I have a temp fix where I am patching main out into my computers line in, but I would prefer to have audio over usb. I am using pipewire-pulse (I am pretty sure). Am I doing something daft or have I overlooked some other functionality?

EDIT: It seems to work via channels 1-2 but 3-8 nothing.

alo internet! New linux user here.
I'm looking to record stuff from Pure Data into something like REAPER. I think JACK would be pretty convenient for something like this.
as of now, i can't get audio sockets for Pure Data or REAPER to show up in JACK. Midi seams working fine.

My setup.

• Thinkpad P50 (Has speakers/headphone and a mic)
• Zoom U-44 (USB audio and midi interface)
• Debian 12 (xfce flavor)

So when I start the JACK server with qjackctrl the system inputs and outputs will show up in the "Graph" (four channels each). This i think is the Zoom U-44. Patched the system in to the system out, this made the U-44 playback its inputs. Also tried lookin at the situation in the "Patchbay" but it looks the same.
Midi sockets also open up for the U-44 and for PureData. No audio though.

A screenshot of the situation.

https://preview.redd.it/rvz8y4l309xc1.png?width=1920&format=png&auto=webp&s=857d2dc023bd20582173b412dfb3fc3d05f69ffd

On the pure data side: When the DSP is turned on, Pure Data shows this small "Audio I/O error" message in red for a couple seconds. Even though i have Pure Data sending to JACK in the audio settings, when i open the "Test Audio and MIDI..." patch, it plays back through the laptop's speakers, which i believe are being used by pulseaudio. [adc~] does not receive input from the laptop's mic though.

https://preview.redd.it/zyhk3vc409xc1.png?width=500&format=png&auto=webp&s=566eaf6d9ab70b3976fb2313530859883769bb65

I've heard that pulseaudio might steal audio devices. The only one it seams to be capturing is the Laptop's speaker and mic. I tried disabling and masking pusleaudio, but that didn't change any of the problems with JACK, so i used timeshift to go back before masking it. (I couldn't figure out how to unmask it)

In theory, i should be able to take the input from the U-44, put that through Pure Data, then record into REAPER. right? It would be super epic if i could steal the Laptop's mic from pulseaudio, but i've heard stuff like that can get messy?

Well there we go, i would appreciate any help with this, gettin a bit annoyed :)
Or maybe some help learning me on how some of this stuff works, i feel like im doing something wrong.
If possible, i would prefer to do stuff through UI rather than the terminal.

Hi, so I'm kind of a beginner in Linux, in order to make a clean switch from Windows I need to figure out the audio stuff.
Basically as a first step I want to be able to play my guitar through a DAW (with or without VSTs) over a youtube video.

My setup :

• Scarlett Focusrite 4i4 3rd gen
• Fedora 39
• Reaper (official package)

In Reaper audio setting, the guitar sound is all good if I choose ALSA and my interface, but then I have no sound in Firefox or other programs (intended dehavior it seems).

So I try jack (pipewire-jack in that case), and then I have audio in all programs, but the sound in Reaper is really low (I can hear it but need to crank the track and master volumes to the max for it to be the same level as in ALSA (or windows).

I have the same problem when keeping ALSA and writing "default" in the device (should use pipewire-alsa) => low volume.

I have the same issue with Ardour and in Ubuntu Studio.

I tried tinkering with qpwgraph, the linux GUI for the Focusrite soundcards (https://github.com/geoffreybennett/alsa-scarlett-gui?tab=readme-ov-file), and some other stuff but to no avail.

Keep in mind I'm fairly new to Linux and may have overlooked something really basic. Thanks a lot for reading me and for any suggestion you might have.

I'm a casual Linux user. Been using linux on laptops in some flavor since 2012. But I've never really built a system for myself for any particular purpose before.

I'm finally getting around to building my PC for Linux a little bit of gaming(games made to run on Linux only / retro emulators) and amateur audio production.

Started buying parts for this 2 years ago so I don't remember the exact details but it's other wise a decently spec-ed machine for gaming, but I have since bought a PS5 and don't really use my PC for anything anymore.

Running an AMD Ryzen 7 without graphics. Large liquid cooled Radiator for low noise. MSI AMD 6800 GPU TB M.2 drive. 32 Gb of ram. 850watt PSU. (Might upgrade later)

I've been thinking about using Ubuntu Studio for the OS. And maybe dual booting Pop Os for non AAA gaming here and there.

Is your audio recording machine a gaming machine as well? Or do you have a dedicated machine for recording? Is it a hassle to game on the same system you game on?

Is Ubuntu Studio still a reasonable base to start an audio recording platform?

Is Ubuntu Studio decent for installing steam and other Linux only games. Or should I dual boot?

Hi! I wanted to switch to Ubuntu or Manjaro as my default operating system and then do a passthrough for Windows for software that requires it, like Live and iLok plugins. My main concern though was MIDI latency and I was wondering if there was any possible solution to overcome that problem, or if it would really just be better to dual boot for when I need windows programs.

​

I use Ableton very frequently although I'm decently familiar with Reaper and Linux suites, so it wouldn't be the end of the world but I would really like to keep my base OS as something other than Windows for everything else that I do.

Hi, for fear of spamming the sub, I have yet another question. I'm trying to create dual boot with my Windows11 install, and no matter what I do, the installer will not. See the empty partition to install to. It's a 500Gb disk, split down the middle. Windows is on one partition and Ubuntu was on th other. I've removed the Ubuntu partition. I created a new one, but the installer wouldn't "ungrey" the "next" button when selected. I've gone into Windows and formatted the partition from there as NTFS.

Both windows and Gparted can see the partition.

The installer wants to install side-by-side on the windows partition and the partition set aside for this install is not even an option.

On "manual install", the "next" button never ungrey no matter what partition is selected.

How do I get it to install side-by-side?

I made an Oxford Inflator clone with faust. It doesn't have a GUI for the moment but is fully functional. Comes in lv2 and vst.

You can check it out here: https://github.com/gaheldev/inflator/

Hi,

I am currently running a RME Babyface Pro (not FS) using Debian Sid 6.7.9 in class compliant mode and achieved absolute basic functionality using PulseAudio. As a native integration is missing and RME's propietary TotalFX does not work, I was wondering how to achieve similar functionality.

For a start I am looking how to achieve basic mixing capabilities. For instance taking the mic input and send it through some compressor (for instance Waves VST), several other digital outputs (i.e. Spotify) and mix them towards a single stereo output. Also, have a loopback device that picks up this mix and use it as a recording source. Think of it for smth like streaming (OBS) or recording a podcast.

I came across Jack however I believe that this is more like a virtual patch panel and requires 'other software i.e. OBS' to be compatible and have required plugins or patches for it in order to work.

Also, how to access sample rates, play with bandwidth and latency.

In the past I used to be able to achieve the mixing part running TotalFX while applying any sort of signal modification via VSTi's using Cubase. Certainly not the leanest way but it worked for me. Happy for suggestions in any direction.

Hi,

As the title suggests, this is a bit of a rant, but it's simply become unusable over the last couple of years.

Aside from the issue of weird tunings with midi instruments, now Ardour won't keep a midi recording. The moment you click the record button off, the whole track recording disappears - regardless of which midi instrument you've used.

I figured I'd upgrade to Ardour 8 (from 7.3), only the installer doesn't work. You get, "Install Complete"....and then open Ardour, and it's still 7.3, despite it finding and supposedly removing it and replacing it with version 8.

The Jack timeclock issue where everything is trying to steal being the master every 20 seconds is a real pain in the arse as well.

Circa UbuntuStudio 20.04 everything just worked out of the box...now it's a daily struggle to make anything resembling music, as by the time you've finished fighting, you've either forgotten what sparked the idea, or you're no longer in the mood due to all the fighting.

I've been using UbuntuStudio/Ardour/Jack et al for the last 15 years...this has to be the worst incarnation of it yet.

Hi all,

This is baffling me. If I use Carla and add an instrument, say Calf Organ or Helm...or pretty much anything other than Windows VSTs running in Wine, they're all nearly a semitone sharp.

...but not always.

After several reboots, and checking that all sample rates are the same in everything I can find (they are, they're all at 48), I added Yoshimi to Carla and played a "G" note.

It was perfectly in tune...and I thought, "at last...not sure what I've done to fix it, but it's fixed".

I walk away for 20 minutes because I was having an issue with bluetooth on my Katana Go (does anything actually work anymore?)...and I return to the keyboard.

Now it's nearly a semitone sharp.

I'm using an Aspire 5 15 laptop running Ubuntu Studio 23.04 with an M-Audo Axiom 61 controller - if this makes any difference.

Failing being able to fix this, what distros are people using? Ubuntu Studio has been going downhill for me for a while, and the last few years have been a bit of nightmare getting things up and running...and staying up and running.

I'm out of ideas.

EDIT: It seems that MIDI is just broken in Ubuntu Studio. I just recorded a midi track in Ardour using Yoshimi. Once finished, I went to resize the track....but the resize cursor didn't show. I unarmed the track...and boom. The whole recording vanished. I then touch the keyboard...and guess what? Everything is now nearly a semitone higher again.

After 15 years, I'm finally going to have to throw the towel in.

Hi All,
Just getting back into Linux so please forgive the noobery:

I seem to recall that it's possible to install the media-focused features of Ubuntu Studio into a completely different (Ubuntu-based) distro, maybe through Synaptic or the Software Center?

I'm setting up a Linux PC for my son and really like the user-friendliness of Zorin OS but wanted to hot-rod the audio/video performance under the hood, taking the apps and the realtime kernel from UbuntuStudio while keeping the overall UX of Zorin.
Does this sound right? I think that's what I did with Linux Mint years ago but don't remember how I did it....

Thanks in Advance

While the installation of the plugin was as easy as it could ever be thanks to Yabridge, I struggle to understand where I'm supposed to move the sample and present folders into in the case of my Arch Linux distro so that the plugin would find them when I launch it in Reaper 7.

Can someone please help me? Thank you!

Hello! I’ve been heavily debating on switching to Linux given that Microsoft wants to shove ads and AI in my face in every corner of their os. I’ve been a windows user my entire life and switching to another os is a little intimidating.

I use the following software for my music productions

Neural DSP plugins for amp sims (these use ilok) Native instruments Kontakt for drums and synths My daw is reaper.

I use the following hardware as well. Focusrite Solo interface Yamaha DGX660

How could I switch over to windows with the least amount of headache possible? What Distro would be best for my needs? Is the process for adding VST plugins the same as it is on windows?

Had to share this, I've been wanting something like this for ages.

https://github.com/xyproto/autodrums

I have a desktop running Ubuntu 22.04.4 LTS.

The front headphone jack only rarely works. If I unplug and replug my headphones, it never works right away. If I repeatedly switch the output device away from and then back to 'Front Headphones - USB Audio' then some fraction of the time sound starts going through the headphones. Once it starts working, it keeps working fine.

Possibly related: While the output device that (sometimes) works is "Front Headphones - USB Audio", the headphones use a standard 3.5mm jack.

The full set of available output devices is:

• HDMI / DisplayPort 2 - HDA NVidia
• HDMI / DisplayPort - HDA NVidia
• Speakers - USB Audio
• Front Headphones - USB Audio
• S/PDIF Out - USB Audio

Sometimes there's an extra option, but it's not showing up now. The extra option is for the line out in the back of the PC. Possibly it only shows up when I reboot with headphones plugged into the line out. The audio quality using headphones with line out is pretty bad, with a constant low crackling in the background.

This has been an issue since I bought my desktop over a year ago. Until today I thought it was a hardware issue b/c the front jack never worked. Then, earlier this morning I randomly plugged headphones in and was surprised that they were working. But on unplugging/replugging they stop working, and I need to go through the malarky described above to get them to work.

Looking for some help with my Focusrite 2i2 and Ubuntu. Audio from the 2i2 is very distorted by default. It appears to be the rate. I found this fix but was wondering if that will be permanent. Anyone else having issues with theirs?

pw-metadata -n settings 0 clock.force-rate 44100

I have a bunch of musics taken from "Place A" which are louder than the ones taken from "Place B", is there a program that I can make every music have the same volume, without distortions?

Edit: In this scenario the files aren't separated like "from Place A and from Place B", so I need to make every single file run through the software, so something like just "Rise the Volume" command won't work for me since it will also raise the ones that are already loud.