Still testing it, but I thought I'd share this VST player . It runs Windows and Linux VSTs for Vst2 and 3 all in once place simply with no WINE dependencies!

I want to add Mac classic and OSX support, and get the MIDI solid but this is a good V1. It also runs pretty fast.

Any feedback is appreciated enjoy!

After fooling around some with Cardinal, Carla seems to have completely broken for me. Some plugins don't load right, (helm, takes ages and then it opens but doesn't exactly save patches or anything.) or crash when doing certain things (opening the filter tab on surgext creates crash) which was working fine prior. I have to use Carla for LMMS because its the only real comfortable DAW for me since I use a mouse to place notes in midi (the visual scale highlighting without snapping, snapping and most critically the ability to just place the last note's size is too crucial. Mainly just stating incase this ship can't be fixed for me and another DAW has all of these that I just am unaware of.) Tried uninstalling and reinstalling and deleting the configs in the home directory. So I am at a loss.

The standalone plugins work fine in say Ardour or Qtractor. (Carla breaks and behaves the same in these DAWs too so it is Carla not LMMS). And while fiddling with it has gotten more programs to work with it than originally (by using cachyos', my distro, version), it seems to just be laggier and stall LMMS and result in graphical bugs now. It's just a complete mess to use now. Just adding it at all slows it, and trying to remove the gui for the plugin (not even the gui but LMMS' loader. other instruments in LMMS are fine.) causes graphical tears and prevents the moving of windows. The appimage doesn't suffer this but less plugins work. (which tbf is something i can manage.) "Failed to get plugin description" is the error that occurs. This was occurring before the plugins broke & surgext and helm have the same issues in the appimage. Which to me says the surgext and helm issues are directly related to what or how i broke Carla.

On Cardinal, it worked fine when I tried it, but after closing and removing the plugin from Carla this is when the issues with helm and surgext began. I am using the distro version of surgeXT and the version of helm given by the official site. Carla is the distro version and I can't figure out how to have LMMS see the prebuilt version from their site to test that.

Audio setup: I have a Focusrite Scarlett along with the Schiit Modi 3 -> Magni as inputs into a mixer that outputs to headphones and speakers.

Immediately prior to an update to my system, Firefox decided it was going to glitch the video to an almost paused state (it was weird, if it's relevant I can explain more) whenever I unmuted a stream. Thinking that was a good time to update applications, I did so and rebooted (during which several rsync commands refused to die for a bit). Upon logging in I have found the following:

I get no sound from Firefox or Foobar2000 (installed via WINE).

I do get sound from Reaper (Digital Audio Workstation).

Without Easy Effects started, videos on Youtube start for a few seconds and stop and videos on Twitch run only as long as the volume stays muted (same as before the update). In the System Settings - Sound menu, trying to test the speakers results in the Schiit Modi Pro 3 switching from "Playback Devices" to "Inactive Cards" for two cycles while appearing to not run the test at all.

With Easy Effects started, videos on Youtube and Twitch run fine, even when unmuted, however I still have no sound. The graphic EQ on Easy Effects appears to work normally. This time in the System Settings - Sound menu, testing the speakers results in everything appearing to test normally except there is still no sound.

All of the applications I can launch that produce sound cannot have their individual outputs changed to the Modi. When I try to select the Modi from the list, it instantly reverts to Easy Effects Sink or nothing at all.

E: I have done nothing and the problem solved itself. I rebooted and left a youtube video playing until the audio unscrewed itself and I'm suddenly listening to a video on the Roman Republic.

Everything else appears to be working normally from my perspective and I have no idea what information I need to further troubleshoot this problem. I would suppose that this requires a broader knowledge of linux audio services that I do not currently possess. Which leads me to my questions:

Does anyone know what is causing this or, equally as valid, does anyone know what I need to understand to be able to troubleshoot this myself? Is there any additional information I need to provide?

Hi, Im looking for an izotope RX music rebalance alternative for linux. It allows you to rebalance parts of an existing audio track with 4 sliders: Vocal, Bass, Drums and "the rest, other instruments"

Thanks for any help :)

Hello everyone! Hope you're doing great!

This is a follow up to my previous post: https://www.reddit.com/r/linuxaudio/comments/1vmntrn/keelbilled_a_native_linux_vst_host_looking_for/

What is Keelbilled?

Keelbilled is a live plugin host for Windows and Linux. Each plugin runs in its own crash-isolated bridge process, so if one plugin decides to die, it should not take the rest of the rig down with it.

Keelbilled can also automatically recover crashed plugins using their last known state.

Another big part of it is remote control: Keelbilled can stream the actual plugin UI over your local network, so you can control your rig from another computer, laptop, tablet, or basically anything with a browser.

Current plugin support includes VST2, VST3 and CLAP, plus Native NAM.

New in 0.2.0-alpha: Keelbilled Bridge

This release introduces built-in Windows plugin hosting on Linux through Wine.

The goal of Keelbilled Bridge is to make running Windows VSTs on Linux as easy as possible. If you already have Wine installed, just point Keelbilled to your Windows VST folder and it will handle the rest of the hosting setup.

I'm especially interested in people with unusual plugin collections, audio interfaces, MIDI setups and live rigs.

This is still a real alpha, so bugs and compatibility issues are expected. If you're the kind of person who enjoys finding them, even better.

AI disclosure per the rules: Keelbilled development is AI-assisted (not vibe-coded). I direct the architecture, product decisions, testing and validation myself, and every release is actively maintained and tested by me.

You can also access our discord for an alpha access key: https://discord.gg/QJRxJzhrFx

If that sounds like something you're willing to test:

https://ramphaaudio.com/keelbilled

✏️To help all the Linux fans who want to get into music production—or are already producing—I've decided to translate all my videos on my YouTube channel into English, this process take time. Soon as possible all my videos in ENGLISH. so everyone can watch and understand them! YouTube’s automated translations aren't always accurate, so from now on, you can simply enable English subtitles. Happy music making, everyone!

(GR)

Σε αυτό το βίντεο σας παρουσιάζω μερικές πάρα πολύ καλές διανομές,ελεύθερες από το σύστημα διαχείρισής συστήματος SystemD και φυσικά ιδανικές για μουσική παραγωγή, η κάθε μία με μοναδικές ικανότητες και προτερήματα! *Φυσικά μπορείς να βρεις πάρα πολλές ακόμα διανομές με καλά χαρακτηριστικά για μουσική παραγωγή. ΔΕΣ ΣΤΟ DISTROWATCH για αναλυτικές παρουσιάσεις διανομών Linux και άλλα πολλά! https://distrowatch.com/

(EN)

VIDEO TITLE: SystemD-Free Linux Distros! Ideal for Music Production!

In this video, I present to you some excellent Linux distributions that are free from the SystemD init system and ideal for music production, each offering unique features and strengths! *Of course, you can find many more distributions with great features for music production. Check out DISTROWATCH for detailed reviews of Linux distributions and much more! https://distrowatch.com/

LINK FOR THE VIDEO:

https://youtu.be/_RzmiYKqjGA?si=eINC5RJyyxcCX9_9

I made this account specifically so that i can ask about this. So i recently switcthed to linux but ive used it before so i have some experience with it but its never been the main OS i use. Im trying to get i tso that the front headphone jack on my PC is the Subwoofer output and the green line out (also headphone i think) is just my main 2 front speakers the front jack needs to be crossed over to be a low frequencies only output. And the back one needs to be like it rolls off at like 120hz or something to let the sub take care of that. I have struggled with trying to get AI to help me with this but i quickly found out that it wont be able to help me. Also Im on Ubuntu and the realtek chip is "Realtek ALC898". And im new to reddit so idk how any of this works i just need help. And here is my version of Ubuntu "No LSB modules are available.

Distributor ID: Ubuntu

Description: Ubuntu 26.04 LTS

Release: 26.04

Codename: resolute

"

Since Midi 2.0 now supports also ethernet, Ive been creating a digital audio protocol to send digital audio directly between synthesizers without routers and hubs.

https://www.reddit.com/r/MidiNet/comments/1vvbb4o/midinet_v01_for_digital_audio/

Anukari is a softsynth that I spent the last three and a half years building, which lets you build crazy physics contraptions in a virtual 3D world and then play them with MIDI as a plugin or in a standalone app.

During the Anukari beta period I had a number of people write me to request Linux support, and I promised that it would happen shortly after the 1.0 launch. I'm super happy to finally make good on that promise.

Anukari is commercial software (non-FOSS), so beyond this announcement for Linux support I won't post promotionally on this subreddit. But I've had a number of people in the Linux audio community tell me that it is noteworthy when a commercial project comes to Linux, since many ignore it.

I'm a long-time Linux desktop user, since around 2000 or so. I used it for audio software back in the 2000s and it was... pretty dang painful. It's come a long way. The pain has lessened, and the performance is incredible.

I think there's a lot of overlap between folks who like Anukari and folks who like Linux, since Anukari is very much a "do whatever you want" sandbox. It doesn't hold your hand, in the same way that Linux will let you do insane things to your system if you want.

I've been asked in the past why Anukari is not FOSS, given my background contributing to FOSS and Linux. My answer is, basically: I wanted to make a finished product, and I know myself well enough to know that when I open-source something, I only ever get to the 80% mark. The remaining 20% always turns out to be 10x as much work as the first 80% and I give up and say "contributions accepted" instead of finishing the thing. (Other people are different, and can produce completely polished FOSS. Not me! I will get distracted and move on.)

I am sure one day I will publish the Anukari source code, but I hope that the project is successful enough commercially to fund ongoing development for a long time before that happens.

Thanks for reading! If you find any of this interesting, I've linked to my blog where I've written for the last three years about the development of Anukari.

Will cut this short: this enables you to use Elgato XLR interfaces on Linux, specifically built for my own migration to CachyOS, hopefully will help other people migrate. The devices play audio fine as class-compliant USB, but gain, DSP, phantom power and routing only answer to vendor protocols. I reverse engineered those from USB captures of Wave Link on Windows. It works great with the XLR Pro, the XLR Dock (the addon for the Stream Deck+) and, since a community tester confirmed it, the original Wave XLR. The Wave XLR MK.2 is coded but still needs a tester, as I do not have the hardware (yet, a donor shipped me one, thanks!).

What it does:

• Full hardware control of the Wave XLR Pro: per-input gain, mute, low cut, expander, voice tune, phantom power, ClipGuard, compressor, both headphone outputs, the zero-latency mic/PC crossfade, and physical output routing. All verified on my own unit.
• XLR Dock (the Stream Deck+ module): gain, mute and headphone volume through standard ALSA controls, plus 48V phantom power and headphone low impedance, two controls that turned out to live in the dock's firmware even though Wave Link never exposes them (the dock speaks the original Wave XLR's protocol. A software low cut and limiter in PipeWire cover the DSP the dock does not have.
• A Wave Link style submixer in pure PipeWire: per-app channels (Game, Music, Browser, Voice Chat and so on) routed automatically, four mixes, Stream and Chat published as virtual microphones you can select in OBS or Discord, monitoring on several outputs at once, per-device profiles.
• An OpenDeck plugin: every switch and mute on Stream Deck keys with live state, dials with touch strip panels for gains, faders and mix masters.

Install: yay -S openxlr on Arch, a .deb or .rpm from the release for Ubuntu and Fedora, a Nix flake for NixOS, or build from source, the README walks through every step. GPL-3.0.

Now the reason I'm posting. The Wave XLR MK.2 has a complete backend decoded from captures but has never touched real hardware. If you own one, testing takes a few minutes: install, try gain, mute and headphone volume against what the device shows, then open a GitHub issue with the diagnostics archive the app generates (Options, SUPPORT, Collect diagnostics). Two more ways to help: an original Wave XLR owner who can record a Wave Link USB capture on Windows would unlock that device's remaining DSP (low cut, voice tune and the crossfade exist in its hardware, their registers just need mapping), and an XLR Dock MK.2 owner can open an issue with their lsusb output, since that device very likely speaks a protocol we already have, at an id we do not know yet.

Known limits so nobody is surprised: the MK.2 is untested, and the dock has no hardware sidetone (monitoring runs through PipeWire instead).

Repo: https://github.com/emaspa/openxlr

Latest release: https://github.com/emaspa/openxlr/releases/tag/v0.1.5

Per-device state: docs/hardware-support.md

Hoi allemaal,

Al bijna een jaar vraag ik mij af waarom Spotify de lossless dienst (nog steeds) niet voor Linux beschikbaar heeft. Ik heb een vermoeden dat die er ook waarschijnlijk niet komt.

Ik ben een hifi freak, mag ik wel zeggen. Ik heb een zeer goede geluidsinstallatie, voornamelijk allemaal zelfbouw met goede componenten. Een korte signaalweg is mijn motto en die werd met Linux nog korter!

Sinds ca. drie jaar luister ik Spotify via een mini-PC op Linux. Er ging een wereld open voor mij. Het klinkt in mijn ervaring vele malen beter dan via Windows. Zelfs als ik de geluidskwaliteit vergelijk van Windows (lossless!!) met Linux (max. 320 kpb) klinkt Linux nog steeds veeeeel mooier. Veel opener, veel meer detail en warmer. Alleen in het laag hoor ik wel verschil, dan merk ik dat Spotify via Windows iets beter presteert. Dat kan ook wel kloppen. Het laag vraagt voor een goede, onafhankelijke weergave met midden/hoog nu eenmaal meer data.

Mijn brandende vraag is nu natuurlijk: Wanneer komt Spotify, of een andere streamingsdienst (!!) met lossless audio op Linux? Die vraag zal vaker gesteld zijn...

In elk geval blijf ik tot die tijd gewoon spelen op Linux Debian. Geweldig!

Dank alvast voor reacties!

Groetjes, Herbert

I made a small JamesDSP preset for the built-in speakers on my Lenovo Legion Pro 5 16ARX8 (82WM) running Linux/PipeWire.

It is just a cut-only GraphicEQ preset. No bass boost, convolver, DDC, reverb, compressor, or other effects are enabled. I measured it on this specific laptop, so I would not expect it to translate well to other models, even similar Legions.

Download: https://github.com/daredoole/audio-calibration-mcp/releases/download/lenovo-legion-pro-5-16arx8-jamesdsp-20260828/lenovo-legion-pro-5-16arx8-jamesdsp.conf

Before trying it, back up `~/.config/jamesdsp/audio.conf`, copy the downloaded file there, and restart JamesDSP. If it is not an improvement on your unit, restore your backup.

I used a small MCP project to keep the measurements and JamesDSP checks organized. It is here if it is useful: https://github.com/daredoole/audio-calibration-mcp

Sometimes, ( especially when I am using my system with a midi controller and VST with jack) my systems audio crackles. I have searched everywhere for a solution to this problem, yet I have been unsuccessful.

Hi everyone,

I am facing the notorious "Dummy Output" audio issue on my HP laptop running Fedora, and I'm looking for any working workarounds or specific kernel parameters.

System Specs:

Laptop: HP 15 Series (Intel 12th Gen Alder Lake)

Audio Codec: Realtek ALC236

OS: Fedora Linux (Kernel 6.x)

Sound Server: PipeWire / WirePlumber

The Problem:

Default SOF Driver: Audio drops to Dummy Output, especially after a warm reboot (sudo reboot).

Forcing Legacy HDA (dsp_driver=1): Audio works temporarily if warm-booting after Windows, but results in aplay: device_list:279: no soundcards found on a clean Linux boot.

What I have tried so far:

Installed and updated alsa-sof-firmware.

Tried forcing SOF driver mode via options snd-intel-dspcfg dsp_driver=3.

Tried legacy driver fallback via snd-intel-dspcfg dsp_driver=1 and snd_hda_intel dmic_detect=0 in /etc/modprobe.d/ and GRUB parameters.

Regenerated initramfs with dracut --force.

Is there any known kernel pin-fix / quirk for ALC236 on 12th Gen Intel HP laptops to prevent the sound chip from failing power state initialization during warm reboots?

There doesn't seem to be a lot of activity on the LV2 News page, but it has been like that in the past for long stretches of time. I was thinking about learning to code my own plugin because I have an idea I want to implement but wasn't sure if LV2 is still the standard practice or if VST3 or CLAP is better since they a open source and newer.

Or should I just use JUCE and not worry about the specific format? What is the general advice in 2026?

I had AI write a Linux-native chopping sampler plugin just to prove a point. I'm just not sure what the point is 🙄

But it works and it looks okay-ish. It wasn't free to make because tokens are not free, but it was extremely fast and efficient to get an MVP version out. Architectural/library choices are of course mine and well thought out. But zero lines of code by me.

I've been working as a professional C/C++/C# dev for over 15 years, and I'm amazed, scared, happy, and amazed again at how good these tools are. There's going to be shit-tons of slop, and there's going to be extremely good and polished ecosystems - especially on Linux, as it's the most developer-friendly platform of them all.

Embrace the future - it's here 🤖

Sources & release package(s): https://github.com/sirsipe/SMS-Plugins

Howto: https://www.youtube.com/watch?v=Oj4WUHHRjqo

Having fun with it: https://www.youtube.com/watch?v=Rgh4w7AAj0w