it happens when i use the command "reaper" or "pw-jack reaper" (pipewire jack). im running it on a crappy laptop running cachyos. the kernel is 7.1.8-1-cachyos (64-bit), and i am using kde plasma wayland. the terminal featured in the images is alacritty. is there a setting i need to change or a flag i need to use? thank you for your time and consideration!
Hi everyone,
Recently installed Fedora as a dual boot on my 2015 15" Macbook Pro, after battling with Apple's limitations for a while to have a somewhat stable experience, most of it is great besides a few quirks. By the way I would not recommend dual booting a Macbook over another laptop if you could, the trackpad experience isn't even that great.
Anyway, the problem that subsists for me is the sound. I can confirm it is no hardware issue because when I boot on MacOS, the speakers are fine, but on Fedora, the speakers are awful, quality is bad overall and the high-mids sizzle like crazy. Tried many things over hours, even put an agent onto it, no success, when the sound EQ itself is modified, some musics sound good while others sound shallow and compressed.
Has anyone had the issue and fixed successfully?
SSL ships no Linux software for the SSL 12. Audio works out of the box (UAC2), but everything SSL 360 controls — internal mixer, monitoring, phantom power, loopback, routing — is unreachable. The card ends up a fixed-function box.
So I captured the USB traffic between SSL 360 and the card, mapped the control protocol, and wrote the tools:
https://github.com/xenicle/douze
- sslctl — CLI for the gain matrix, monitoring, preamps (48V/HPF/inst/polarity), loopback, headphone buses
- Douze — local web GUI (127.0.0.1) with the matrix, live meters, profiles - Douze FX — VST3 host that drops plugin chains into the PipeWire graph, one process per strip; a strip can publish its own virtual mic or sink, so any app picks it up as a normal device
The control interface is a separate vendor-specific USB device behind an internal hub, with no kernel driver — plain bulk endpoints, so everything is user-space (pyusb). No decompilation, no vendor code.
-> The protocol docs and all 25 captures are CC0 / public domain. A documented protocol is a fact about hardware, not a literary work — I'd rather it ended up in a kernel driver than stayed in my repo, so there's no attribution requirement at all. The code is AGPLv3 (JUCE).
-> Caveat worth stating up front: this is verified on exactly one card, mine, firmware bcdDevice 1.44. I use it daily and the protocol is mapped end to end, but I don't know if another unit behaves identically. If you own an SSL 12 I'd love to hear whether `sslctl status` returns something sane — that's the one thing I genuinely can't test alone.
Not affiliated with SSL; trademarks are theirs.
Bonjour à tous,
Étant gros consommateur de musique surtout depeche mode, j'aimerais savoir comment supporter du dolby atmos sur les distributions Linux.
Ainsi que l'installation d'une chaîne hifi des années 80 une kenwood en dolby prologic.
J'ai acheté des enceintes premium des Colonnes hi fi MORDAUNT SHORT MEZZO 8.
Je vous assure qu'avec sa l'écoute musicale est top.
Donc si quelqu'un peut me donner des informations du style quelle configuration pc pour faire tourner tout sa sur du linux.
N'importe quelle distributions, je me débrouille assez bien dans tout ce qui est tech, et j'ai sauté le pas de linux début 2026 avec zorin, Kubuntu, mint.
Voilà merci d'avance pour vos retours.
I'm a high school student who is developing a free plugin!
A few weeks ago you probably saw that I posted about COGG, a free plugin I built that metallizes and industrializes any sound. Specifically, harmonic distortion, metal plate convolution reverb, and pitch layering are there all in one chain.
I just shipped v1.1 with 8 factory presets so you can dial in different metallic textures instantly instead of tweaking all the knobs from scratch. Everything is there, from subtle texture (like the preset RUST), to full chaos (like presets SHRAPNEL and MELTDOWN).
Still free, no account needed, VST3 + AU for Mac, VST3 for Windows.
Download now for free: https://plasma-hype.itch.io/cogg
Trailer/demo of the update: https://www.youtube.com/watch?v=HfD2drnrMjQ
Thanks to everyone who tried v1.0 and gave feedback, this update came directly from figuring out what people actually wanted to do with it. I personally have used COGG in almost all of my FL projects ever since I released it. Please give it a shot and let me know what you think of the presets and plugin as a whole for a v1.2!
So I managed to have izotope plugin working with wine. I'll try my best to explain how I did it.
I'm on fedora 44 using bitwig.
As you may know izotope product portal doesn't work with wine because of incompatibilities with key encryption or something BUT you can still authorize your products if you run native acces (I recommend doing it fast because izotope was sold from NI so it may not stick around for long).
You must install wine 9.21 (important otherwise the Ui won't work)
After installing wine you must install native access 2 but it won't work if you install directly. I used this script to install it : https://github.com/jefrecantuledesma/Native-Access-Linux-Compatibility/tree/main
If you don't NA will tell you its already running. Then if you manage to fix this it will tell you to grand permision to install dependencies. You also have to modify the username to make the script work.
Now for the part I have trouble to explain, the script didn't entirely do it for me so I tried installing NTKDaemon : https://drive.google.com/file/d/1kLTmdkVzN4VyUICrfLNxYRIrERqqRR7C/view
You can install it like any normal wine program but this still didn't work. So out of desperation I searched in my c_drive every NTKdaemon_setup.exe and launched them all one by one and I don't know why but this worked, I could then login and install my plugins.
If NA gives you an error when downloading try changing your download location. Then setup yabridge and you should be able to insert your vsts.
If the UI still reacts to audio but you can't click the buttons and move the parameters, use wine 9.21. If you're on wine 9.21 and having this issue I'm sorry but I can't help you. I still encourage you to check out yabridge github for help
Hope this helps some1one !
Edit : Whether you use X11 or wayland the UI should work either way
long story short, asrock suks, got a new motherboard, waiting on amd cpu rma. however my asrock had a alc1200/1220 and this msi has a alc897. going back in a day, it sounds like a piece of foam or cardboard is in front of my speakers. most motherboards suks, i was lucky to get a mediatek wifi chip cuz i didn't really have any choice...anyways...
it looks like soundcards are dead and everyone uses usb =/ i really dont want yet another thing on my desk though. all i can find are used asus xonar cards. is there anything just for generic use anyone would recommend? all i see are $20 small pci cards on newegg and creative which as i know, cares nothing for linux (my soundbar is creative, or i'd just use that over usb i guess).
only other things ive seen are small dongles that im skeptical on the usefulness off, especially for the price.
thanks for any info.
I'm looking for a FOSS application that I can use with my toddler that allows them to easily switch and play with sounds, and possibly interacts with the LED pads on devices like the Novation Lauchpad Mini and LaunchKey Mini.
My hope is to find something that's easy to explore and muck around with for them.
I realise this might not be available but things like easy to use touch screen icons would be helpful.
I'm also open to Reaper Plugins if there's any suggestions
i don't need much at all, all i'd use it for is making rough demos for my band, any recommendations?
To summarize this:
I am using bottle to try install native access + spitfire audio + fl studio as it is what I frequently used as DAW before transfered into linux a while ago.
I got fl studio to work but for both native instrumetn and spitfire audio. I couldn't get it to work. I am clueless on how to do continue doing this and need help
The symptom I see when trying to install both.
for native access: It start installing but get stopped by a small window that pop up saying "Native instrument is running click okay to close it, if it doesn't close, try closing it manually."
clicking cancel would just stop the process and clicking okay telling me that it cannot be close.
for spitfire audio: I get it to finish downloading but when launch the program. It just give me a window with blank white screen.
I just installed bottle a few days ago and haven't done much beside download some dependencies and use what I might use to run fl studio.
So, I haven't try using environment variables, DLL overrides, snapshot or any other feature that could possibly be the solution.
Picture for illustration and my system information will be attaches so you can look into it if it helps by any means.
Most voice input on Android - SpeechRecognizer.createSpeechRecognizer(context) calls — gets routed to Google's network-backed recognizer. I wanted that path to run locally, so I wrote one.
The service hooks the framework's SpeechRecognizer API. Once it's set as the default, any app calling createSpeechRecognizer(context) (no ComponentName) ends up in our pipeline and gets back transcription that never left the device. Pipeline is Silero VAD + Parakeet TDT v3 (114 languages, ~890 MB INT8) on ONNX Runtime with NNAPI.
Honest caveat: Gboard, Samsung Keyboard, and Google Assistant ship their own recognizers and skip the system default. So the default-IME voice button on most phones won't go through this. What does: accessibility tools, custom dictation UIs, and anything calling the framework API directly.
Models download on first use (~1.2 GB) via a foreground WorkManager job so it survives backgrounding. After that, fully offline.
Setup + demo APK: github.com/soniqo/speech-android
audio.soniqo:speech:0.0.9 on Maven Central
Library:
Happy to answer questions about the binder lifecycle, the foreground worker setup, or why SpeechRecognizer is such a tarpit of edge cases.
Hey,
I've been working on RustMusic for the past few months. It's a free and open source HD music player written in Rust.
I mainly started it because I wanted a player that takes audio quality seriously while treating Linux as a proper platform, not just another build target.
Version 0.2.0 is now available and I'd really like to get some feedback from Linux users and audiophiles.
🎯 Bit-perfect audio
On Linux, RustMusic can use ALSA exclusive mode with D-Bus device reservation, allowing PipeWire to properly release the audio device when needed.
There is also WASAPI Exclusive support on Windows and CoreAudio hog mode on macOS.
🎧 DSD support
RustMusic can play DSF and DFF files and send DSD64 up to DSD256 using DoP passthrough to a compatible DAC.
If your DAC doesn't support DoP, RustMusic can convert DSD to PCM instead.
🏷️ Tag editor
You can edit metadata for MP3, FLAC, DSF and DFF files, either individually or in batch.
There's also a spreadsheet-style editor for larger libraries and automatic metadata and cover import using Deezer's public API.
📡 Built-in DLNA / UPnP server
You can share your music library directly with compatible amps, streamers and other devices without having to configure a separate DLNA server.
🎚️ ReplayGain
Track and album modes are supported, with an adjustable preamp.
There are also a few other features like MPRIS support, sleep timer, mini-player, always-on-top mode, multiple profiles and a high contrast accessibility mode.
Linux builds are available as .deb, .rpm and .AppImage.
I also provide a glibc 2.35 compatible build for Debian 12, Ubuntu 22.04, Linux Mint and similar systems.
RustMusic is GPL-3.0. No account, no ads, no telemetry.
What I'm mainly looking for now is feedback from people using different Linux setups.
If you try it and something breaks, tell me. If something feels wrong or badly designed, tell me that too. That's much more useful to me than just hearing that everything looks good.
I'm especially interested in feedback from people using external DACs, PipeWire, DoP, Arch, Fedora, NixOS or SteamOS.
🔗 Site: https://rustmusic.dev
🔗 Downloads: https://rustmusic.dev/downloads
🔗 Source / issues: https://github.com/larevuegeek/rustmusic
🔗 Changelog: https://rustmusic.dev/changelog
If you have any questions about how it works, feel free to ask. I'm the developer, so I'll do my best to answer., alternative to Foobar2000 / Audirvāna / Roon
On Nobara 44, my laptop keeps recognising other people's MacBooks as audio devices. There doesn't seem to be anything coming through them, but it makes using certain apps like Discord very difficult. Screenshot of QjackCtl: https://pixeldrain.com/u/6Lqt5vi8
Any ideas where this could be coming from and how to stop it happening?
If you want to participate in the testing of the new winepipewire.drv with or without spatial support, then head out to official CachyOS discord guild: https://discord.com/channels/862292009423470592/1514956087550541924
After the new installation of fedora kde 44 does anyone have any issue with the sound of their laptop being very low even in 100%? Or any issue with the previous config of easy effect when updated to Fedora 43.
ToneShiftEQ is a modern 12-band parametric equalizer with APO support, designed for transparent tone shaping, corrective equalization, mixing and mastering. Each band can also operate as an independent dynamic equalizer with configurable Threshold and Ratio controls.
Available as:
New in this release:
Release Page:
https://github.com/brummer10/ToneShiftEQ/releases/tag/v0.8.0
Project Page:
I posted my chorus plugin here a couple weeks ago and someone told me allpass interpolation was the wrong tool for a modulated delay, since the recursive state goes stale while the delay length is moving under it, and that I should look at Lagrange. I was still on linear at that point. So I wrote a 4-tap cubic Lagrange interpolator by hand instead of pulling in a crate, mostly because I wanted to understand it rather than trust it.
What's actually in the thing now, so nobody has to guess from the post:
- 6 ms base delay with the LFO modulating around it, cubic Lagrange on the fractional read. Taps at -1, 0, 1, 2 off the integer index, each one wrapping the ring buffer separately.
- One-pole highpass at 160 Hz sitting inside the feedback loop so the low end doesn't stack up.
- One-pole lowpass on the wet path, cutoff swept by the LFO between 1k and 7k. Most of the character comes from that, not the delay.
- Separate voice struct per channel, LFOs offset 90 degrees. Collapses to mono without eating itself.
- Dry/wet is a plain linear crossfade. There's roughly a 3 dB dip at 50% and I'm fairly sure that's comb cancellation from summing a correlated wet signal with the dry, not bad crossfade math. If that reasoning is wrong I'd like to know.
One debugging note that saved me: check your coefficients at frac = 0.5. They should land on -0.0625, 0.5625, 0.5625, -0.0625. Mine didn't, and it was a typo in one of the products in the third coefficient.
At my defaults (1.6 Hz, about 4 ms depth) the improvement is small. A little less grit on held notes up top. I assume it opens up more if you push the rate.
Two things I still can't answer:
It's free, GPLv3, VST3 and CLAP, mac/windows/linux. It's built for pitch-corrected vocals specifically, which is the only thing I use it on. If you'd rather hear it than read about it, it's at pyfessional.tech, the download button picks your OS for you and the source is linked from the same page. Would rather have someone install it and tell me it sounds wrong than get upvotes.
Hello everyone! Hope you're doing great!
I've been working really hard on Keelbilled, a live plugin host for Linux and Windows.
It supports native Linux plugins as well as Windows CLAP, VST2 and VST3 plugins. LV2 isn't supported yet, but it's on the roadmap.
Each plugin runs in its own crash-isolated bridge process, so if one plugin decides to die mid-performance, it doesn't take the rest of your rig down with it.
Automatic plugin recovery is also available, so a crashed plugin can be respawned with its last known state.
Keelbilled can also stream the actual plugin UI to other devices on your LAN. So you can control your plugins from another computer, a laptop, a tablet... basically, if it has a screen, you can probably use it.
I've tested it on the Linux distros and machines I have access to, but now I need people who can go far beyond my own testing setup: different distros, audio interfaces, plugin collections, hardware...
And, most importantly, a strong will to make things break. haha
This is a real alpha, so bugs and compatibility issues are expected. That's exactly what I'm looking for.
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.
EDIT: Sharing the discord link here! https://discord.gg/QJRxJzhrFx
Hi Everybody,
I've been casually working on a DAW using Cursor. My original goal was to create an Ableton Live like DAW that would be cross platform and a feasible option for Linux users. I've since started a few more projects and this one is seeing some neglect. I figured I'd open up the GitHub and see if anyone else wants to contribute.
I wasn't aiming to make any money from it, just offer a worthy, free to use, Linux compatible DAW.
https://github.com/mark-speer/SkeletonHive
https://skeletonhive.carrd.co/
Anyone is welcome to download and compile it for use or clone and tinker.
Feel free to reject or delete this post if I'm breaking any rules.
Edit: It's now officially Open Source.
This post is about showcasing 0.4.0 as 0.5.0 is a naming conflict resolution release (thanks to Pallaswept on GitHub for giving me a heads up on it), which took me awhile and AUR being down kind of helped. Its all in the release notes. The app offers a-lot but in this post I am highlighting only the new added page- Signal Paths
Quick context if you haven't run into the project: PipeWire Controller is a GTK4/libadwaita app for managing your pipewire/audio setup. it includes audio management, signal paths, virtual devices, live patchbay, parametric equalizer, performance monitoring, filter chains, microphone cleanup/effects (echo/noise), HRIR virtual surround, routing snapshots, per-app policies, LADSPA/LV2 effect inserts and more.
Version 0.4.0 adds a new page called Signal Paths- The idea is simple. A **source** is where sound comes in — one app, a microphone, or everything on your default output — and it carries its own chain of processing. A **mix** carries a chain of its own and feeds real devices. Sources on the left, mixes on the right, and the sends between them drawn as curves so you can see what's going where
One source and one mix is just a straight line, which is what most setups are.The second column only earns its place once a chain has to split — say one source corrected four different ways for four different pieces of hardware, without building it four times.
**You don't have to build it from scratch.** An empty board *is* a template catalog — four complete paths that build both halves in one click, then 26 ready-made strips going from a plain speaker mix at the top to the stuff people actually run for broadcast at the bottom. Bass boost, a loudness curve for quiet listening, crossfeed for headphones, a turntable chain, the gate → tone → compressor → limiter voice chain, a mastering bus. Each card draws the chain it's about to build, and if a template wants a plugin you don't have it says so and leaves it out instead of handing you a strip that won't start.
Create new Signal Path - Overview Screen
Template Browser (Color coded to sources/mixes)
Per card - add effects, plugins, convolver files and drag them in any order
**One process per chain, not one per plugin.** Every stage in a strip gets compiled into a single filter graph. Twenty effects is one entry in your device list and one buffer hop, instead of twenty of each. If you've ever wired this up by hand you know it goes the other way and you end up eating twenty quanta of latency for it.
The screenshot below shows 6 sources, 5 mixes, a plugin chain on every strip, one 7.1 pair and a two-device output
6 sources, 5 mixes, a plugin chain on every strip, one 7.1 pair and a two-device output
- **EQ bands move while the audio plays.** These are built from biquad filters rather than the preset-file kind, so frequency, gain and Q take effect as you drag them. No restart, no gap.
- **The board is handled directly.** Drag a stage along its chain to reorder it, or onto another card to move it there. Drag a card onto the opposite column to connect the two. Drag an app between strips to move what it's playing through. Click a stage to take it in or out of the signal.
Drag and reorder sources/mixes
Again this is just highlighting the new page Signal Paths, which took a-lot of work. Here is a screenshot of the dashboard (which I'm currently working on revamping to match efficiency)
There's alot more the app offers and I'd love for people to come check it out. I have fixed the bugs/issues I've been receiving but would like more feedback. I will add a copy pasta from the Readme.md below. thanks to all the people who've been helping.
- GitHub: https://github.com/knightinfected/PipeWireController
- AUR: `paru -S pipewire-control-center` (or `yay`)
- Full changelog with screenshots: https://github.com/knightinfected/PipeWireController/blob/main/CHANGELOG.md
Not on Arch? There are install instructions for Fedora, Ubuntu/Debian and
others in the README — it's pure Python, no build step.
**What I'd like from you**
Honestly, feedback. I use this daily on my own setup, which means I mostly find
the bugs *I* happen to walk into, and every PipeWire setup is wildly different
from every other one. If you try it and something is broken or confusing or
just annoying, open an issue — a one-liner helps more than you'd think. Half of
this release exists because two people took the time.
