I was tired of choosing which app I could hear.

A lecture in one window, a friend on a call, music in the background. On Linux, all of it came out at the same volume. Something always had to be muted, and I was always missing something.

So I built my own mixer.

Fonyx works like a mixing desk for every app on your computer. Each app goes into a group, and each group has its own fader. The call stays clear, the music sits quietly underneath, and nothing has to be switched off.

🎚️ Mixing.

Every app's audio goes through Fonyx, so you balance them all from one place, with EQ, a compressor and a gate on each group.

🔉 Ducking.

When someone starts talking on a call, the music lowers itself, and it comes back up when they stop. No reaching for the volume.

Once the core worked, I kept going:

🔀 Routing.

Send any group to any device, or to several at once. The call goes to your headphones, the music to your speakers, and you can preview something in your headphones without anyone else hearing it.

🧩 Plugins.

Built-in effects you can stack on any group: deep bass, a stereo widener, a mids enhancer and a soft clipper.

It also has scenes you can switch with one click, rules that put each app in the right group as soon as it starts, and recording for any group.

I wrote it in Rust on top of PipeWire. The hardest bug was Linux killing the audio engine without any error message. The cause was tiny numbers called denormals, which slowed the filters down every time a song went quiet. The fix was one CPU flag; finding it took days.

Fonyx is free and open source. The AppImage is available now, and the Snap Store release is coming soon.

GitHub: https://github.com/ArdenDiago/Fonyx

Website: https://fonyx.ardend.dev/

If your audio has ever made you pick between two things you wanted to hear, give it a try and tell me what's missing.

#Linux #OpenSource #Rust #PipeWire #AudioEngineering #BuildInPublic

While building it I kept losing the center, sub and rear right speakers after a reboot, sleep or driver reload. The card was receiving the audio, but those channels came out silent. Most restarts lost at least one of them, and switching the PC off at the power supply sometimes "fixed" it.

It turned out to be random on every start.

The cause: the card's sound processor only applies the centre/LFE and surround output settings when they change, and at start-up it sometimes misses them. The Linux driver never re-sends a setting that hasn't changed, so those channels stay silent.

The app now switches those outputs off and on once, a couple of seconds after the card starts. Every restart I tested since has come up with all six channels.

There's also an optional patch for the kernel driver (Linux 6.17+) that does the same inside the driver. It also fixes the bass redirection crossover: the stock driver sets it 10 Hz too high, and it overwrites it with the X-Bass setting whenever the speaker setup changes.

What the app does:

- Creative's SBX effects and all ten of their profiles, with the same values as the Windows app

- 10-band EQ with Creative's presets

- Room view of your speakers with level meters; click a speaker to test it

- What's playing and the path its sound takes through PipeWire and the card

- Pure mode, 5.1 or 2.1, and stereo on the fronts, fronts + rears, or all speakers

- CrystalVoice mic effects, per-game profiles, tray icon

Tested on one ZxR with CachyOS, KDE Plasma and kernel 7.3. Other CA0132 cards (Sound Blaster Z, AE-5, AE-7, Recon3D) are untested. If you have one, I'd like to hear how it goes.

I built it with a lot of help from an AI coding assistant (Claude), and the commits say so.

GitHub: https://github.com/ImZoomBoy/soundblaster-command-linux

Not affiliated with Creative.

Looking for some help with this.

I have installed yabridge and a couple of plugins:

Oracle Gorestager

RiftFeedback Lite

After installing the plugins and syncing them, thus creating the .so wrapper I then clear cache/re-scan in REAPER. The plugins show as expected. However, when I click to add them, the add FX window just hangs and REAPER freezes and I need to kill the process.

I am running wine-staging 9.21, have tried it in an x11 environment, have tried grouping via a .toml file, done a wine regedit and changed Graphics entry to Wayland. Also tried enabling virtual desktop environment and running the plugin as a separate/dedicated process and running it with a generic GUI.

When I run REAPER from the terminal and try to add Gorestager, I get error messages to the effect of "Wine STDERR" (I'm not at my machine right now and can't remember the exact wording and can't currently paste in the log).

None of the things I have tried work at all and I don't want to keep fumbling around and cause more issues. Any suggestions or fixes/things I haven't tried would be welcomed.

Thanks in advance.

Here's what I've got working so far:

ALAC Lossless / Hi-Res (external helper required)

Direct ALSA output (hw:X,Y), selectable in the GUI

CLI bit-perfect checks

GUI audio format display (e.g. ALAC 24-bit / 96 kHz)

AAC 256 kbps, PipeWire/PulseAudio, MPRIS, lyrics and playlists

Built with Rust and egui (no Electron).

Still a work in progress. Feedback, bug reports and contributions are welcome!

GitHub: https://github.com/nekomario28/melba

Built with extensive help from Codex and DeepSeek.

I've been building this for a while and it's finally at the point where other people can use it: a DAW that runs as one binary, links only glibc and libasound, and installs as .deb, .rpm, AppImage or tarball on anything with glibc ≥ 2.28.

FL-Studio-style workflow rather than a clone: channel rack with drums and synths, piano roll, mixer with buses/sends, VST3/CLAP/VST2 hosting, offline export. 88 of FL's 135 stock plugins have a full equivalent, 47 partial, none missing — that inventory is a file in the repo, counted by a script, not a claim in a README.

The part people tend to react to: type "give me a trap beat at 140 BPM with an 808 bass" into the assistant and it sets the tempo, picks drum samples from your own library by role and style, and writes kick/snare/hat lanes plus the 808 line. It needs your own API key (DeepSeek, OpenAI or Anthropic) — no cloud account of mine in the loop.

Install: grab the AppImage from the release and run it, or sudo apt install ./sonix_*.deb.

Two honest limits, so nobody wastes an evening: the step grid is 16ths, so trap hi-hat triplets aren't possible yet, and the aarch64 builds are green in CI but I haven't run them on real ARM hardware. Stem separation is an opt-in build feature.

GPL-3.0. Contributions welcome — there are three issues labelled for newcomers, and I answer everything.

Repo: https://github.com/alexwest1981/sonix

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I'm trying to move from using Windows to Linux for playing guitar. I'm using Neural DSP Nolly X's app (not through a DAW) through Wine with PipeASIO. I have the buffer size set to 64, which works just fine on Windows without any drops. On linux, most of the time its fine (the DSP load is ~10%) but occasionally spikes to 100% for an instant and I get xruns.

I put my user in the pipewire group which did help, as it bypasses the rtkit max priority of 20 and as you can see below pipewire runs at -89 priority, but it's still not perfect. Any other advice?

    PID USER       PRI△ NI  VIRT   RES   SHR S  CPU% MEM%   TIME+  Command
   2576 rtkit       RT   1 21872  3652     0 S   0.0  0.0  0:00.00 /usr/libexec/rtkit-daemon
   5011 user       -89 -11 46232 19828     0 S   0.0  0.1  0:02.76 /usr/bin/pipewire
   5065 user       -84 -11  111M 20940     0 S   0.0  0.1  0:00.03 /usr/bin/pipewire-pulse
   5090 user       -84 -11  550M 39268     0 S   0.0  0.1  0:00.00 /usr/bin/wireplumber
  10762 user       -84  -2 3452M  293M     0 S   0.0  1.0  0:02.74 C:\Program Files\Neural DSP\Archetype Nolly X\Archetype Nolly X.exe    PID USER       PRI△ NI  VIRT   RES   SHR S  CPU% MEM%   TIME+  Command
   2576 rtkit       RT   1 21872  3652     0 S   0.0  0.0  0:00.00 /usr/libexec/rtkit-daemon
   5011 user       -89 -11 46232 19828     0 S   0.0  0.1  0:02.76 /usr/bin/pipewire
   5065 user       -84 -11  111M 20940     0 S   0.0  0.1  0:00.03 /usr/bin/pipewire-pulse
   5090 user       -84 -11  550M 39268     0 S   0.0  0.1  0:00.00 /usr/bin/wireplumber
  10762 user       -84  -2 3452M  293M     0 S   0.0  1.0  0:02.74 C:\Program Files\Neural DSP\Archetype Nolly X\Archetype Nolly X.exe

pw-top output

S   ID  QUANT   RATE    WAIT    BUSY   W/Q   B/Q  ERR FORMAT           NAME                                                                                              
I   30      0      0   0.0us   0.0us  ???   ???     0                  Dummy-Driver
S   31      0      0    ---     ---   ---   ---     0                  Freewheel-Driver
S   53      0      0    ---     ---   ---   ---     0                  Midi-Bridge
S   64      0      0    ---     ---   ---   ---     0                  bluez_midi.server
S   67      0      0    ---     ---   ---   ---     0                  alsa_output.pci-0000_01_00.1.hdmi-stereo
S   37      0      0    ---     ---   ---   ---     0                  alsa_input.usb-JOUNIVO_JOUNIVO_JV601P_20180508-00.analog-stereo
R   46     64  48000 116.0us   0.2us  0.09  0.00   11    S32LE 4 48000 alsa_input.usb-Focusrite_Scarlett_Solo_4th_Gen_S12D5M456AD3D6-00.pro-input-0
R   47      0      0   2.2us   0.9us  0.00  0.00    0    S32LE 2 48000  + alsa_output.usb-Focusrite_Scarlett_Solo_4th_Gen_S12D5M456AD3D6-00.pro-output-0
R  100     64  48000   2.5us 110.2us  0.00  0.08   18     F32P 2 48000  + Archetype Nolly X
S   45      0      0    ---     ---   ---   ---     0                  alsa_input.usb-046d_HD_Pro_Webcam_C920_FF11D6DF-02.analog-stereo
S   86      0      0    ---     ---   ---   ---     0                  v4l2_input.pci-0000_13_00.4-usb-0_1.4.2_1.0

System:

• Ubuntu 26.04 LTS
• kernel 7.0.0-38
• Ryzen 9950X
• RTX 5080
• 32 GB RAM

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Hi everyone,

I'm having a frustrating Bluetooth audio problem on MX Linux XFCE and would appreciate some help.

My OPPO Enco M32 Bluetooth earbuds were working fine yesterday. I shut down/restarted my laptop, and now the audio is extremely choppy and distorted. Sometimes it sounds like a robotic or alien voice. The earbuds work perfectly with other devices.

My system:

• OS: MX Linux XFCE
• Kernel: 6.12.107+deb13-amd64
• Bluetooth adapter: Realtek USB Bluetooth Radio
• USB ID: 0bda:b001
• Bluetooth stack: BlueZ 5.82
• Audio server: PulseAudio
• Init system: SysVinit

Troubleshooting I've already tried:

• Restarted the Bluetooth service using sudo service bluetooth restart
• Disconnected and reconnected the earbuds
• Tested with Wi-Fi completely turned off
• Tried disabling USB autosuspend for the Bluetooth adapter temporarily
• Checked that the Bluetooth adapter is powered on and not blocked
• Checked the PulseAudio Bluetooth card and its available A2DP and hands-free profiles

None of these steps fixed the problem.

Relevant errors from the logs:

• Bluetooth: hci0: corrupted SCO packet
• SCO packet for unknown connection handle
• urb ... submission failed (90)
• sending frame failed (-90)
• Repeated mSBC decoding error messages

My Bluetooth adapter appears to be working, but the audio remains choppy. I only have kernel 6.12.107 installed at the moment.

My questions:

1. Could this be a kernel regression or a Realtek Bluetooth USB driver issue?
2. Is there a known fix for USB ID 0bda:b001 on kernel 6.12?
3. Would testing a newer Debian kernel (such as 6.18) be a sensible next step?
4. Are there any additional logs or diagnostic commands I should provide?

PLEASE HELP WITH THIS I AM TRYING TO TROBLESHOOT THIS FOR OVER 2 HRS!!, SHOULD I SWITCH TH KERNEL, AND IF SO DO I WILL LOSE MY DATA AND SETTINGS

We've all been there: buffer's too small, xrun hits, click or dead air. This library tries something different — instead of silence, it synthesizes a plausible continuation of the signal from a running spectral analysis of recent audio, and splices it in.

Short-time Fourier analysis runs continuously; on a miss, each bin continues by extrapolating phase/amplitude, or — for bins with two sinusoidal partials — a fitted two-pole AR model (generalizes the classic phase-vocoder freeze beyond the single-partial case). Two seam-repair strategies, picked per-occurrence by spectral similarity. There's a feedback-runaway failure mode when concealed audio gets re-analyzed; a damping term kills it.

Built it in a GrandOrgue fork, and the demo build actually lets you force xruns at a controllable rate during real playback and A/B the raw dropout against several concealment variants live — genuinely the easiest way to judge whether this is snake oil or not. Highly experimental, hasn't had broad real-world mileage yet. Technical disclosure with measured NMR results: https://github.com/sebion7125/xorcist

GPLv3-or-later, standalone C library with an OpenCL-optional per-bin solve.

EDIT:
I've made a demo page where you can listen to the difference Xorcist makes at a 25 % gapped signal:
https://sebion7125.github.io/xorcist/listen.html

The moment we started this project, we already had a vision of what we wanted from it: a multi-instrumental pedal that could run any kind of audio plugins. Dream on...

Sounded like a colossal task, making something like this from scratch required so many different fields of knowledge together. In 2023, after a lot of meetings and snooping around, we consolidated a small and balanced team of nerd musicians. 3 Engineers (Electronic, software and even an industrial one) and a UI/UX/Product designer (yours truly). Later the team would change and grow.

v0.0 Baby Steps:

We soon had our hands in our first soundboard design. Combining it with a Raspberry CM4 we had ourselves Prototype Zero (those pictures were not exactly glamorous, and we had to leave some of them out anyways). As you can imagine, it had a lot of problems. Some of them turned to be major challenges for us.

v1.0: A Small Victory:

It was working! Now it had touchscreen, XLR/jack connections and the ability to run Linux compatible plugin. From here path ahead seemed clearer. A solid proof of concept.

v2.0: A Model Body

Finally we now had an enclosure to keep it safe and stylish. Now we had a USB input to use MIDI virtual instruments. It also included LED rings around the footswitch and a mini-card for real analog distortion.

v2.4: The NAM Fam

It was around here when NAM caught our attention. We were looking at a genuinely groundbreaking field of audio technology, with an enthusiastic community. It turned out to be extremely fertile soil for us. We got so much feedback and ideas from this community and suddenly WizeCore was being shaped after understanding the needs of those who'll use it. Version 2.4 had our own native plugin that would browse and download NAMs from Tone3000 API.

v3.0: Gettin' Serious

We figured out from the community feedback and our own tests that if we wanted to really serve the purpose of WizeCore's existence, we needed better input/output options, more footswitches, larger screen, etc...

We adapted the software, made new optimized boards that solved many of the technical challenges that were tormenting us since Proto Zero. Here it started feeling like the real deal.

v3.1: Breaking the Cage

IT HAPPENED! We managed to solve our deepest existential problem. We finally made it possible to run Windows VSTs on an ARM-based processor running Linux.. That was the last piece of the puzzle. Our Holy Grail. We could finally make FX chains using the VSTs we have always been using on our DAWs... in a pedal... in WizeCore.

v4.0: The Final Form

WizeCore reaches the stage of being ready to see the outer world. The Crowdfunding edition comes with so many improvements such as:

- added a 4th footswitch (since many said 3 or fewer was a dealbreaker)

- a wider screen

- a single-board design that allows the device to be much thinner

- rotary footswitches with ringlights are back

- and so many other changes. You can check the full specs at our website.

WizeCore could improve a lot in terms of software development and user experience due to the fact we were developing in parallel WizeHost: our own VST or standalone app for your computer. WizeHost and WizeCore are designed to be almost 1:1 in features and compatibility. It will soon be on early access, available free for everyone.

Crowdfunding soon:

Very soon we will make available WizeCore pre-orders via Kickstarter. Thanks to everybody who shared their interest and gave us so valuable feedback. You're all awesome ;)

Tell me what would you do with a pedal like that. I'm sure there are plenty of use-cases that never even crossed my mind.

Also it's quite a challenge to verbally define what WizeCore is. It's like a multieffect pedal of course, but it also does audio plugins, midi virtual instruments, sequencers... damn, it could even run GTA San Andreas. It has also a DAW aspect to it, where you can record separate track and mix them...

Would you think its fair to call it a PAW (Portable Audio Workstation)?

*title says v1.3, i meant v1.4, sorry

Update on my FREE plugin COGG: I added the PRE-METAL toggle and 8 new PRESETS in v1.3!

I'm a high school student who is developing a free plugin!!!

A few weeks ago you probably saw that I posted about COGG v1/v1.1/v1.2/v1.3, a free plugin I built that metalizes and industrializes any sound. Specifically, harmonic distortion, metal plate convolution reverb, and pitch layering are there all in one chain. Great for experimental music and mainstream alike.

I wanted to update you guys with version 1.4's release! It comes with the highly requested PRE toggle, which allows the metal effect to become distorted along with the rest of the sound, rather than just acting as a “topping!” Additionally, there are 8 new presets for you to use, adding up to a total of 20 free presets!

Of course, it's still free, no account needed, VST3 + AU for Mac, VST3 for Windows, VST3 for Linux.

Download now for free: https://plasma-hype.itch.io/cogg

Trailer of V1.2 if you’re curious [a previous update]: https://www.youtube.com/watch?v=5q5EUMhUcaw

Thanks to everyone who tried v1.3, v1.2, v1.1 and 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 COGG v1.5!

Also, feel free to submit some PRESETS! I'll add them and credit you on the website!

A while back, I shared SROVA here when it supported TIDAL, local music and Internet Radio. Quite a bit has changed since then, so I thought I'd share where the project stands today.

For those unfamiliar with it, SROVA is a free, open-source, headless music player for Debian-based PCs and Raspberry Pi 4/5 systems.

The idea is straightforward: connect a Linux machine to a USB DAC, put it out of sight, and control your music from a browser, phone or tablet.

The Linux audio side

SROVA uses a dedicated ALSA playback path, opening the selected USB DAC directly rather than routing normal music playback through the desktop audio mixer.

• Bit-perfect lossless and Hi-Res PCM playback
• Automatic sample-rate switching
• No intentional DSP, software volume changes or resampling
• Exclusive DAC access, with release and recovery handling
• Support for AMD64 and ARM64 Debian-based systems

Music sources and features

• TIDAL and Qobuz, including Hi-Res playback
• Local music libraries, including USB and network storage
• Internet Radio
• Spotify Connect (separate from the bit-perfect playback path)
• Infinite Play and Auto-Mix for music discovery
• Artist, Track and Album Radio
• Shared playback queue across sources
• Browser-based control and a dedicated Android Remote

There's also SROVA Cast, an optional Now Playing display that can run on a Raspberry Pi Zero 2 W. It displays artwork and playback information while the audio stays on the main Linux player and its connected DAC.

One of the more interesting challenges has been managing exclusive DAC ownership while allowing other audio applications to coexist, particularly on desktop Linux systems.

I've been developing and testing SROVA across Debian PCs, NUCs and Raspberry Pis with several USB DACs. It's very much an ongoing personal project, and real hardware testing continues to shape its development.

For transparency, I've used AI assistance during software development, with the architecture, integration, testing and hardware validation directed and reviewed by me. The listening-room artwork accompanying this post is also AI-generated.

SROVA is free, and the Linux player source is publicly available.

Source: https://github.com/srova-music/player

Downloads and documentation: https://srova.music

I'd particularly welcome feedback from anyone working with ALSA, USB DACs or dedicated Linux audio systems. Always interesting to hear how others handle exclusive audio access and device ownership.

no that's it, they're all bad and buggy.

Plugins are mediocre at best and none of them have a good workflow.

If your gonna make a daw for you, keep it to you. I don't need it when there's like a 100 other option.

And I'm not talking about ardour, I hate that daw.

Bandlab, Bitwig and reaper work fine and are both tailored towards different ends of the spectrum.

I think the main problem here is that people make some simple plugins and chuck a good looking UI and call it a day.

You need a solid foundation before you think about changing the UI, and that's the thing vibecoders don't understand.

That's all, that's my little rant, hate if you want, but you will be met back with hate from yours truly. Good day

I built a lossless sample codec for polyphonic samplers that stays compressed in RAM and decodes 8 voices at once via AVX2

Built this for a pipe organ sampler (GrandOrgue fork) where sample sets routinely exceed available RAM, but the technique is domain-agnostic — anything playing many independently-positioned compressed audio streams concurrently.

The problem: normal lossless codecs decode with a data-dependent branch per sample, which kills vectorization. This one instead uses fixed-length blocks, a small family of fixed-order integer predictors, and a single fixed byte-width per block — branch-free, fixed-stride, so up to 8 independent streams decode together in one AVX2 pass.

Real numbers from my own largest sample set: 93.75 GB raw → ~41 GB compressed (2.27x), fits in RAM on a machine that otherwise couldn't hold it. ~85% faster decode measured in production. Full technical disclosure with measured results (published to establish prior art, I don't want this patented out from under anyone): https://github.com/sebion7125/v3codec

GPLv3-or-later, standalone CMake library, no host-app dependencies.

Hey everyone,

I recently put together ReplicantPeak, a small project I built to help streamline [insert core task: e.g., audio/workflow/modding]:

Lightweight & fast: Focused on getting out of your way and doing one thing well.

Open source: Free to use, inspect, and modify.

Check out the repo here:

https://github.com/DanimanX/ReplicantPeak

It’s still evolving, so I’d really appreciate any thoughts, feature requests, or bug reports from the community!

They randomly flip to extreme values, and don't go where I expect them to when I drag them. Sometimes they work intuitively and then a minute later they don't. I can't figure out what I'm doing wrong. Surely I can't be the only one?

I noticed that even though there are hundreds of thousands of music player applications that work just great, there isn't one that fits exactly my needs : I want the dark mode to fit a specific shade of blue and to it to give me the ability to organize my libraries in an optimal streamlined manner.

So I was thinking that the solution is for me to create an app with the heavy lifting of AI (Claude Opus for example)? Or better I can employ hundreds of agents to reverse engineer existing software like VLC and provide a polished optimized app for FREE and ask for donations? The cool thing is that I can ask for your precious feedback so I can feed it back to the LLM and make sure that for the following update it makes no mistake.

I'll make sure to include tons of emojis in the app description so everyone can understand it well and post it in hundreds of subreddits and other forums.

Would you all be in for such a plan?

⚠️ Trigger warning: AI project!⚠️

Backstory: I found out that the project https://github.com/thomas-nelson23/Openface-Mixer from u/hoppintruck23 wasn't compatible for my FireFace 400 which is an older firewire device. The FF-400 is an amazing unit, but it's almost 20 years old at this point. I then basically one-shotted a reworked Openface Mixer as https://github.com/BadgerSoft-NL/cachymix-rme-ff-400 for personal use.

Anyway, I got excited and wanted to build something in that same style to replace https://github.com/rncbc/qpwgraph which, with all due respect, I find a bit messy and I don't really like using it.

So I made Pecan Mixer, which takes the TotalMix approach from the Openface-Mixerand combines it with a patch bay from qpwgraph, and, oh yeah, a built in multitrack recorder that records hardware inputs and apps, because Linux audio is awesome.

MIDI is disabled for now because I haven't been able to test it.

Open sourced personal project, but I thought it was cool enough to share here

"It works on my machine": CachyOS with a recent AMD CPU

https://github.com/BadgerSoft-NL/pecanmixer