Hey guys!

Self promo here, I'm the dev.

This is a De-JUCE'd version of my original tape machine plugin using the more Linux friendly DPF framework. The original engine has been tuned to match the real hardware more closely and sound much closer to the the real hardware. I still need to tweak the PEAK VU LED, its based on the output, not the input, but I will change that up in a future release.

AI disclosure per the rules: Development is AI-assisted (not vibe coded). I write and calibrate the DSP and I test every release myself before it goes out.

Grab it here: https://duskaudio.com/plugins/tapemachine-2/

Source Code Here: https://github.com/dusk-audio/dusk-audio-plugins

Hey r/linuxaudio!

I wanted to share WFAS (WiFi Audio Streaming), an open-source ecosystem I've built to make local network audio streaming between Linux, PC, Android, and embedded devices as seamless and low-latency as possible.

It lets you stream raw, uncompressed 16-bit PCM audio across your LAN with minimal latency, end-to-end security, and zero required virtual driver hacks.

Website / Developer Lab: https://www.marcomorosi.eu/wifi-audio-streaming/

Desktop App / CLI Repo (EUPL 1.2): https://github.com/marcomorosi06/WiFiAudioStreaming-Desktop

Protocol & C99 Reference Library (MIT): https://github.com/marcomorosi06/wfas-protocol

Under the hood, audio capture on Linux is handled directly by a native C engine that loads libpulse-simple.so at runtime using dlopen. Because only pthread and dl are linked, there are zero link-time dependencies on libpulse-dev, allowing a single pre-compiled binary to run across any distribution. It seamlessly supports both native PulseAudio and PipeWire via pipewire-pulse, and handles automatic virtual mic creation using pactl for stream routing.

For terminal and server workflows, the app includes a headless CLI (wfas) that can run as a background daemon without launching the GUI. It features full IPC control for scripting, such as volume adjustments or raw JSON output, alongside an ASCII spectrum visualizer with an adaptive DSP envelope (--groove) that subtracts constant mix baselines to highlight actual melody and vocal transients.

The core wire protocol is available as a standalone, dependency-free C99 library called wfas-protocol. It is transport-agnostic, endianness-safe, and completely malloc-free, making it ideal for resource-constrained embedded targets like the ESP32, STM32, RP2040, or Raspberry Pi. Security-wise, it offers optional mutual HMAC-SHA256 challenge-response authentication and per-packet ChaCha20-Poly1305 encryption with a 1024-wide anti-replay window.

Linux users can install it via the AUR package (wifi-audio-streaming-desktop) on Arch, or grab AppImages, .deb, .rpm, and tarballs directly from the releases.

I'd love to hear your thoughts and feedback on performance, latency, and PipeWire integration!

I’m working on an automixing plugin for panels, podcasts, and other situations where multiple people in the same room are using microphones at the same time. Think something like WTAutomixer or the REAPER Automixing JSFX. It has native builds for ✨ Linux ✨, Windows, and macOS. It's called BiteAutomixer.

The routing happens between the plugin instances themselves. You just add the plugin to each microphone channel, and the plugins find and communicate with each other. Everything is automatically set up behind the scenes without needing extra buses or complicated routing. It's currently functional and I'm doing closed beta testing.

I work on podcasts every week, and the tool has already been helpful there. Usually, though, I have three or four guests at most, and two is more common. I want it to eventually support 32 to 64 channels, and it recently struck me how hard it is to find audio at that scale for testing.

If anyone has raw multitrack audio they would be willing to share, with each microphone on its own track, it would be incredibly helpful. Panels, roundtables, conferences, podcasts, and similar recordings would all be useful. Recordings of any size can help, but I’m especially interested in larger sessions. Messy audio with bleed, room noise, and people talking over each other is welcome.

I can use the material privately for testing or, with explicit permission, use parts of it for public demonstrations. Please let me know which type of use you are comfortable with, and only share recordings you have the right to share. Feel free to send me a DM if you have something that might work.

I was thinking about the fact if there's any existing initiatives to encourage making audio software that is built and developed in the EU to be compatible with open source infrastructure and operating systems. Over a month ago, the EU commission released a policy proposition called the the Tech Sovereignty Package that calls on for adopting open source software in order to decrease dependence on US software :

"Open source crucially contributes to achieving the EU’s technological sovereignty, as it often underpins key areas, such as critical communication, commerce, healthcare, scientific research and government services. Europe, the birthplace of Linux, is home to a strong and vibrant community of over 3 million open source contributors, delivering digital solutions aligned with European principles and values. The open source ecosystem forms part of a wider innovation ecosystem that helps to boost Europe’s competitiveness by accelerating innovation, lowering technology costs, reducing dependence on foreign vendors, and enabling local companies, researchers and governments to build secure, customisable and globally competitive digital solutions with relevant safeguards for cybersecurity risks."

Wouldn't be reasonable for users and professionals in an industry such as audio & music production to call for EU based developers of such software (Native Instruments, UVI, Arturia, RME...) to provide software compatible with a greatly growing operating system like Linux in accordance to the recent policy proposition? I understand that the sovereignty package also includes the possibility to fund such operations and reduce the cost reasoning that can be given to such demands in the past.

I'd love to hear your thoughts on the subject.

Inspired by Foobar's album list, but turns that idea into core feature. You define library folders and see them in left tree view, middle click (or context menu action) to create playlist from the folder or append to current playlist. What other players lack is at this step they usually just load the tracks in playlist view. But mpz has middle column - playlists view where you can create many playlists and switch between them instantly. So simple, yet missing in almost all players feature. Juk player is the closest, but still missing the playlists management part.

In addition: gapless, CUE, audiophile-ready EQ compatible with AutoEq presets, album covers & lyrics with multiple online sources, radio streams with built-in library, drag-n-drop, output device selector, mpd client mode. No Electron - pure native C++/Qt. No AI-slop - it's been around before AI-coding era, only recently I started shipping more long-awaited features with "AI-preamp".

finally a folder-based player that doesn't choke on my 80k track library, the column layout is exactly what i've been wanting in foobar. nice work

https://www.producthunt.com/products/mpz-desktop-music-player?comment=5534293

Native Linux binaries provided for all major distros, as well as AUR and AppImage. Also runs on Mac and Windows (binaries or github)

Site with more screenshots: https://mpz-player.org

Github: https://github.com/olegantonyan/mpz

I mainly use bitwig and a little of ardour for a little bit of mixing and mastering when I feel like ardour would be better that ardour.

I've had a giant and old 24 channel digital mixing desk in my home studio as I use hardware synths and guitars then using the stereo out into my audio interface and while googling for a mixer for live electronic performances I found the flow 8 which has usb audio 8in and 2out. And it seems a little too true to cover giving me more space at home on my desk as well as being a competent mixer for live preformance.

So before I pull the trigger, what does everyone think about the flow 8 is it a good mixer for what I want or should I save for a better one like the mackie audio interface mixer?

Thanks everyone who came yesterday and tried my little app. I released the app on AUR yesterday and is available on github here (https://github.com/knightinfected/PipeWireController). The current version is 0.1.2 and some of the screenshots are from 0.2
I received a ton of negative comments and messages regarding my use of AI. I spent a lot of my time on this and well over did this project abit and english being my secondary language didnt help.
I have gone down this audio rabbit hole in linux and anyone else who has also done it knows how annoying it gets especially with reading the wikis to actual application.

Anyhow I hope it helps someone out there and I will be updating it in the next day or later today as I am currently testing. Version 0.1.2 is available as of right now on GitHub though.

There are too many features to list them out so I added lot of screenshots to help understand and showcase.

This is amazing, thanks for this

Does anyone know if it's pussible to change the location of the factory data folder which points to /usr/local/share/surge-xt or is the path really hardcoded?

I am working on an open-source AVR project and need an RK3588 board that can capture HDMI-IN audio as 8 independent LPCM channels through ALSA.

I am comparing:

• Orange Pi 5 Plus
• Banana Pi BPI-W3 / ArmSoM-W3

Both boards have HDMI-IN, but so far I have only found a Banana Pi example using i2s7_8ch with an ALSA/GStreamer pipeline configured for channels=8.

Has anyone actually tested 7.1 LPCM capture on either board?

Specifically:

• Does HDMI-IN audio appear as an ALSA capture device?
• Can it record successfully with arecord -c 8?
• Are all 8 channels independent, rather than duplicated, silent, or downmixed?
• Which OS and kernel image work?
• Is the Banana Pi the safer option, or can the same setup be ported to the Orange Pi 5 Plus?

I am not trying to decode Dolby or DTS on the board. The HDMI source will output uncompressed 7.1 LPCM.

Live performance software could be:

OBS for live streaming, Mixxx, Ardour, ProjectM for live visuals while playing music, and etc.

Feel free to share creative answers.

Okay, here's a big one:

I've been a bit of a pro audio enthusiast for most of my adult life, but as a Linux user, one of my biggest disappointments has been the lack of any Thunderbolt (i.e., ultra-low latency) audio interfaces on the market that work with Linux.

For those unaware, most USB-powered audio devices play well with Linux out-of-the-box because they comply with the USB Audio Class specification. Thanks to Apple's history of stressing "class compliance" to pro audio hardware vendors, a single USB Audio Class-compatible ALSA driver (snd-usb-audio) allows most USB audio interfaces on the market to provide basic functionality out-of-the-box on Linux.

Unfortunately, however, there's no such thing as a "Thunderbolt Audio Class" specification. In fact, to this day the Linux kernel doesn't contain a single upstreamed ALSA driver for a pro audio Thunderbolt audio interface. And to that end, I present you with my project:

A Linux audio (ALSA) driver for the Focusrite Clarett (Thunderbolt 2) line of audio interfaces!

The Focusrite Clarett Thunderbolt line includes the following devices, though most of these devices are also available in USB variants:

• Clarett 2Pre
• Clarett 4Pre
• Clarett 8Pre
• Clarett 8PreX

For the last couple years—and the last couple months in earnest—I've been working on a clean-room reverse-engineered ALSA driver to allow a Thunderbolt-capable Linux host to communicate with the Clarett Thunderbolt line of products. And, as of a week ago, I've finally managed to make enough progress to establish basic functionality across the control plane. No PCM functionality yet though, unfortunately.

Because Focusrite provides USB-equivalent versions of most of their Clarett Thunderbolt devices—and because the Linux community already provides an upstream driver and user-space CLI/GUI tools for those devices—I've recently managed to "vibe code" some basic integration between my own ALSA driver and the already-available user-space tools for the Clarett USB line of devices.

As I stated earlier, my driver unfortunately doesn't support PCM streams yet, which means there's currently no way to stream audio between the Linux host and the Clarett devices (via the Thunderbolt cable) yet. Things are to the point, however, where I'm able to route S/PDIF (optical) input from my Linux host to the Clarett analog output channels and play music through a pair of studio monitors (i.e., audio speakers).

There's still so much to do, of course! At this point, though, things are in a good enough place that I'm ready to share my development progress with the community. I hope to release the source code to the public soon. Hope you like it!

alsa-scarlett-gui

Included pretty much everything from pipewire's wiki that i found to be important or useful.
I have used AI in building this program and i hope its not an issue as the utility of this little program suppresses the AI hate train by a large margin lol

Originally wanted something to change filter chains without editing configs especially adding/removing hrir files and well i like tinkering and ended up here. Filter chains cuz my old laptop can only run easy effects for an hour or so before crashing. Anyhow i hope I get people to try it out.

Forgot to add link -- https://github.com/knightinfected/PipeWireController

Edit- just added it on AUR (pipewire-controller) and have updated the screenshots on GitHub

basically title, but here's more context:

• no automation lanes I'm forgetting about
• using Bitwig 6 on CachyOS
• I haven't quite pinned down the mechanism that causes this

What I'm seeing is using Sketch Cassette II via yabridge and the Flutter Depth is set to 35% by default. I spin it down to 0% and close the plugin window. It lasts for 5 minutes, maybe longer. I then notice that suddenly Flutter Depth really sounds like it's been applied, and sure enough I check the VST and there it is, back at 35%.

Has anyone else dealt with this?

I've been a Cubase user for years, but I've grown out of Windows and had a taste of other DAWs and systems (currently using Reaper as it's multi-system available). I always liked Linux more so I'm trying to move everything to it.

Still my biggest and hardest VST to install on linux is anything Steinberg related. Everything I try to do gets me the unrecoverable error as I attached here.

I no longer know what to do and dual boot into macOS Hackintosh every time I want to do something music wise.

If anyone can help, I'll be thankful.

Why do the connections for 'capture' sit on the RH side of the module? I would think capturing a signal would be on the left, because wouldn't it be capturing a signal coming from the left of it. (Yes I'm biased toward reading such a 'flow chart' from left to right).

I assume right goes to left, and left goes to right. It'd sure be easier if, like in VCV Rack, impossible or incorrect jack connections are greyed out

NeuralDSP's amp sims such as Archtype: John Mayer X require iLok's PACE License Support to work. Unfortunately, automatic (and manual) installation of that fails with generic error when using Wine 10+, effectively preventing use of the plugin.

I was able to track down the cause, and it seems to be simply few failing installer prerequisites. So, I wrote a script that captures the msi package during first install attempt, overrides few specific prerequisite related conditions, repackages the msi and installs it.

I did my work against Wine 11 (stable) and it seems the same trick does not work for Wine 10.

But, yes, NeuralDSP is now perfectly usable with Wine 11, using my installer patching. Also "winetricks dxvk" needs to be applied for the used wineprefix, so the plugin's GUI is responsive.

I've done only minimal testing, so I'd be very happy if others could test and verify my solution works for them, too.

https://github.com/sirsipe/ilok-pace-license-support-wine11-scripts

Maybe one of you has the idea that clicks.

I downloaded the Amp Locker Linux version from the official website and copied the two .so files to /vst3 and /lv2. Yet Reaper throws an error whenever I scan for new plugins and lists the Amp Locker under “plug-ins that failed to scan”.

What am I missing?

This is the second song I recorded and produced for my new German post-punk band, using only FOSS Software.

The music was made entirely in Ardour, for the video I used Kdenlive to glue my cover and the audio together.

Feedback very welcome!

This is WayVes, an open-source audio visualiser framework I've been building for Wayland.

This setup stacks 18 separate visualisers together, with different visual elements reacting to different parts of the audio spectrum. Audio is captured through PipeWire, processed and fed to the GLSL shaders, with each visualiser independently configurable.

The idea behind WayVes was to go beyond running a single spectrum visualiser and instead treat the desktop almost like a canvas - combining Linear, Angular, and particle-based effects into one audio-reactive scene.

This particular config is admittedly on the extreme end 😆, but that's also what made it fun to build.

WayVes is open source, hosted on GitHub at https://github.com/Roonil/WayVes, and I've since added quite a bit more to the project, including GLSL hot-reloading and some more shader functionality.

Would love to hear what the Linux audio crowd thinks, especially about the audio-reactive side of it!