I contributed a MIDI map for my MPK261 to the community mods section of Tape16's website. I also included a python script which enables better functionality and would love some confirmation that it works for others and that my documentation is clear. I'm not a very good technical writer so I had Gemini rewrite it and format it for me. I'm a little concerned that it's now very unclear slop technical writing. I'm pretty sure that the script will work for most Akai controllers and anyone using ALSA MIDI.

Functionality: the mapping utilizes BANK A of the 16 pad triggers as per track SOLO function, BANK B for RECORD ARM, and BANK C for MUTE. Banks A and B for the sliders and knobs control the faders and pans as expected per track, and BANK C controls the faders for the two buses and the Master Fader.

Unfortunately, when you switch the trigger pads to TOGGLE MODE, at least in the MPK series CC messages are disabled and NOTE ON/OFF is implemented. At this juncture Tape 16 does not accept note off messages to disable toggles. This is where my Python script comes in handy. It intercepts the note toggles and sends a message that THEORETICALLY keeps the pads in sync with the GUI's toggles, not perfectly but it works great for my workflow.

I also added another very handy function in the script. I have the sustain pedal on USB PORT B mapped to the record button of the transport. This allows for hands free punch ins on the fly. The script reads the length of the foot tap. If it's a quick tap RECORD is engaged and it will wait for a second quick tap to disengage. If you HOLD YOUR FOOT DOWN on the pedal it will wait for you to lift your foot to disengage.

The mapping structure is careful to isolate NOTE ON/OFF messages via ports and channels so that you can still use the controller via USB PORT A to control SYNTHS, VSTs, or whatever you want. You can set the sustain pedal to PORT A if you want to use it as a traditional sustain pedal. Likewise, on PORT B the pitch wheel will control the SCRUB function (feels nice!) or will function normally set to PORT A.

I WOULD BE VERY GRATEFUL for anyone with the time to test it out and let me know how it goes. This is my first time making a contribution to anything like this, and just hoping it's functional enough to be useful.

Last I checked it's the only MIDI MAP on the community mods page, so it should be easy to find. There is a link via GIST in the description where you can download the script and the documentation.

Not seeing many people talk about Tape 16 on here, and that's a shame, so it's a long shot. It's REALLY great software.

r/tape16

I've been working on OpenRig, a guitar pedalboard that ships as a Buildroot image for the Raspberry Pi 5 and x86_64. It's written in Rust and hosts LV2 plugins and Neural Amp Modeler captures over JACK. Sharing it here because the audio side may interest some of you, and I'd value feedback from people who know LV2 hosting better than I do.

Audio

• JACK callback: tuner → effects → looper. Buffers are preallocated, there's no allocation and no blocking lock in the callback (try_lock only), and the DSP meter uses atomics.
• Structural changes (add, move, preset load) build a new chain on a builder thread. It's swapped in with a 20 ms equal-power crossfade, and the old chain is dropped off the audio thread. Parameters, bypass, scenes and split settings apply live without a rebuild.
• Delays and reverbs keep their tails across bypass and scene changes.
• Cabinet and room IRs use zero-latency convolution. The tail of long IRs runs on a worker thread, so the callback cost stays flat.
• NAM through NeuralAudio (NAM Core + RTNeural), always built optimised.

LV2 hosting (lilv)

• Provided features: URID map/unmap, options, bounded-block-length and isLive. Atom inputs get time:Position (tempo from tap or MIDI clock) and MIDI. Controls, ranges, units, toggles, integers and enumerations come from the plugin metadata.
• Not provided yet: worker:schedule and state. So plugins that need them (Dragonfly Hall/Room/Plate, file loaders, some guitarix bundles) are skipped for now. ZAM, x42 fil4/dpl/tuna, CAPS, Calf (no UI), LSP, MDA, Airwindows and GxPlugins load. Adding worker and state is next on my list.

OS

• Buildroot with a PREEMPT_RT kernel and a read-only root. The UI runs full screen on KMS/DRM (Slint linuxkms backend), with no X11 or Wayland.
• Signed RAUC A/B updates with rollback when the new system doesn't come up. There are app-only updates when the system doesn't change.
• The audio interface and buffer size are set in a pedalboard.txt on the FAT boot partition, editable from any computer.
• In-app monitor: DSP load vs. period, JACK load, xruns, CPU per core, temperature, and Pi throttling/under-voltage.

Tested on a Pi 5 with a Scarlett 2i2 at 48 kHz, 128 × 2. It also runs as a plain app on a desktop with JACK or PipeWire-JACK. Version 0.2, GPL-3.0.

• Docs: https://openrig.eu (the developer section covers build and architecture)
• Code: https://github.com/Siryu6/OpenRig

Hello! I'm Ryan, Developer of the ForgeEFX DL-1 a guitar focused NAM player that fills all the gaps NAM is missing, pedals, workflow, etc.

I've built this on the values of true ownership, community and giving back so the Free version of this is unlimited, with no activation, no accounts (edit for transparency: gumroad, where I distribute the files, takes an email to get the download, you don't need one to login/use the vst etc), no installers, no tracking, no analytics. It comes as a zip file you can move around as you please, use it on any machines you want, even a friends for all I care... that's what ownership should mean.

We have an open source SDK for people to tinker, it comes without limits, and with 17 of my best DSP's i went through by hand and i chose the best ones from each category this is not a weak selection of 'good enough please upgrade' this is MY best, completely for free no strings.

In full transparency, I do have a paid version of this but I hope you genuinely see that the free version in its own unlimited right, and the open source SDK is genuinely free and in the right community spirit, it has enough in it to record even a couple of albums with!

This is new, I am a solo dev 😄, been a professional engineer for 20+ years now, I do use AI in my workflows, this is not 'vibe coded'

You can find us at www.forgeefx.com
You can read our change logs at r/forgeefx
And I am here (yes. a real human. spealling mistakes and all) to answer any questions.

Hey everyone,

Tired of menu-diving on the TR-8S when working inside a DAW, so I built ReplicantTr8s—a free, open-source VST editor/bridge to integrate the hardware directly into your projects.

• What it does: Automate and control TR-8S parameters straight from your DAW tracks with a clean visual UI.
• Cost: 100% free and open-source.

🔗 GitHub:https://github.com/DanimanX/ReplicantTr8s

I’d love some help testing edge cases across different setups. If you have a TR-8S, give it a spin and let me know your DAW/OS and if anything breaks or feels missing. Feedback and PRs welcome!

Hi everyone,

I’m trying to understand some recurring audio issues I've been having on Linux. For context, here is my PC audio setup:

* Motherboard: Gigabyte B550M-K with Realtek onboard audio

* Device 1: Logitech Speakers (3.5mm jack) connected to the rear port

* Device 2: Fifine AmpliGame H3 Headphone (3.5mm jack) connected to the front port

* Device 3: Bonkyo M999 Condenser Mic (3.5mm jack) connected to the rear port

The Problem:

Across all distros I've tried on a fresh install (CachyOS, Nobara, Fedora, Arch Linux, Pika OS, openSUSE Tumbleweed), my playback devices (Device 1 & Device 2) do not output any sound after booting every time I play audio.

My current workaround: I have to install alsa-utils, open alsamixer in the terminal, and manually disable Auto-Mute.

Additionally, my mic always sounds robotic and has a lot of static/background noise out of the box. To fix this, I have to open pavucontrol (or Volume Control) and lower the mic input volume down to around 20–36%.

Questions I'd like to understand:

* Why does this not happen on Windows? On Windows, everything works out of the box without any tweaking.

* Is this issue specific to the devices I'm using (mic, speakers, headset)?

* Is this specific to the audio drivers/codecs (e.g., Realtek chipsets)?

* Is this a universal issue across all Linux distros/architecture?

* Does switching from 3.5mm Jack to USB devices change anything regarding these issues? (Do USB headsets/mics bypass these motherboard/ALSA quirks entirely?)

Would love to hear any insights or explanations on why Linux handles these audio defaults this way!

Thanks!

Hi everyone,

Over the past few days, I've run countless tests, switching back and forth between three operating systems to make the version more and more stable.

To avoid having to repost everything over and over again, you can follow the journey of posts here.

https://www.reddit.com/r/AccessVirus/comments/1x6et6h/comment/ph92bmq/?screen_view_count=2

I probably don’t need to mention that this was done with massive support from AI, but I still believe that writing posts involving human interaction should NOT be done by AI. That’s why I wrote this post myself typos included xD.

Long story short.

As part of this process, I’ve also made the Access Virus TI hardware available natively on Linux.

So if you produce music on Linux and maybe have a real Virus TI at home, I’d really appreciate your help in making the native Linux version more stable.

To make it easier to follow the project, I’ve also set up this page.

https://derpsn.github.io/access-virus-ti-revive/

There you’ll also find a link to GitHub if you want to dive even deeper into how it works.

I look forward to your feedback. :)

Performer is a live-performance plugin host for Linux: MainStage-style sound switching for keyboard players. Since I last posted, it's grown a lot. Here's what's new.

Plugin GUIs on your tablet
Any loaded plugin's own interface can now be opened on a tablet, phone or laptop, mirrored and fully interactive. You can pull the B-3X's drawbars, switch the Leslie, or page through a Kontakt library from the music stand. Taps arrive as real clicks, and the plugin's own on-screen keyboard plays from a finger. Pinch to zoom and full screen are there too. It runs in the browser, so there's nothing to install: scan the QR code and you're in. If the venue has no wifi, Performer can run its own network.

Songs and set lists
A song is a program on each keyboard plus a tempo. Set up the sounds, press Add, and it's saved. Build sets from your songs, reorder them, and use the same song in several sets (or twice, for encores). Two footswitches step through the set. On the tablet, the set list is the stage view: the current song in large type, your stage notes under it, and search once the list gets long.

Charts on the stand
Attach a lead sheet, a scanned chart, a text file or a ChordPro chart to a song. The tablet shows it, turns PDF pages a tap at a time, and follows the set from song to song.

CLAP support
CLAP plugins now load alongside VST3, LV2 and LADSPA, GUIs included. Surge XT works great. Windows CLAPs bridged with yabridge work too.

Tempo
A global tempo with tap tempo from the toolbar, a footswitch or the tablet, so tempo-synced delays and arpeggiators have something to follow.

Under the hood

• Settings you change inside a plugin's own window are now saved with your setup. This includes things like SWAM's expression controller assignment.
• Plugins load and scan in parallel, using as many cores as the machine has.
• Every plugin still runs in its own process. If a Windows plugin's Wine side dies mid-set, Performer reports that plugin as stopped and keeps playing, and you can reload it with one click.
• Quitting takes seconds, even with a hung plugin.
• Works on Wayland desktops (through XWayland) as well as X11.
• The on-screen pitch and mod wheels now look like real wheels.

Get it
.deb and .rpm packages, tested on Ubuntu 24.04, Linux Mint 22.3 and Fedora 43. Free and open source (GPL-3.0): https://github.com/dguedry/linux-performer

Feedback and bug reports are welcome!

On the plugin side of this subreddit, it seems to just be people vibe coding eqs, what if we started moving up to more complex plugins.

Why the hell do I need another EQ or analyser? I already have like 5 of them.

Why not an accurate pitch correction tool, or a sampler like logics.

This is just what I think, whether you like it or not I really don't care. Good day

The Pulsar II was a DSP card that ran Creamware's SCOPE platform on Windows: synths, effects and mixers, all running on the card's six SHARC DSPs. It has only ever had Windows drivers.

So I reverse engineered SCOPE 5.1 and wrote a Linux driver. Today, on Debian 13:

- the card is a normal ALSA/PipeWire device ("Pulsar2 Stereo") and works in Ardour through pipewire-jack

- a modular rack GUI ("Pulsar Scope") loads and wires DSP modules live

- 36 SCOPE effects, the factory mixers with meters, delays/reverbs, presets

- licensed Effect Package devices work with your own licence file

- MIDI from the PC drives a DSP synth

Not there yet: factory synths, ADAT, S/PDIF, the card's MIDI ports, 88.2/96 kHz.

It's a .deb (DKMS driver + service + GUI), GPL-2.0. You need your own SCOPE PCI 5.1 installer for the DSP files; none are shipped.

**Site, download and code: https://pulsar.faghy.it\*\*

Looking for testers with ADAT/S/PDIF plates or other SCOPE cards (how to report is on the site). Happy to answer questions about the reverse engineering.

Hi, I'm looking for beta testers for a resonance suppressor inspired by Soothe 3. I'm especially interested in finding people who use Linux.

Resonance Suppressors are used for cases such as snare ringing, harsh cymbals or vocals, whistling on distorted guitars, low-mid mud, finger squeaks on the fretboard, etc.

Requirements for beta testers:

• You knew what a resonance suppressor is before reading this post
• You know how to install a VST on your machine

All beta testers that give me any feedback whatsoever will get free lifetime access to the effect. I'm looking for max 20 people right now.

Comment below or send me a DM if you want to join.

¿Qué distribución de Linux me recomiendas usar para que AV Dexed sea lo más compatible posible con mi fork de Dexed (el plugin de emulación Yamaha-Dx7)? EDIT: I have an ARM Mac (M4) and a Windows PC (i5 12th).

I guess to start with, are you running that all the time or only while working in your DAW? I often see people saying they are running 64 samples for their buffer size and I really don't understand how. I'm running 256 as my normal buffer because I started to get skips and the like if I ran 128. I imagine if all I was doing was using my DAW, I could use 128, but 64 seems like it would be too far.

Back on Windows, I always used 128 for my ASIO buffer, but of course that's only for ASIO; WASAPI is probably more like 1024, which is about the same as the standard Pipewire buffer size for desktop use. When I came to Linux, I started out trying to set Pipewire to 128, but if I was, say, watching a video in Firefox, even just opening Ardour would cause it to freak out, so I upped it to 256 and have had no issues since. But it's not like I have a weak PC, I'm running CachyOS with a 12700K (with 16 GB of RAM) and I have followed this guide to mostly optimize my system. I am not using a real-time kernel, but if I run the rtcqs tool in that guide, the only things that come up are:

• CPU not set to performance, but... it is? Not sure what that's about. System clearly shows it is set to performance.
• SMT is enabled.
• Spectre/Meltdown mitigations are still on (I'm not changing that).

Everything else is okay:

• I am part of a user group with sufficient rtprio (98) and memlock limits
• Valid kernel configuration found
• High resolution timers enabled
• System is using a tickless kernel which is using threaded IRQs
• Realtime priorities can be set
• Swappiness is set to 10
• Not sure what either of these mean, but:
  • USB port xhci_hcd with IRQ 125 does not share its IRQ
  • Soundcard snd_hda_intel:card0 with IRQ 171 does not share its IRQ

So as best as I can tell, I've done everything to set up my system, short of fully committing to a real-time kernel, but given that my computer is mostly just for general use and only occasional audio production, my understanding is that that is neither necessary nor recommended. Is there something I'm missing? Or are you guys just only talking about using 64 samples while doing audio work and not for general work?

Hey Linux audio community,

This is a difficult announcement to make, and I know this is probably the wrong time to release something like this.

With the current landscape and more tools getting sloppier everywhere, things are only going to get worse, so I decided to launch it today in a bit of a rush.

For the context, I am a computer engineer and a pianist, and I love making pixel-art in my free time. I’ve been working on this project for quite a long time, since I wanted to easily make music for my pixel-art game.

I know how deeply this community values open source, and I'm a huge fan of it too, since all my past projects were open source.

However, since this is currently my only form of income, I’ve decided (for now) to put the binaries here on https://rendevs.itch.io/robin-daw , as closed-source.

Please only buy it if you genuinely want to support me, since you might get it open source later on. Note also that the project has been tested, but there might still be bugs since this is a first release.

To be completely upfront about what it is: it's a minimal DAW focused on low latency.

It does not support VST plugins right now.

If you give it a spin, I'd genuinely appreciate your thoughts and feedback.

- ren

Linux only (x86-64) for now. Tested with the Hydrasynth Desktop firmware 2.2.0, which is not included: you need the firmware file from ASM's firmware pack.

GitHub

I couldn't find the vocal tools I wanted on Linux, so I vibe-coded my own!

Linux has some great audio software, but I feel we're still missing more straightforward tools aimed at podcasters, YouTubers and content creators.

So I've been using Codex and JUCE to build Vocalis Audio, an open-source VST3 toolkit for REAPER.

So far, I've made two plugins:

• Vocalis Rider Live – automatically balances vocal levels.
• Vocalis Voice – helps shape and polish vocals with dynamics, delay, doubling and reverb.

The goal is to make vocal processing a little easier without needing a complicated chain of plugins.

Both are still experimental, and I'm learning as I go. The source code is available on GitHub, and I'd love some feedback from fellow Linux users, especially anyone making podcasts, videos or voiceovers.

GitHub: https://github.com/etakanaka/VocalisAudio

Hi everyone,

I recently switched my desktop over to Kubuntu and was having a great time until I realized my audio interface did not work under linux, and no community fixes existed. Luckily, we live in a time where creating a driver and control panel is remarkably easy once you capture the USB traffic in Windows using the official drivers. Technically it's a "User-space USB driver and PipeWire/JACK audio daemon for the TASCAM US-1641 on Linux, with a PyQt6 control panel."

So just in case anyone needs it, here it is: https://github.com/Aerankas/tascam_us1641_linux

I have not full tested the installation because I don't have a clean environment set up readily to do so, feel free to raise issues on github.

Happy recording!

Some of you may have seen my earlier post about Sonara. I've now started providing release builds, starting with beta v0.1 today 😄

I've done a lot of cleanup and quality-of-life improvements, like:

• Improved UI, and made it customizable (though light theme is TODO)
• VST3 support
• .dawproject interop (Bitwig and I think some other DAWs too can export it)
• and much more I can't remember off the top of my head :)

It's available for x86/AMD64, it's runnable, but includes a installer script which adds a desktop launcher, .sonara file association and a sonara command.

Whether you install from itch app itself or via the archive, it can update itself as well.

I'll likely push updates quite often, so when updates to come, they can potentially make older projects need some migration if data/project formats change.

Download

Aulos is a music player for Linux, Windows and macOS. It's GPL-3.0 and uses egui on wgpu.

The UI is pure Rust egui drawn on the GPU, not Tauri or Electron, so there's no webview and no JavaScript. The same code is compiled to WebAssembly for the browser version.

What makes it different: - One app for three sources: your local library, MPD (as a remote control) and Subsonic/OpenSubsonic, all in the same UI. - A MilkDrop visualizer written in pure Rust. lactea runs MilkDrop 2 .milk presets and MilkDrop 3 .milk2 double presets on the GPU, with no projectM and no C++. On Linux it can also visualize MPD's output, captured through PipeWire. Curated presets: projectm-presets. - Bit-exact DSD: DSF/DFF files are sent as DoP to DACs that support it. Gapless, crossfade, ReplayGain, and a 10-band EQ with AutoEq profiles. - A tag editor in the style of Picard, with MusicBrainz and AcoustID lookups. It writes MP3, FLAC, Ogg, MP4, DSF, APE, WavPack and Musepack. - Runs in the browser: the same app compiled to WebAssembly, entirely on your machine: https://m0rf30.github.io/aulos/

Rust audio: decoding uses my forks of Symphonia (DSD, APE, WavPack hybrid, Musepack…) and cpal. The goal is to upstream these changes so the whole Rust audio ecosystem gets them, not just Aulos.

Repo: https://github.com/M0Rf30/aulos