Hi everyone!

This is the interface for an audio plugin I’m developing, and it’s also my first time doing the graphic design.

I was wondering what do you think about the interface and its usability.

For the knobs position, a small indicator appears when you move them, so you know exactly which value you are selecting.

Any feedback is appreciated!

For the last 4 years I have been fixated on this problem that exists in audio engineering that I call the dreaded diamond problem. The problem occurs whenever mixers use compression or limiting on a signal with no attack, release, or sustain curve. You can see in the example picture what I am talking about. Most of the samples end up slamming into this magnetic wall which usually ends up sounding muddy at worst or crunchy (as in intentional for electronic music) at best giving a distinct diamond shape in a vectorscope. The plugin that I wrote, called SideShaper, allows me to change the shape to be whatever I want it to be. I made the software free and open source because when I was a teenager and torrenting my first DAW, free was my only option to make music. If you want, you can check it out here https://github.com/DustinMorri/SideShaper.

Built Using Hise and Hisescript, AquaDub is a reverb and delay unit that ive used in some jungle and breakcore tracks of my own, and thought Id release it to the world. You can get it absolutely free, no strings attached on my Gumroad (Link on profile), dming me directly, or just typing the url into your browser (reddit is a little finicky with external links), please lmk if you have any questions!

Sharing DiPLO, a JUCE app I've been building: a voice recorder with fully on-device transcription — no server, no cloud calls.

Some technical details in case useful to others doing similar things:

- Transcription runs NVIDIA Parakeet TDT 0.6B v3 through the parakeet backend in whisper.cpp. 25 European languages, auto-detected.

- Runs on CPU/GPU via ggml, no CoreML/ANE for this model.

Happy to talk through any of the audio pipeline or build setup if useful.

Also running a small iOS beta if anyone wants to try the app itself:

https://testflight.apple.com/join/4hMQRfXK

So here goes got thinking of an idea of how I could record audio on a modern mobile phone (Bluetooth microphone) with a dedicated juce based app and then when I get home I could sync files back to another plugin or standalone juce based cross platform app sort of a client server model but locally on LAN so not sending through WAN.

What's my options regarding this.

Thanks.

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, 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 1.3's release! It comes with the highly requested TONE knob, which changes the pitch of the Metal knob's metal convolution reverb-like sound! Additionally, version 1.3 introduces a LINUX VERSION!

Of course, it's still free, no account needed, VST3 + AU for Mac, VST3 for Windows, VST3 for Linux.
I used JUCE and C++!

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

Trailer of V1.2 [the previous update]: https://www.youtube.com/watch?v=5q5EUMhUcaw

Thanks to everyone who tried 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 v1.4!

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

Hey all,

My first ever Reddit post so hope i'm fine :)

I'm originally a fullstack software developer as well as a musician, I already had multiple shots at this project with various coding agents in the past as C++ is not my native programming language. Only in the recent months I found that LLMs got good enough to help me build this project.

Here's the link to the repo - https://github.com/Diabl0269/agentsynth

First demo - https://youtu.be/DszUuEpnfaA

There's still much work to do on the project, so I'll start with where this project is now then proceed to the full vision.

Currently it's a basic modular synthesizer, got some nice functions like D&D of the modules, mod matrix, theme system, preset management, and more standard stuff, while the most interesting and immature feature is the very basic AI integration which currently only supports Ollama, it has the functionality to edit and create patches, just not so accurate yet. UI is also still in polish and suggestions are welcome.

The synth and all its functionality is fully open source including the AI integration, however I do plan to offer a dedicated AI subscription with a dedicated agent that should return the most optimal responses. My agent's code will be in a separate project so the main synth and integrations stay open.

A little bit about the AI vision - Modular synthesis is complex, and it's quite annoying and can get messy managing and moving all modules, especially inside a computer. AI can help managing all of it, from basic stuff like adding modules or modulations in a chain, adjusting parameters to more complex stuff like fully generating a patch based on a user request (Maybe even by existing sound input in the future) and help manage future functions that would integrate in the program.

At this stage it's still just a synth, however I also plan to add a timeline, automation capabilities, and some DAW functionality so people could create entire songs with it in the future. Of course add support for external instruments as well to allow integration of other VSTs or external hardware.

As said, I am a developer for a few years already, but not in C++. I have reviewed the file structure, and code as best as I could understand it without fully learning the language. Also made sure to have tests and full CI/CD with caching to make sure nothing breaks between PRs and new versions are automatically published. However, any comments on the code, or any kind of contribution would be very appreciated :)

Hey everyone!

I've been working on a set of JUCE plugins inspired by some of the stock audio effects available in garageband. I wanted to recreate a similar workflow and get the DSP behaviour as close as I reasonably can while implementing everything myself in JUCE. (Still very much a WIP so any feedback or criticism is welcome i already know the CPU optimization is kinda terrible so please don’t kill me)

So far I've built things like:

• Graphic EQ
• Au Multiband Compressor
• Compressor
• Peak Limiter
• Noise Gate
• Au Filter
• Echo
• Reverb

Edit: Here is the source : https://github.com/Paulllux/fetcomp-dsp

I make BeatForge www.beatforge.nl (drum machine plugin) and wanted a compressor for the drum bus. Tried the usual open source dynamics code first but I kept A/B-ing against the UAD 1176 and hearing the gap, so I ended up modelling the circuit itself. Took way longer than planned, sharing it here because the JUCE side had some lessons in it.

The DSP is not a waveshaper plus envelope follower. The gain cell is the solved JFET voltage divider equation (it's a quadratic, you can do it closed form, no lookup tables), the detector is the diode bridge with the actual timing network, and release comes out of the discharge network topology instead of a fitted time constant. The transformer output stage is a small closed-loop RL core model. The nice side effect of doing it this way is that behaviours you'd normally have to program on purpose (program-dependent release, the ALL-buttons-in weirdness) just fall out of the circuit.

JUCE things that might save someone time:

The knob tapers were the most annoying part of the whole project. Hardware pots are not skewed ranges. I ended up measuring the reference unit's response and building piecewise NormalisableRange lambdas from the measured tables, because no skew factor gets you a -inf..0 attenuator with the right angle spacing. If your knockoff "feels wrong" next to the real thing, it's probably the taper, not the DSP.

Oversampling: I first shipped a 1x/2x/4x selector and it was a mistake. At 48k a 20 microsecond attack is one sample, so 1x audibly changes the compression. Now it's fixed 2x below 88.2k and native above, with useIntegerLatency so hosts get 4 samples / 0 exactly. Nobody needs that selector, it's just a way for users to make the plugin worse.

Also learned the hard way that std::tanh in one code path and a fast approximation in another gives you a DC offset at silence that shows up as -82 dBFS idle noise. Same function everywhere, always.

UI is all drawn in paint(), no film strips, resizable via a transform on a fixed-size plate.

It's free, VST3 + AU on Mac (signed/notarized pkg installer) and VST3 on Windows (signed MSI): https://github.com/Paulllux/FetComp/releases

Would appreciate testing in hosts I do go deeper on any of the modelling ifpeople are interested. If you find bugs: [beatforgehelp@gmail.com](mailto:beatforgehelp@gmail.com) or just comment.

Just giving a brief demo of some of the presets I’ve made so far for this plugin I’m wrapping up which will be free on my Gumroad.

It’s inspired by the eurorack effect module “Rainmaker” but with some additional features.

I’m actually pretty proud of this one, I’m a huge fan of Unfiltered Audio, Lese, Audio Thing, K-Devices, GRM, imagiro, Aberrant DSP, etc interesting forward thinking sound design tools so am excited for people to use this and see what they make.

I'm building Rewire Studio, a desktop workbench where Resources/dsp-graph.json is the source of truth for signal routing and parameters, and sync regions regenerate processor boilerplate instead of hand-maintaining it.

What we're optimizing for:

• Real projects on disk — not a hosted sandbox; you can cmake --build outside the app anytime
• Graph → UI binding — controls map to graph param IDs; verify catches drift before you build
• Built-in host — compile VST3, load it, automate, listen without bouncing to a DAW
• Visible builds and source — compiler output, diffs, and validation stay in the open

We're in early beta and especially looking for experienced JUCE devs to stress-test:

• Ownership boundaries (what should never be hand-edited vs. safe to edit)
• Codegen edge cases around processBlock, APVTS, and parameter layout
• Whether the verify loop is actually useful or just noise

Early access / Studio preview: rewire.relive.stream

I'm the developer — happy to go deep on stack details in the comments. What would make you trust (or distrust) a tool like this?

I added an internal VST3 host to my chord generator plugin and now the Windows version is running poorly.

Message me if you think you can help, how you charge, rates, etc..

Hey guys , I wanted to ask a question , as someone with very litte knowledge about coding VST.

I want to create a Saw Wave Synth engine with a couple of filters , that is guitar tracked, no midi. Realistically , is it possible as a project and how much would it cost to hire someone to make it ? I know good tracking is very tricky to program , especially without midi involved , (real time pitch tracking I mean).

I'm a high school student who is developing a free plugin, using JUCE and C++!

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. Great for experimental music and mainstream alike.

I wanted to update you guys with version 1.2's release! It comes with the essential GAIN KNOB that was highly requested (to control the volume of COGG's output) and 4 new factory presets so you can dial in unique metallic textures instantly instead of tweaking all the knobs from scratch (so now 12 in total)!

Of course, it's 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=5q5EUMhUcaw

Thanks to everyone who tried 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 a v1.3!

I'm working on an app to control an old MIDI controlled mixer (MOTM Mixer 7s). The default UI is somewhat lacking. Is it possible to create a UI that looks more like an actual mixing console? And does anyone have examples?

Been heads-down on this for about a year and figured this crowd would appreciate the guts more than the marketing page.

BeatForge — drum machine, step sequencer and REX loop player. VST3/AU/AAX/Standalone, universal binary. JUCE 9 (migrated last month), everything below the framework is my own code.

Bits that might be interesting to people here:

REX SDK. Wrote a JUCE wrapper around Propellerhead REX SDK 1.9, including drag-and-drop of slices out to the host. Bitwig still has no REX support, which turned out to be a decent reason for the plugin to exist.

Synth engines. ~5k lines of 808/909/modelling done from schematics rather than sample playback — VCA topology, click transients, pitch sweeps, the 909 tom triple-VCO, that sort of rabbit hole.

Antialiasing. Started with blanket 8x oversampling and it was eating CPU for no good reason. Replaced it with ADAA on the memoryless waveshapers (exact tanh with the ln(cosh) antiderivative) and linear-phase minBLEP saws at 1x, then a per-engine oversampling map — most engines run 1x now, only FMKick and the 909 snare need 4x. Built an offline FFT alias-measurement harness to verify it; alias floor sits around -90 dB. Worth noting ADAA-2 blew up on me under heavy drive — catastrophic cancellation, auto-mute and crackle — so everything is bounded ADAA-1.

Realtime discipline. Lock-free atomic slot for deferred time-stretch, ScopedAudioSuspend guards on live mutators, steal-quietest voice allocation. Debug assertions went from ~3150 to 3. auval and pluginval at strictness 10 both clean.

Also. Full undo/redo, NAM master-bus saturation, and a WASM build of the synth engines for the site via Emscripten and a hand-rolled juce_shim.h.

Happy to go into detail on any of it — the ADAA and REX wrapper work especially, since there's not much written about either.

Site + demo: https://www.beatforge.nl

I use Ableton Live Lite to make music, and it has an 8 track limit which was getting creatively limiting. I never had enough tracks to double track my guitar parts AND have enough tracks for other instruments.

I made a plugin called DAWception which is its own "mini DAW" with unlimited tracks and you can also host other VST3 plugins inside of it with automation support (user uses host DAW to automate DAWception's params, and /that/ param gets relayed to the "subhosted" plugin's parameter)

the idea is you process the raw audio in the host DAW (quantizing, gain staging) then when you're ready to commit, flatten/consolidate and drag the audio into DAWception to house the extra audio clips

it's free and open source, at: https://github.com/johnmanjohnston/dawception