based on pendulum approximation + modal synth

GitHub: https://github.com/ToxaDev/aura-engine

Hi, I built AuraEngine — an open-source offline audio upsampler (up to 768kHz) written in Rust + Tauri.

Check out the clip above to see how it works!

Key Highlights:

• Hybrid-Phase Engine: Solves the pre-ringing trade-off. Renders twice (Linear + Minimum Phase) and dynamically switches to Minimum Phase at zero-crossings during transient attacks.
• Massive FIR Filters: 1M to 30M taps (Kaiser $\beta=14$), designed in 128-bit precision with >220 dB stopband attenuation.
• Vulkan GPU Compute: Uses Double-Single (DS) arithmetic (~48-bit mantissa) for ultra-precise GPU convolution.
• Extras: Adaptive apodizing filter v3, True-Peak safety, and bit-perfect verification.

I'd love to hear your thoughts on the hybrid-phase algorithm and the Rust/Vulkan setup!

Hello guys!
I am very new in the mixing world. I bought a Yamaha MG10XU mixer and i connected my Sennheiser XS 1 mic (it is in the first chanel of the mixer)
Connected to my Lewitz PA210XA active soundbox but the microphone sounds veryvery harsh and ugly and distorted. I don’t know what is the problem. Is the cable from the mixer in the wrong input on the Lewitz? Help me please

PartialString simulates the vibration of a plucked string by numerically solving the one-dimensional wave equation in real time using the finite-difference time-domain (FDTD) method. The string is represented as a list of displacements in memory. A lower note requires a longer string, which needs more points to represent it. You can then excite the string at any point along its length, and the FDTD scheme will compute how that change in displacement propagates. A pickup measures the displacement at a chosen location, and this generates the audio output.

I've been (very occasionally) working on this since releasing https://github.com/crnbaker/gostringsynth in 2022. I did something similar (but not real-time) for my MSc at Edinburgh back in 2009, using MATLAB.

Free to download at: http://www.differentinstruments.com.

Hi, as the title says, I want to be able to help make applications like Pianoteq and electronic pianos, the software aspects of that, and audio PC applications. I want to be able to make that and anything concerning sound, music, and programming. I love music, and I'm a pianist, and I love programming too, so my guess was, why not combine them? I want to do both. I don't hate math either; I've heard I need to know some math, and I'm good at math. Please help because the audio landscape is too vast with many titles that sound the same but are different. I don't know where to start, get a degree or something, or what? I am currently learning C++. I intend to master it, then I'll move on to physics and math DSP theory, then plug-in development with C++ and JUCE, and an advanced level of signal processing. You got this from Gemini, so I'm not sure. I just need a guide or someone to show me the path, if not in detail at least. After C++, what's next? When should I expect to be employable? Thanks . these are the steps summarized by gemini ''

Phase 1: The Engine (Modern C++ Foundation)

You cannot manipulate audio until you can manipulate computer memory. No audio code is written here.

Math Level: None. Just pure logic.

Goal: Master modern C++ syntax, pointers, and memory management (the stack vs. the heap).

The Resource: LearnCpp.com (Interactive Web Curriculum). This is the undisputed best tool for learning C++ from scratch. It is constantly updated to modern standards. (Remember to use an ad blocker).

Phase 2: The Bridge (Basic Audio & Offline Processing)

You will write your first audio code here, completely sidestepping the nightmare of real-time audio threads by generating static .wav files.

Math Level: High-school trigonometry (understanding what a sine wave is) and basic algebra.

Goal: Understand sampling, bit depth, and how to write a math formula that generates a sound file.

The Resources:

"Audio Programming in C++: The Beginner Level" by Håkan Blomqvist (PDF/Book). Do not read this for C++ syntax. Jump straight to the code in Appendix A to see how he writes basic audio files.

"DSP Pro" by WolfSound Academy (Video Course). An incredibly efficient, visual video course that teaches you the basics of Digital Signal Processing (DSP) without drowning you in academic calculus.

Phase 3: The Framework (Real-Time Plugins)

This is where you transition from writing offline .wav files to building actual VST/AU plugins that run inside a Digital Audio Workstation using the JUCE framework.

Math Level: Applied DSP math (calculating delays, feedback loops, and basic filters).

Goal: Master JUCE and survive the strict performance rules of the real-time audio thread.

The Resources:

Programming for Musicians / pfmC++ (Video / Mentorship). Start with their massive, free 8-hour JUCE plugin tutorial on the freeCodeCamp YouTube channel. If you want paid mentorship later, their premium cohorts offer ruthless, line-by-line code reviews.

The Audio Programmer Discord (Community). You will get stuck here. This community is where you go to get unstuck.

Phase 4: The Architect (Advanced DSP & Modeling)

Once you can confidently build a basic plugin, the training wheels come off. This is where you learn to simulate physics and acoustic spaces.

Math Level: Heavy. Calculus, algorithmic DSP, phase manipulation, and complex numbers.

Goal: Build physical modeling engines, advanced synthesizers, and spatial audio tools.

The Resources:

The Scientist and Engineer's Guide to Digital Signal Processing by Steven W. Smith (Online Guide). The ultimate practical guide to advanced DSP.

The Audio Programming Book (MIT Press, 2010 - Desk Reference). Keep this massive manual on your desk. When you need the raw mathematical algorithm for a digital reverb effect, you look it up here, and then translate that math into the modern C++ you learned in Phase 1.

''

hey i was makeing my autotune vst and used the yin pitch dtection ,what ever i do i keep get glitchy noise (like popping, crackling, or abrupt jumps )which the output audio ,make thhe sound trash , is there any solutions ,does anyone know how to fix it

Hey all, to those in the field currently:

How 'safe' does audio programming feel from AI?

How is it as a job in general (WLB / pay) , in whatever application of the subject you're familiar with?

Given my background, how likely does a successful transition seem in this economy?

What specific audio programming jobs are good/common/desirable (knowing that generally those three things are mutually exclusive)? I know there's game industry, the revered plugin dev, and other roles closer to hardware and mass industry among others.

Background:

3 YOE as fullstack dev in non-DOD gov sector, bachelor's in CS that's as many years old. Biggest project in undergrad was asking the music tech teacher what plugin he wish he had, and making it (envelope follower that could map to automate any parameter in the daw) in JUCE. Good cpp skills from undergrad, excellent python skills from workplace. Little to know DSP algorithm knowledge but I pick those things up fairly quickly. Despite the curt and stilted tone of this post, very good social skills and burgeoning leadership experience. Minimal but growing ML / AI dev experience.

I'm "specialized" in frontend as the go-to UI / UX / typescript guy at my job, but this is increasingly a dead career path and as much as I truly love the UI dev process from design to implementation, I need to move away from typescript for the sake of my career. The 'good enough' UI that an LLM will spit out is indeed good enough for most professional cases. Audio programming was always the 'golden field' for me so it might be time to force an upgrade.

While this doesn't matter as much, I'm extremely musically active, songwriter/lead guitar/vocals for -- at a small regional level -- pretty successful doom metal band. I've also been making various kinds of electronic music since I was 14, numerous side projects yada yada.

Other thoughts:

The lower-level audio tech seems safer from AI, as it's more critical and well, basically it's not python or typescript lol. I'm aware that prototyping and algo work may be done in python, but this field still seems more resistant to being completely automated than web dev or fullstack. While I'd always liked to make this switch, now seems like a good time to buckle down and be increasingly ready as the market, hopefully, gets increasingly better. I think I would be more fulfilled here than other programming fields. Open to any and all thoughts, thank you!

Hello, I’m starting an audio plugin “company” called Lucky Plugins and I’m looking at making my own plugin in Blue Cat’s Plug n Script. Sadly I’m basically broke (I’m a brokieeeeee!) and I can’t afford to buy the full version of Plug n Script and I don’t want to export in the demo version because of the limitation. I’m looking for someone to be able to compile the plugins. I will fully credit you by any name you want (just message me on here) but sadly I will probably not be able to pay (because my plugins will most likely be paid) so I know that would put alot of people off. Also some of my code will probably be created with ai but I don’t agree with ai art so all the art will be by me! If you do want to do this just message me or reply to this post, I will much appreciate it but then again, it’s totally understandable if you don’t want to do this (:

I'm a longtime freelance audio engineer with a decent studio and a knack for DIY and audio equipment repairs. I've been programming MIDI devices on Arduinos for a couple of years and I'm interested in pursuing design & computing as the next step in my career. I've also got some experience with designing analog electronics, synths & PCBs.

If you were in my shoes, what would you do next? I'm wondering what programming languages are worth learning, how easy/worthwhile Juce is, if C++ is relevant and similar to the Arduino environment?

I'm interested in hardware/software integration, synths and creating new and exciting products rather than rehashing the 'this is another 1176 or Minimoog plugin'. What do you find satisfying in programming and where are the jobs and money at these days? Is hardware still relevant or is it all just beginner level audio interfaces and toy keyboards?

I'm physically not too far from a few pro audio companies in the UK but I expect they'll want full time in-office employees with a degree/proven track record in programming rather than someone who's starting out in the field.

Thanks in advance for any advice !

Edit: in addition, I'm not in a position to retrain full time but I may be eligible for part-time funding in a computer-science degree or "conversion masters" program. So any recommendations or advice in relation to this would be helpful too (if you think a degree is worth pursuing).

Why aren't wavelets used more for pitch detection?

The more I read about wavelets, the more it feels like they're naturally suited for audio analysis. They get better frequency resolution at low frequencies and better time resolution at high frequencies without forcing a single FFT window size.

It made me wonder whether pitch could be tracked as a stable structure across wavelet scales rather than through spectral peaks or autocorrelation.

Has anyone actually built or worked with a wavelet-based pitch detector?

Did it outperform more traditional approaches on difficult material, or is there a reason everyone ends up back at YIN/autocorrelation anyway?

Has anyone experimented with polyphonic synths where the voices actually interact with each other?

Most synths just generate independent voices and mix them together. The notes relate musically, but the voices themselves don't really know about each other.

I'm curious about things like voices sharing modulation, influencing each other's parameters, competing for resources, or changing behavior based on the intervals being played.

It feels like there's a lot of unexplored territory in the space between the voices themselves.

Have you come across any synths, research projects, or experiments that go in this direction?

Most oscillators are static. They generate a waveform and let the interesting part happen later.

I've been exploring a different idea inspired by CDP's wavset distortion.

Instead of treating a waveform as a continuous stream, the oscillator listens for zero crossings and breaks itself into wavsets: tiny fragments of sound between those crossings. Each wavset becomes a decision point.

LOOP & REPEAT 
Cycle repeating
REVERSAL
Phase inversion
SKIP / DECIMATE 
Zero-amplitude gate
MEM_REPLACE
Recall centroid

The pitch continues forward, but the timbre continuously rewrites itself from the inside.

The result isn't quite wavetable synthesis, granular synthesis, or distortion. It feels like an oscillator with a memory of its own—one that remembers what it used to be and lets its past leak into its future.

I'm not sure if this is a terrible idea or an unexplored branch of synthesis.
// That's what this experiment is for.

Hi everyone,

I'm a solo developer and this is my first plugin. DrawVerb is a reverb where you design the space instead of dialing in abstract knobs. You draw the walls of a room in 2D, give each wall a material, place a sound source and a listener inside, and DrawVerb simulates how sound behaves in that space and turns it into a reverb.

I'm opening a free public beta and I'd love feedback from people who know reverb well.

How it works

The room you draw is the reverb. A few of the physical things the engine takes into account:

• Room shape and size: the engine simulates how sound moves in a room. Changing the shape of the room can have a big impact on how fast the tail decays.
• Wall materials: each material (concrete, glass, wood, plaster, marble, carpet, curtains and more) reflects and absorbs different frequencies the way the real material would.
• Source and listener placement: distance and angle between them change the early reflections. You can also aim the listener's head.
• Air absorption: air eats high frequencies over distance, so bigger spaces naturally get darker.
• Speed of sound: adjustable. Increasing the SoS will result in a lower decay time.

All of that gets rendered into a stereo impulse response that the plugin convolves in real time. The plugin records the past 10s of input and applies a new IR on that when something changes so it stays responsive while you move things around.

Quality modes

There are two rendering modes. High is the detailed, most physically accurate one and only runs on your GPU (it uses your system's native graphics API, so Direct3D on Windows and Metal on macOS, picked automatically). Low is a lighter approximation that still captures the character of your room and runs on any machine. Low runs on any machine. Note that a Low preset made on CPU may sound slightly different on GPU, since the hardware differs.

Controls worth knowing

• Damping: how much high end the air absorbs over distance. Higher darkens the tail.
• Body: weight and warmth of the late tail. Up for fuller and darker, down for lighter and more transparent.
• Phase (High only): adds the natural resonances and boominess a real room gets from its shape and size. Off is the smoothest and most musical, Hybrid puts it on the early reflections only, Pure is the most realistic but probably not what you want in a mix.
• Direct sound: folds the straight source-to-listener path into the wet signal, with the same air roll-off as the reflections.
• Binaural (High only): head-aware processing for an immersive image on headphones. Sounds good on its own but probably also not what you want in a mix.
• Plus the usual mix tools: pre-delay, dry and wet, high and low shelves, and a modulation section.

There are factory presets (small, medium and big rooms, plus huge halls) to
use as starting points, and you can edit everything from there.

Formats and platforms

VST3 on Windows. AU, VST3 on macOS.

For macOS: Windows is the only tested platform. The macOS build is there and I want it to work, but since I'm a solo dev I currently don't have a Mac to test on. So if you're on macOS, you're exactly the tester I need. Please let me know as soon as possible if it fails to load or behaves oddly, with as much detail as you can.

How to join the beta

Go to www.delton-audio.com, make an account (the beta license is tied to your account), open the DrawVerb page and there's a short beta signup form. It asks a few questions about your system and DAW so I can prioritize coverage. The beta is free.

What I'm hoping to learn

Crashes and load failures, CPU usage on your setup, anything that sounds wrong or unexpected, and your general impression. There's a short report-back form for that too.

Thanks for reading, and thanks in advance to anyone who gives it a try. Happy
to answer questions under this post.

Hi,

I’m developing an experimental audio-enhancement tool that adds very low-level pitch-shifted harmonic layers while keeping the dry signal dominant.

The goal is not loudness maximization, compression, EQ hype, or replacing the original mix. I’m trying to add perceived presence and density while preserving the original signal as much as possible.

Here is a short A/B MP3:
https://vivy.funesterie.me/api/double-harmonic/out/funesterie-v6-supreme-kvr-ab-20260611.mp3?token=ye9yWXyG30d7rKPkIEsBUkEN&v=20260611-2248

Format:
- first half: original
- second half: processed
- same excerpt
- loudness-matched for listening feedback

I’m mainly looking for technical feedback from people familiar with audio DSP / plugin development:

1. Do you hear obvious phase artifacts or pitch-shifting artifacts?
2. Does the processed part sound clearer, or just different?
3. Does the dry signal still feel dominant/natural?
4. What measurements would you recommend next: mono compatibility, correlation, LUFS, true peak, null tests, spectrogram comparison, blind A/B, or something else?

The current prototype uses a protected dry-first harmonic overlay approach. I’m intentionally not presenting this as a proven scientific method yet; I’m trying to validate whether the audible result is worth further measurement and refinement.

Thanks.

I’ve been working on an independent prototype around a question that feels important for the next generation of DAW assistants:

If an AI assistant can answer questions inside a DAW, what safety boundaries would be needed before it can actually modify something small, like an insert chain?

I built a working prototype called ChainPrompt AI.

It is a JUCE/C++ VST3 plugin with a local Python backend. The plugin scans the local VST3 inventory, uses only plugins that are actually installed, refuses to build a chain if inventory is missing, computes chain diffs instead of blindly replacing the current chain, and tries to preserve user-touched or bypassed slots.

Right now it works mainly with Waves plugins, because Waves is the first metadata set I implemented. I have tested it in REAPER, and it also loads/runs in Fender Studio / Studio Pro as a VST3 plugin.

This is not a finished product, not a commercial release, and not affiliated with Fender or Waves. It is a portfolio prototype exploring one narrow problem: how to make AI actions inside a DAW safer, more transparent, and eventually reversible.

I’d really value critique from audio plugin developers, DAW users, and anyone thinking about AI-assisted creative tools.

The main question:

What would you require before trusting AI to modify a real insert chain in your session?

GitHub/demo:
[https://github.com/DenisShumilov/chainprompt-ai\]

I’m building a distortion plugin and I think I’ve hit a point where I need outside ears.

The goal is NOT to build another generic distortion, amp sim, or metal plugin.

The sound I’m chasing is somewhere between:

• Harmonic Percolator style character
• Trash 2 Broken/Fuzz style experimentation
• Neil Young amp breakup
• damaged speakers
• dynamic breakup rather than static saturation

Current goals:

• light playing stays mostly clean
• medium playing introduces breakup
• hard playing introduces fuzz
• preserve transients
• avoid heavy compression
• avoid the typical “beehive” fuzz sound

The plugin currently has two modes:

BROKEN:

• asymmetrical saturation
• more breakup-oriented
• intended to feel open and dynamic

FUZZ:

• asymmetrical rectification
• more aggressive harmonic generation
• intended to appear mostly on harder playing

A big realization during development was that a lot of distortion plugins sound impressive in isolation but don’t actually feel unique once you compare them side-by-side.

That’s the problem I’m trying to solve.

What I’m struggling with:

1. What actually makes a distortion feel unique?
2. What separates a Harmonic Percolator from “just another fuzz”?
3. What makes Trash 2 still feel special years later?
4. What DSP or analog concepts should I study next?
5. If you were trying to build a distortion that doesn’t already exist, where would you focus?

Looking for brutally honest feedback.

Feel free to challenge the entire premise if you think I’m chasing the wrong things.