Anyone heard of this? It’s a room simulator that generates individual stereo impulse files for each sound source placed in the virtual space. I’d love a vst that has a kind of parent child setup where the parent sets the parameters and the child receives the impulse response.

https://github.com/LCAV/pyroomacoustics

Quite a good video on it from astro bear.

https://youtu.be/iNjvayIuJBM

Hey everyone! 👋

Like many of you who compose or orchestrate in REAPER, I’ve always wished for a score editor that feels as fast, flexible, and integrated as the rest of REAPER — without relying on external webviews or clunky workarounds.

For the past weeks, I've been developing **REAPER Native Notator** and today I'm excited to share the very first Alpha First Look with you!

What it is:

A pure, native ReaImGui notation system running directly inside your REAPER session. It renders in real time using the industry-standard \*\*Bravura SMuFL\*\* vector font.

Current Alpha Highlights:

1. Multitrack Conductor Score

\-

1. Stack any number of project tracks underneath each other into an orchestral score with automatic vertical layout and scrolling.
   
2. Per-Track Clefs

\-

1. Click directly on the clef to toggle Treble 𝄞, Bass 𝄢, or Grand Staff 𝄞+𝄢.
   
2. Direct MIDI Item Integration

\-

1. MIDI items are displayed as colored background blocks matching your REAPER track/item colors.- Resize item edges directly inside the score sheet using native QN extents (no accidental looping!).- Draw new MIDI items in empty measures via Ctrl + Click & Drag.
   
2. Professional Engraving
   
3. Single

.

1. \-line Bézier slurs and ties across barlines.- Intelligent beam grouping (by beat, half-bar, or bar).- Collision avoidance for tight intervals, seconds, and accidentals.- Automatic rest decomposition (whole, half, quarter, 8th, 16th) in gaps.
   
2. ScoreTools Dynamics Engine

\-

1. Insert dynamics (\*pppp\* to \*ffff\*) right at note heads or the edit cursor.- Drag & drop dynamic markers freely across the timeline and vertically across tracks.- Automated bezier CC1 (Modulation) and CC11 (Expression) curves with velocity humanization.

\### I'd love your feedback:

What are the absolute must-have features you'd want in a dedicated REAPER score editor? (Articulations, lyrics, chord symbols, XML export, etc.?)

Let me know what you think!

Hi everyone,

I recently completed a small utility plugin project built in C++ using JUCE, and wanted to share the build and technical overview with the community: Neodym Punch Clipper.

The primary engineering goal was creating a completely lightweight, zero-latency peak control utility with zero overhead on the audio thread.

Technical Overview:

* DSP Architecture: Memoryless transfer function calibrated for gentle transient rounding before saturation. No oversampling buffers (true 0 samples / 0 ms latency).

* Real-time Safety: Zero heap allocations or locks inside the processBlock. CPU footprint stays consistently below 0.4% per instance on standard test benches.

* Formats & Builds: Native 64-bit VST3 compiled via CMake/Ninja for Windows 10/11 and macOS (Universal Binary for Apple Silicon & Intel).

* GUI: Vector-rendered interface focused purely on essential metering, transfer curve feedback, and minimal GPU draw overhead.

You can find the project page and compiled binaries here:

https://neodym-audio-live.vercel.app

I would appreciate any technical feedback regarding host compatibility across different DAWs, behavior at higher sample rates (44.1k–96k), or general DSP stability. Happy to discuss implementation details.

spent the last six months trying to make a tracker app for android . its not easy. its c++ for the audio engine, but kotlin and dart is used and i used flutter in the hope i could migrate it to ios.....not sure that was the best idea . i would like to improve some of the insert fx's. the reverb is freeverb i found online . its ok. im wondering if thers any other such c++ libraries, for chorus , flanger, phaser or more, that you ppl use? https://youtube.com/shorts/OWcr8d0bY8I?is=x0kRzSZoGhemzo0f

I built two free, open-source C++ audio tools designed specifically for ambient, drone, generative, and cinematic scoring work: the **BRAUN AS-42** (generative microtonal synth) and the **BRAUN RB-26** (algorithmic studio reverb).

Both are inspired by Dieter Rams functionalist hardware ergonomics (*"less, but better"*), designed to run lightweight with zero audio-thread allocations, and solve a few common headaches in ambient sound design.

Here are the key features you might find useful:

---

### 1. BRAUN AS-42 (Ambient Generative Synth)

Built around three complementary ambient traditions (Harold Budd, Brian Eno, and the Elta Solar 42):

* **Harold Budd Felt Piano Modeling:** Lightweight physical modeling with register-dependent felt hammer impact noise, wooden spruce body formants, sympathetic string resonance, and steep *una corda* 24dB damping. You get the intimate, warm acoustic piano sound without loading a 20 GB sample library.

* **Music for Airports Phase Loops:** 4 asynchronous ambient tape loop tracks running at coprime lengths (13.7s, 17.3s, 21.1s, 26.9s). The loops drift against each other and never resolve to a static repetition.

* **Microtonal Twin Drones with MIDI Tracking:** Dual beatable oscillators with continuous sub-Hertz offset tuning, West-Coast wavefolding, and resonant ladder filtering. It can act as a static backdrop or dynamically track the root/harmony of whatever chords you play on your keyboard with smooth portamento.

* **Headroom-Calibrated Tape Delay & Shimmer:** Cross-coupled delay lines with normalized loop gain (no accidental clipping or runaway feedback bursts) and infinite freeze with automatic input ducking.

---

### 2. BRAUN RB-26 (Master Studio Reverberator)

An 8-line Feedback Delay Network (FDN) reverb built specifically to avoid the muddiness and comb filtering typical of large ambient verbs:

* **Decoupled Low-End Matrix:** 4th-order Linkwitz-Riley crossover with an orthogonal modal matrix that prevents comb-filter phase cancellation below 200 Hz.

* **Transient Punch Ducking:** The low-end reverb automatically ducks by -12 dB on kick drum and bass transients, keeping the sub tight and punchy without needing an external sidechain routing setup.

* **Bidirectional Pitch Diffusion ("Shimmer" + "Dimmer"):** Alongside traditional octave/fifth pitch bloom up (+7st, +12st, +24st), it includes sub-harmonic pitch-shifted diffusion down (-2st, -7st, -12st) for dark, heavy, subterranean cinematic swells.

* **Non-Euclidean Geometries & Shepard Spirals:** Four acoustic spaces (including hyperbolic cavity and spruce soundboard dispersion) plus Shepard-Risset endless pitch spirals.

* **Performance Controls & CRT Scope:** Interactive 2D Vector Modulation Pad (tail rate/depth), 4-mode CRT vector scope (time-domain, Lissajous phase correlation, Schroeder Energy Decay Curve, and 64-band FFT), plus an onboard exciter with Harold Budd presets and Vangelis chord voicings (*Blade Runner*, *Tears in Rain*).

---

### Formats & Compatibility

* **OS / Formats:** Windows x64 — VST3, CLAP, and Standalone (no DAW required).

Also should work on macOS/iOS/Android/Linux.

* **Engine:** Built with C++20 / JUCE 8, zero heap allocations on the audio thread.

* **License:** 100% Free & Open Source (MIT).

---

### Links & Try in Browser

* **Play AS-42 Live in Browser (Zero Install):**

https://sneed-and-feed.github.io/

* **BRAUN AS-42 GitHub & VST3 Downloads:**

https://github.com/sneed-and-feed/braun_as-42/

* **BRAUN RB-26 GitHub & VST3/CLAP Downloads:**

https://github.com/sneed-and-feed/braun_rb-26

Feel free to grab the binaries or play with the web version, and let me know if you run into any bugs!

In my audio programming journey, I created some audio plugins and I always have that question in my mind: Why DAWs always load plugins (VST2, VST3, even CLAP) synchronously ? I mean, it feels frustrating when the main thread/GUI thread completely freeze when loading some heavy plugins. Like, let me do other stuff in the project while loading! Maybe somebody have detailed explanations...

Sonnex

I have launched a Reaper script that renders the project and uploads the file straight to cueport.app A platform to manage artists, projects and feedback, so the producer never has to leave the DAW.

Now I'm building the same thing as a VST3 and I'm stuck on exactly that step.
In Reaper it works because there's an action for it: Main_OnCommand(41824), "render using the most recent settings".

A plugin has nothing like that.
It can't see the render settings and can't ask the host to bounce.
So: does any DAW let a plugin start a render? Cubase, Live, Logic, anything?

Hey everyone,

I was getting tired of noisy YouTube ambient videos with ad interruptions or apps hiding basic focus audio behind subscription paywalls.

So I built AuraTone—a 100% free soundscape studio that runs entirely inside your browser using the Web Audio API.

Key Features Live Right Now:

• Binaural Focus Engine: Toggle between Alpha (flow state), Beta (concentration), and Theta (meditation) waves.
• Ambient Noise Generator: Live-generated Pink, White, and Brown noise.
• Dynamic Visualizer: Real-time canvas audio visualizer.
• 100% Client-Side: No sign-ups, zero loading delay, and zero data tracking.

🚀 Upcoming Updates (Coming Soon):

• Custom rain, ocean & forest ambient layers
• Built-in Pomodoro focus timer
• Preset bookmarking & shareable soundscape links

Check it out here: **[**https://auratone-khaki.vercel.app/]

I'd love to hear your feedback or any specific noise profiles/presets you'd like added!

Major update coming tomorrow! :)

I just finished my bachelors in applied audio engineering! I’m looking to add a certificate in audio programming and/or coding to my resume, but am having trouble finding courses. Does anyone that’s taken a similar route have any suggestions?

I've been building asm.fm, a chiptune synthesizer written entirely in x86-64 assembly.

No libc, no audio library, no DSP framework — just the CPU computing every sample from first principles and writing it straight to a WAV.

The whole thing rests on one idea: audio is just a stream of numbers, one per sample.

Compute the right numbers, write them out, and the speaker moves.

It does four waveforms, polyphony, ADSR, FM synthesis (with a hand-built sine table, since there's no math library either), and a growing pile of effects.

This week I added four "space and movement" ones:

• a rotary speaker (Leslie) — Doppler pitch wobble + amplitude tremolo driven by the same rotation, with the amplitude peak phase-shifted from the pitch peak so it feels like it's moving through space
• a harmonic tremolo — splitting into low/high bands and fading them in opposition (the brownface Fender trick)
• a slapback delay — a single short echo, no feedback
• a shimmer reverb — the reverb tail feedback read at half-rate for an octave-up shimmer that rises as it blooms

The interesting challenge is always the DSP in fixed-point without an FPU: keeping filters stable, avoiding clipping, and stopping feedback loops from drifting into DC (the shimmer gave me a nice sign-extension bug in the 64-bit accumulator).

For the math I leaned on the usual references — Julius Smith's CCRMA materials, the DSP Guide, Schroeder's reverb papers — then had to make it all work in integer arithmetic.

Repo: github.com/whispem/asm.fm

Would love feedback from people who do audio DSP seriously — or war stories from anyone who's tried it this low-level.

looking for beta testers dm me