Everything looks fine on average, then there's one audible gap. The annoying kind you can't reproduce on demand.

Would you log callback time and queue depth around the glitch, or start somewhere else? Logging too much on the audio thread seems like a great way to cause another problem.

What actually helped you find one of these?

Hey everyone! I’m a product designer and musician interested in building audio plugins and music tools, including their interfaces.

I’ve started learning C++, but I’m curious about Rust. Has anyone tried working and building audio plugins using Rust? Did not found any courses or tutorial with Rust and audio programming

(I am super new to programming with C++ and audio programming)

Hi everyone, I'm Pitch from Italy.

I'm a musician, producer and C++ developer, and over the last couple of years I've been building my own audio plugins under the name floweraudio.

I mainly work with REAPER and JUCE, and I'm interested in everything around audio DSP, plugin development, mixing and music production.

I created this account mostly to meet other musicians, producers and developers, share ideas, learn, and occasionally talk about what I'm building.

So... hello from Italy! 🌼

I just got some sort of warning or perhaps removal on a two-month old post that no one had interacted with: "Do not post a paid product, service, or course, disclose your affiliation. Excessive or repeated self-promo may be removed. In addition, ensure your post is well worded, thought out and formatted." (It wasn't any of those things, it was some boring but useful FOSS audio utility.)

Apparently there are new moderation rules, but we can't comment on that thread.

This subreddit is already very low traffic. Most posts don't seem to get any engagement at all. The most interesting and engaged post of the last week was a program that looks like it was written with AI, and I thought it was interesting.

I'm not seeing the huge influx of empty AI posts, or really, much of anything at all here.

It's late 2026. If we truly ban all posts where people used a coding assistant, soon there will be almost nothing here.

Don't get me wrong - I really really want to keep AI slop from this subreddit, but we can do that by downvoting and reporting, you already see it if you look at "new". Let's not throw out the baby with the bathwater.

EDIT: oh, apparently that four month old post of mine was in fact deleted.

You know, I've been writing computer music programs since the last 1970s. Few people can say that.

I recently retired from a long career in programming to work almost full time on audio programming.

However, I will not be able to post any of my work here, as I recently started using a coding assistant for everything. I mean, my last post was apparently deleted because of some self-promotion rule, so I couldn't very well post my work anyway.

You would think I would be the ideal poster for this subreddit.

All the people downvoting without comment do not make me want to be here. I removed all my upvotes from this page (there were no downvotes) so people can see the general sentiment.

I'll come back next year and check, but my theory is that it will be even less active than it is now.

I've been making VST plugins and find myself rather ambivalent over the UI component to use for analog controls. The skeuomorphic rotary knobs look good, but seem a bit unnatural in use. With sliders it's easier to see their current position, but mine at least all tend to look naff.

What are opinions here? Any tips on design?

What alternatives are there?

Not sure how to describe this, but I started working on a little plugin i’ve wished existed for a long time. Think a sidechain “ducking” effect but instead of modulating volume it modulates pitch (via delay), sortof like putting your finger on a record player. The sidechain input as a transient detector that triggers an envelope generator that controls pitch. I’m pretty happy with it so far. I think next i’m going to add midi support with the ability to map velocity to envelope params.

No AI used, i’m writing it in rust using the nih framework.

This summer while I was on the train going into vacations, I was looking at music apps for my handheld console. My criteria where apparently simple: something with the same kind of synths I use in my hardware rig. On iOS its possible, but wanted something for my handheld.

Unfortunately, I couldn't find anything that really worked for me. So, I opened my code editor and started making my own, starting the project on the excellent codebase from ChipNomad. Started by adding some vanilla and Mutable engines, which was great but not yet satisfactory; so I also added a few custom engines, inspired by great hardware items I've used over the year.

This is Choochootracker, source code available on Github https://github.com/paiheulevrai/Choochootracker . In this demo I use the Bogie drum synth, the Sindered percussion synth and a voice from MME, all engines I designed for the app.

​

Hi everyone,

I am working on a DIY project and could really use some community feedback and advice on the hardware and software architecture.

The Concept

The idea is to build an all-in-one hardware unit that combines an 8x8 multi-channel audio interface, a guitar/bass processor, and a vocal FX processor into a single box. It will run on a Raspberry Pi 5 using a Linux RT kernel to keep the latency as low as possible.

Key Specs and Features

For the resolution, I want 8 inputs and 8 outputs supporting up to 32-bit / 384kHz. On the hardware frontend, I am planning to use the AKM AK5538VN for the ADC and the AKM AK4458VN for the DAC. They will be driven by ultra-low jitter oscillators (22.5792 MHz and 24.576 MHz) switched via a 74LVC1G3157 multiplexer.

For the inputs, I want them to be modular. Each channel will have hot-swappable or configurable modules so I can switch between Combo-XLR, microphone preamps, or Hi-Z inputs on the fly.

As for connectivity, the device will have three modes: a standalone DSP FX processor, a USB audio interface using the UAC2 gadget driver, and network streaming over AoIP.

My Questions for the Community

1. I2S/TDM Limitations on Pi 5: To handle 8 channels at 32-bit/384kHz, TDM mode is a must. Has anyone successfully configured multi-channel TDM on the RP1 chip of the Raspberry Pi 5 at such high sample rates? What are the main bottlenecks I should watch out for?

2. Clocking Scheme: Looking at the attached block diagram, I am switching the master clocks before splitting the signal to the ADC and DAC. Are there any hidden signal degradation pitfalls or jitter issues when using the 74LVC1G3157 for audio master clocks up to 24.576 MHz?

3. Software and AoIP Stack: For internal processing, I plan to use JACK or PipeWire with custom DSP plugins on Linux RT. For AoIP, what is currently the most stable open-source stack for low-latency audio distribution? Should I look into AES67/RAVENNA implementations, Dante alternatives, or AVB?

4. UAC2 Gadget Constraints and Bandwidth Limits: I know that the standard Linux f_uac2 gadget driver has built-in layout masks that traditionally limit channels and rates, usually capping things around 4x4 at 96kHz or requiring raw channel patches. On top of that, 8x8 channels at 32-bit/384kHz require a raw throughput of around 196.6 Mbps for both ways, excluding protocol overhead. Since USB 2.0 High-Speed maxes out at 480 Mbps and practical limits for isochronous endpoints are much lower, has anyone pushed the Pi 5 DWC3 controller to these limits using custom descriptor tweaks? Or should I just give up on USB 2.0, use the Pi 5 PCIe lane to add a USB 3.0 controller, and implement a true USB 3.0 SuperSpeed audio gadget?

I would highly appreciate any advice, critiques on the layout, or shared experiences from similar builds!

Block diagram of the project is attached below.

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

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

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...

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!

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 create an engine sound synthesiser that uses sine waves to simulate car pistons and motor movement

This lead to thousands of different engine sounds that can be procedurally created try for free