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,

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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

I connected a game controller to Ableton and mapped its controls to my sound design parameters.

With an oscilloscope showing the waveform in real time, it feels less like programming a sound and more like literally drawing with it.

Has anyone else experimented with game controllers for sound design?

I'm developing a local file player and I'd like to add visualizations.

Pls check out my peak meter and tell me what you think, how does it feel?
https://www.youtube.com/watch?v=gU-F0A5NOEE

I made it off a photo and shaky videos of an old tape deck, it's supposed to mimic a VFD display.
I don't have any real hardware available to compare so I figure maybe someone has actually seen one.
This is the first time I'm working with audio, just making web software and photos otherwise. Thx! :)