Is the initial programming setup really that complex?

For these kinds of embedded audio boards, the C++ stuff is usually unavoidable if you want fine control over the hardware, but fighting the compiler isn't the only path. If the goal is just to hear cool sounds fast, you need to bypass the raw register manipulation for the controls part and focus on high-level abstractions. Many of the development platforms have simplified libraries designed specifically for basic synthesis; these libraries handle the encoder input mapping and frequency calculation behind the scenes, letting you focus on the wave generation itself. You are likely looking at a situation where the standard examples are meant for showing how the low level works, not how fast you can prototype. Instead of trying to write a full digital filter from scratch, which is where the C++ complexity blows up, start with simple waveforms. Generate a basic sine wave or triangle wave first. Most audio boards have built-in digital signal processing routines that make generating a simple oscillator incredibly quick. You just feed the parameters (like frequency and amplitude) to the function, and the board handles the math for you. This is far simpler than managing every single bit of the audio output buffer manually. Once you have a stable tone, you can introduce the controls. Don't start by trying to implement a complex LFO or a full envelope. Just connect the encoder to control one parameter, like the pitch (frequency) of the sine wave, or the volume. This keeps the code small and focused on one input mapping. You'll find that the most intimidating code block is often the one trying to do too many things at once.

If you want a true higher-level language experience, you might need to check if the board manufacturer offers a wrapper or an alternative programming enviroment that uses a simpler API, perhaps something that looks more like a function-call structure rather than bare metal C++. But generaly, even in C++, the libraries themselves are what provide the easier entry point into sound design; they are the abstraction layer over the confusing hardware details.

The C++ wall is real, but sometimes the code isn't the problem, it's just a lack of visibility. I struggled with encoder input mapping for ages until I grabbed a cheap digital scope. Seeing the raw output buffer visually made the whole difference. Instead of hunting through endless code to figure out why the frequency wasn't moving right, I could just watch the voltage levels change as I turned the knob. It helped pinpoint exactly where the signal was getting lost in the abstraction layers.

Ha, this takes me back. Before anyone had nice libraries for this stuff we were soldering pots to parallel ports and writing asm by hand, so count your blessings. One thing I'd suggest: mock the whole thing up in a visual patching environment on your PC first. Pure Data or Max, even old Reaktor. Build your oscillator, wire the encoder logic as a slider for now, get the sound happening where debugging takes seconds not compile cycles. Then port the logic over to the board bit by bit. That's how I've prototyped half the gadgets in my shop for twenty years, and it saves you staring at a compiler for weeks. The thinking transfers, only the syntax changes.

Update: took the suggestion to mock things up in Pure Data first and it's been great for figuring out what I acutally want the controls to do. Still lost on the hardware side though. When I move the patch over to the board, the encoder example code is still a wall to me. Is there a middle step, like a stripped-down example that just maps one knob to one parameter?