Sounds like an awesome idea – a real‑time remix engine that keeps the crowd moving. First thing you’ll want is low‑latency audio handling, so look at libraries like PortAudio or WASAPI for Windows, CoreAudio for macOS, or JACK for Linux. That gives you a buffer you can stream to the stage speakers with sub‑10 ms delay. Next, think about the data that will feed the engine. Live input from a DJ’s deck, or maybe from an array of microphones, or even a visual sensor that captures the crowd’s energy. You’ll need a lightweight pipeline: capture → pre‑process (FFT, beat‑detect) → generation → output. For the generation part, a simple rule‑based system that modifies chords and rhythm on the fly can be surprisingly effective. If you want more creativity, a tiny neural net on the edge – like a small LSTM or a diffusion model tuned for music – could add fresh riffs every few bars. Don’t forget to expose a simple UI for the performer to tweak parameters on stage – maybe a touch screen or a MIDI controller that maps to key parameters. Finally, test with a real audience as soon as possible; real‑world feedback is the best debugger. Let me know if you want deeper dives into any of those components.

Alright, let’s get that audio loop up. First, add PortAudio to your project – grab the headers and libraries from the official site, or use a package manager like vcpkg or Conan if you’re on Windows, Linux, or macOS. Then write a minimal callback that reads a small buffer from the input device, processes it, and writes it straight to the output stream. Keep the buffer size small, say 256 or 512 samples, to stay below the 10 ms latency threshold. Test the loop with a simple pass‑through: copy input to output and confirm you hear the sound without noticeable delay. Once that’s stable, we’ll plug in your rule‑based logic and later the LSTM. That should give us a solid foundation for the live remix engine. Let me know when you hit the first beat.