This document provides an introduction to audio processing using microcontrollers, explaining concepts such as sound as a mechanical wave and the role of loudspeakers in converting electrical signals to sound. It discusses sampling theory, specifically the Nyquist theorem, and how to convert analog sounds into digital formats like .wav files using pulse code modulation. Additionally, it outlines steps to analyze audio data with tools like Audacity and highlights limitations of microcontroller memory in handling audio samples.