I'm new to DSP and was thinking of sampling in the traditional music creation way, in that you capture a sound and play it back, warp it, etc., but when I thought about the Nyquist theorem, I realized that can't be what's happening when sound is being captured.
If we treated a sound clip that is 1 second long as its own waveform, then it's 1Hz and according to Nyquist you'd have to capture it at least at 2Hz. That's only 2 snapshots for 1 second of sound.
What are those 2 snapshots capturing? Let's say it's 16 bits of depth. What can that combination of 1s and 0s capture exactly? I've caught that the bit depth determines how many dBs you have to work with (more range), but I'm a bit lost on the rest. Is it capturing a unique waveform tone? Is it specifically a Sine wave tone? Is it multiple tones? I'm having trouble picturing what's being sampled and how it's being accurately played back.