Here's something you might find interesting. Have you heard of the ? It tells us that we can transform a continuous signal into a discrete signal if we sample at twice the frequency of the signal. Just because something is continuous doesn't mean it can't be represented digitally and the theorem tells us how to do that.
All frequencies are based on the time it takes for its wavelength to complete a cycle. All EM signals are composed of a multitude of different frequencies overlaying each other to create a waveform. which is represented by that seemingly randomly jagged line you see with audio files.
https://files.catbox.moe/99zj2c.png
Using Fourier transforms, a waveform can be broken down into its individual frequencies. All waveforms are just layered frequencies. Multiples of a frequency add to that frequency's amplitude. A frequency 180 degrees out of phase of another one eliminates them both completely (wave cancellation).
So you can take the random-looking up-and-down waveform of an audio sound (or any other type of waveform) and reduce it to its constituent frequencies to see how it's made up. The highest frequency will have the most number of cycles (shortest time intervals). The lowest frequency will determine the longest time intervals of the overall signal (often used as a carrier wave). Doubling the highest frequency provides a means of getting a purer quality waveform when converting to digital because you're breaking the signal down into twice as many component parts.
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