How audio work in computer

There's algoriths such as (Fast) Fourier Transformation which can help you with that. You give it some waveform signal on the input (a few milliseconds for example) and it gives you a list of frequencies/amplitude in return. You can then read and change the amplitude of those frequencies, then do the reverse transformation to get some waveform signal back and you have your equalizer 🙂

Note: The more data you feed your FFT, the more precise your equalizer gets (you get more detailed frequency ranges) but the more latency you have of course. FFT simply reads a list of levels for each period of time, and it outputs a list of amplitudes of each frequency. You then don't work with time (as a waveform does) but with frequency (as a RTA does).