[AtomicOrbital]

A intuition for creating a lowpass filter ... audio happens in the time domain ... Fourier transform outputs the frequency domain representation of the audio you feed it ... number of audio samples you feed into the FFT call determines the element count of the array returned ... now you have an array of your audio in the frequency domain ... for lowpass crank down to zero array elements above the first quarter of them ... this silences those frequencies ... to render the audio ( generate sound ) feed your edited frequency domain array into an IFFT ( inverse Fourier transform )

[samdafi]

This does work, but the sound of it isn’t super pleasing and it limits your filter design options. Checking out time domain IIR filters and methods for generating coefficients gets you flexibility and efficiency with no latency. I posted some links above but the RBJ Cookbook is a good place to start.