I'm trying to build VoIP into my browser-based game, and holy shit are sound processing people bad at explaining stuff.
Every stackoverflow answer has badly named variables, noone names the algorithms they're using (which makes research near impossible), and literally every single Web Audio API pipeline I have seen so far contains at least one unexplained effect with no parameters, but it's a different effect each time.
One guy had implemented some kind of smoothing for catching up with the stream after interruptions (where the playback speed is proportional to how far we're behind the intended latency), without ever mentioning it anywhere. And this is meant to be a basic example!