Recently, I have been studying IIR and FIR filters, and trying to create a parametric equalizer using a microcontroller. More specifically, I use the ADC on it to sample the audio at about 44 kHz, and in between grabbing samples, I use the time to process the signal and pass it out on the DAC.
What is a bit confusing to me though is that this process seems to be pretty intense, and often takes most of the time in between samples to do all the processing (with 8 2-pole IIR bandpass filters).
Programs like Spotify, or devices like most Galaxy phones have similar equalizers but they are running on non-dedicated hardware, and I am really confused how they are able to pull this off? Does anybody know if they are basically doing the same thing just faster, or is there some secret sauce to this that I am missing?
I have never tested either of the previously mentioned equalizers in those platforms, so maybe they are just really crappy, but at least just by listening to them, it doesn’t sound horrible.