ITT a whole lot of whooshing <| sep |> haha thanks bro , didnt need to be a complete dick about it but it looks like your statement wasnt even correct hence why i asked . oh and btw for the record it was a legitimate question <| sep |> One of many Dsp books will explain fir filtering . In time domain it ' s convolution your input signal being multiplied and summed with a flipped version of the impulse response which is also the filter coefficients . In frequency domain your input signal and filter coefficients are transformed to frequency domain and then multiplied together . I suggest reading a few of those book chapters on fir and then implement in MATLAB or python . Those books also explain circular convolution and linear convolution and windowing techniques as well . I ' d probably buy think Dsp book . Its in python so you won ' t need to pay for it like you would with MATLAB .