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let's say i want LPF to pass freq. from 0 to f1 and i have sampling frequency fs. How do i start designing FIR filter? Do i start with z transform H(z) or impulse response h(t)? if so how do i get them?
To get a basic idea, you can try a classical design method, called "direct synthesis":
- perform an inverse fourier transform of the intended frequency response. Assume a pure real-valued function for simplicity.
- apply a window (e.g. Hanning, Hamming, Blackman) to achieve a smooth decay of the pulse response borders