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?
thats the problem, i only find websites that allows user to pick freqs and other specification and get the result. Or, simply starts of digital filter from analog specification.
I guess you didn't follow the link I provided in post #6, the first link on that page is the book "The Scientist and Engineer's and Guide to Digital Signal Processing"
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
I guess you didn't follow the link I provided in post #6, the first link on that page is the book "The Scientist and Engineer's and Guide to Digital Signal Processing"