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FIR filter design problem

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Hi,

why FIR?
With IIR it´s easy.

With digital filter one frequency information (0..40Hz) is not enough. you additionally need sampling frequency.

Klaus
 
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?

thanks
 
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 wanted to know how FIR LPF filter are derived

thanks
 
If you want theory not practical implementation, then you should say so from the start.

To start get a book on DSP. You can get a free ones here. I'm sure you can find all the math and theory to satisfy your derivation desires.
 
This is a good book about DSP fundamentals, too
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"
 

i wanted to know how FIR LPF filter are derived
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
 

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