# FIR filter design problem

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#### rameshrai

##### Full Member level 3
hi,
want to design FIR Low Pass Filter which allows to pass signal upto 0 to 40Hz, how to start designing FIR digital filter?

thanks

#### KlausST

##### Super Moderator
Staff member
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

matrixofdynamism

### matrixofdynamism

points: 2

#### rameshrai

##### Full Member level 3
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

matrixofdynamism

points: 2

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matrixofdynamism

### matrixofdynamism

points: 2

#### rameshrai

##### Full Member level 3
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

matrixofdynamism

### matrixofdynamism

points: 2

##### Super Moderator
Staff member
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.

matrixofdynamism

### matrixofdynamism

points: 2

##### Member level 4
This is a good book about DSP fundamentals, too

matrixofdynamism

### matrixofdynamism

points: 2

##### Super Moderator
Staff member
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"

#### FvM

##### Super Moderator
Staff member
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

##### Member level 4
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"
So, two votes for Dr Smith's book

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