Continue to Site

Welcome to EDAboard.com

Welcome to our site! EDAboard.com is an international Electronics Discussion Forum focused on EDA software, circuits, schematics, books, theory, papers, asic, pld, 8051, DSP, Network, RF, Analog Design, PCB, Service Manuals... and a whole lot more! To participate you need to register. Registration is free. Click here to register now.

Register Log in

FIR filter design problem

Status
Not open for further replies.
R

rameshrai

Full Member level 3
Full Member level 3
Joined
Aug 16, 2010
Messages
158
Helped
7
Reputation
14
Reaction score
7
Trophy points
1,298
Activity points
2,272
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
 

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
 
You can try playing around with FIR filter design on the following website.


here is another one.
 
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.
 
wade_hassler said:
This is a good book about DSP fundamentals, too
Click to expand...
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
Click to expand...
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
 

ads-ee said:
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"
Click to expand...

So, two votes for Dr Smith's book
 

Status
Not open for further replies.

Similar threads

Part and Inventory Search

Welcome to EDABoard.com

Sponsor

Back
Top