Help - Custom Filters

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Kássio Rezende

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Hi everyone

I started dsp this semester and I need some help. I have to design a custom filter from a given signal, that represents this filter in frequency domain. To get started (as the book says: http://www.dspguide.com/CH17.PDF), I have to "convert the frequency domain to rectangular form". The problem is, I don't know exactly what this means. I'm using Freemat(http://freemat.sourceforge.net/#download) to write the code. This is what i have so far:

Code C - [expand]
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//This is not C, I'm using Freemat//
 
clear all;
clc;
 
test=0:1024;
 
signal_filter = [0:1024];
 
a=linspace(0,3*pi,205);
b=0.25*sin(a)+0.5;
signal_filter(1:205)=b;
 
c=linspace(0,195,196);
d=(2/195)*c+0.5;
signal_filter(206:400)=d(2:196);
 
signal_filter(401:554)=1;
 
signal_filter(555:595)=0;
 
signal_filter(596:800)=1;
 
e=-224:0;
f=((1-.10)/(224*224))*e.*e + .10;
signal_filter(801:1025)=f(1:225);
 
plot(test,signal_filter);
xlim([0, 1024]);


Hope you can help.

Farewell!
 
Last edited:

Hello.
I am unable to understand what you want to ask. I guess you are asking on developing a finite impulse response filter.. what you need is get the filter coefficients.. if the coefficient is in time domain, you just need to convolve with the signal you need to filter..
 

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