MATLAB CODE TO DESIGN A CHEBYSHEV I LOWPASS FILTER

CODE:

clear all

alphap=1;  %passband attenuation in dB

alphas=15; %stopband attenuation in dB

wp=.2*pi; % passband freq. in radians

ws=.3*pi; % stopband freq. in radians

%to find cutoff freq. and order of the filter

[n,wn]=cheb1ord(wp/pi,ws/pi,alphap,alphas); %syatem function of the filter

[b,a]=cheby1(n,alphap,wn);

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MATLAB CODE TO DESIGN A ELLIPTICAL LOW PASS FILTER

CODE:

clc;

clear all;

N = input('Order = ');

Fp = input('Passband edge frequency in Hz = ');

Rp = input('Passband ripple in dB = ');

Rs = input('Minimum stopband attenuation in dB = ');

% Determine the coefficients of the transfer function

[num,den] = ellip(N,Rp,Rs,2*pi*Fp,'s');

% Compute and plot the frequency response

omega = [0: 200: 12000*pi];

h = freqs(num,den,omega);

plot (omega/(2*pi),20*log10(abs(h)));

xlabel('Frequency, Hz'); ylabel('Gain, dB');

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MATLAB CODE TO DESIGN A BUTTERWORTH BAND REJECT FILTER

CODE:

clear all

alphap=2;  %passband attenuation in dB

alphas=20; %stopband attenuation in dB

wp=[.2*pi,.4*pi]; % passband freq. in radians

ws=[.1*pi,.5*pi]; % stopband freq. in radians

%to find cutoff freq. and order of the filter

[n,wn]=buttord(wp/pi,ws/pi,alphap,alphas) %syatem function of the filter

[b,a]=butter(n,wn,'STOP')

w=0:.01:pi;

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MATLAB CODE TO DESIGN A BUTTERWORTH HIGHPASS FILTER

CODE:

clear all

alphap=.4;  %passband attenuation in dB

alphas=30; %stopband attenuation in dB

fp=400; %passband freq. in hz

fs=800; %stopband freq. in hz

F=2000; %sampling freq. in hz

omp=2*fp/F;oms=2*fs/F; % To find cutoff freq. and order of the filter

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MATLAB CODE TO DESIGN A BUTTERWORTH BANDPASS FILTER

CODE:

clear all

alphap=2;  %passband attenuation in dB

alphas=20; %stopband attenuation in dB

wp=[.2*pi,.4*pi]; % passband freq. in radians

ws=[.1*pi,.5*pi]; % stopband freq. in radians

%to find cutoff freq. and order of the filter

[n,wn]=buttord(wp/pi,ws/pi,alphap,alphas) %syatem function of the filter

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MATLAB CODE TO DESIGN A BUTTERWORTH LOWPASS FILTER

CODE:

clear all

alpha=.4;  %passband attenuation in dB

alphas=30; %stopband attenuation in dB

fp=400; %passband freq. in hz

fs=800; %stopband freq. in hz

F=2000; %sampling freq. in hz

omp=2*fp/F;oms=2*fs/F; % To find cutoff freq. and order of the filter

[n,wn]=buttord(omp,oms,alpha,alphas) %syatem function of the filter

[b,a]=butter(n,wn)

w=0:.01:pi;

[h,om]=freqz(b,a,w,'whole');

m=abs(h);

an=angle(h);

subplot(2,1,1);plot(om/pi,20*log(m));grid;

ylabel('gain in dB');

Xlabel('NORMALISED FREQUENCY')

subplot(2,1,2);plot(om/pi,an);grid

ylabel('phase in radians')

xlabel('noramlised frequency')

OUTPUT:

n =4

wn =0.5821

b =  0.1518    0.6073    0.9109    0.6073    0.1518

a =1.0000    0.6418    0.6165    0.1449    0.0259

 

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Matlab code for Computation of Linear convolution with DFT

Code:

clc;

clear all;

g=input('Type in the first sequence:');

h=input('Type in the second sequence:');

ga=[g zeros(1,length(h)-1)];

ha=[h zeros(1,length(g)-1)];

G=fft(ga);

H=fft(ha);

Y=G.*H;

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MATLAB CODE TO FIND OUT THE DFT & IDFT THEN PLOT MANITUDE AND PHASE RESPONSE OF IT

CODE:

clc;

clear all;

close all;

x=input('enter the data sequence');

N=input('enter the N point DFT');

fs=input('enter the sample freq.');

Y=fft(x,N)                       %this will give DFT

f=fs/N;

G=ifft(Y,N)                      %this will give IDFT

z=0:f:((length(Y)-1)*f);          %freq vextor for the x axis

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MATLAB CODE TO FIND OUT THE FREQUENCY AND PHASE RESPONSE

CODE:

clc;

clear all;

a=input('Enter the values for numerator : ');

b=input('Enter the values for denomator : ');

t = tf(a,b,0.1) %give transfer function in Z domain

freqz(a,b); % give freq as well as phase response

OUTPUT:

Enter the values for numerator : [1 2 3]

Enter the values for denomator : [4 5 6]

 Transfer function:

 z^2 + 2 z + 3

---------------

4 z^2 + 5 z + 6

 Sampling time: 0.1

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Matlab code for Addition of the signals frequnecy and amplitude value will be enter by user

Code:

clc;

clear all;

close all;

n=input('enter the no. of freq components :');

a=input('enter the amplitude matrix :');

f=input('enter the freq matrix :');

fs=input('enter the sampling frequency :');

disp(a); % display the amplitude componenets

disp(f); % display the frequency componenets

for i=1:1:n

    if (i+1)>n

        break;

    end

       if f(1,i)>f(1,i+1)

           f(1,i+1)=f(1,i); %find outthe maxium freq components

       end  

end

if fs<(2*f(1,n)) %check out the sampling theroem (fs>=2fm)

    fs=input('The fs entered was unadequate. Plz enter the required sampling frequency :');

end

z=0:(.1*fs):fs;

for i=1:1:n  %display the each cosine signal

    x=a(1,i)*cos(2*pi*(z/fs)*f(1,i));

    subplot(n+1,1,i);

    stem(z,x);

    grid on;

    hold on;

end

x=0;

for i=1:1:n  %addition of all signals

    x=(a(1,i)*cos(2*pi*(z/fs)*f(1,i)))+x;

    subplot(n+1,1,n+1);

    stem(z,x);

    grid on;

end

Output:

enter the no. of freq components :4

enter the amplitude matrix :[1 2 1 2]

enter the freq matrix :[5 6 7 5]

enter the sampling frequency :12

     1     2     1     2     5     6     7     5

The fs entered was unadequate. Plz enter the required sampling frequency :16

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