DIGITAL SIGNAL

PROCESSING
(UEC502) LAB
ASSIGNMENT

Submitted by:
RishabdevDhawan(101806024)
ECE2
 INDEX
S. NO. EXPERIMENTS
Write a program to: a) Generate unit impulse, unit step and signum functions. b) Generate square
1 waves with user input frequencies and duty cycles c) Generate sinusoidal signal with user input
multiple frequencies, amplitudes and its phases.
Write a MATLAB program to determine linear convolution (without using inbuilt commands) of any
2 two discrete sequences.
Write a MATLAB program to: a) Determine the N-point DFTof the user input sequence (without
3 using direct command) and plot its magnitude and phase spectrum. b) Compute IDFT of the result
obtained in (a)
For any two user input discrete sequences a) Compute circular convolution b) Compute linear
4 convolution using circular convolution (as computed in (a)) c) Compute circular convolution using
linear convolution and verify with result obtained in (a). d) Compute circular convolution using DFT-
IDFT method
Write a MATLAB program to determine the time domain and frequency domain response of fixed
5 windows.
To design FIR Low Pass Filter using Rectangular and Triangular windows with user input
6 specifications for pass band edge frequency, transition width, stop band attenuation and sampling
frequency.
To design FIR Low Pass Filter using Hamming and hanning windows with user input specifications for
7 pass band edge frequency, transition width, stop band attenuation and sampling frequency
Write a MATLAB program to design FIR BANDPASS filter using Kaiser windows with the user input
8 specifications for pass band range, transition width, passband ripple, stop band attenuation and
sampling frequency.
Write a MATLAB program to design digital LOWPASS filter using Butterworth prototype with the
9 user input specifications for pass band edge frequency, stopband edge frequency, passband ripple,
stop band attenuation. Use impulse invariant method.
Write a MATLAB program to generate following sequence i) sinusoidal sequence of user input
10 normalized frequency ii) sum of two sinusoids with multiple normalized frequency. Perform up
sampling for both by factor 4 and down sampling by factor 5 on sequence of length 51.
EXP1

AIM:
Write a program to:
a) Generate unit impulse, unit step and signum functions.
b) Generate square waves with user input frequencies and duty cycles
c) Generate sinusoidal signal with user input multiple frequencies, amplitudes and its phases.

CODES:
a)
clc
clear all
close all
n=-20:4:20
u=sign(n);
v=(n>=0);
w=(n==0);
subplot(3,2,1)
stem(n,u)
xlabel('n');
ylabel('x[n]');
title('101806024 signum');
subplot(3,2,2)
stem(n,v)
xlabel('n');
ylabel('x[n]');
title('101806024 step');
subplot(3,2,3)
stem(n,w)
xlabel('n');
ylabel('x[n]');
title('101806024 impulse');

b)
clc
clear all
close all
sf=1000;
n=-0.05:1/sf:0.05
f=input('fr');
d=input('dutycycle');
u=square(2*pi*f*n,d);
subplot(2,2,1)
stem(n,u)
xlabel('n');
ylabel('x[n]');
title('101806024 square')
subplot(2,2,2)
plot(n,u)
xlabel('n');
ylabel('x[n]');
title('101806024 square')

c)
clc
clear all
close all
n=-1:0.01:1
f=input('freq');
amp=input('a');
phase=input('p')
u=amp*sin(2*pi*f*n+phase);
subplot(2,2,1)
stem(n,u)
xlabel('n');
ylabel('x[n]');
title('101806024 sinusoidal')
subplot(2,2,2)
plot(n,u)
xlabel('n');
ylabel('x[n]');
title('101806024 sinusoidal')
 EXP2
AIM:
Write a MATLAB program to determine linear convolution (without using inbuilt commands) of
any two discrete sequences.

CODE:
clc
clear all
close all
n=0:4;
x=(n>=0)-(n>=4)
subplot(2,2,1);
stem(n,x);
xlabel("n");
ylabel("x[n]");
title("conv 101806024");
subplot(2,2,2);
n=0:8;
y = n.*(n>=0)-2*(n-4).*(n>=4)+(n-8).*(n>=8);
stem(n,y);
xlabel("n");
ylabel("y[n]");
title("conv 101806024");
l1 = length(x);
l2 = length(y);
X=[x,zeros(1,10)];
Y=[y,zeros(1,6)];

for i=1:(l1+l2-1)
c(i)=0;
for j=1:i
c(i) = c(i)+X(j).*Y(i-j+1);
end
end

subplot(2,2,3)
stem(c);
xlabel("n");
ylabel("x[n]*y[n]");
title("conv 101806024");
 EXP3
AIM:
Write a MATLAB program to:

a) Determine the N-point DFTof the user input sequence (without using direct command) and
plot its magnitude and phase spectrum.

b) Compute IDFT of the result obtained in (a)

CODE:
x = [2 2 1 1 2];
N = length(x);
X = zeros(5,1);
for k = 0:N-1
for n = 0:N-1
X(k+1) = X(k+1) + x(n+1)*exp((-1i)*pi/2*n*k);
end
end
i=length(x);
r=zeros(5,1);
for a = 0:N-1
for b = 0:N-1
r(a+1) = r(a+1) + x(b+1)*exp((1i)*pi/2*b*a);
end
end
t=0:N-1;
subplot(2,2,1)
stem(t,X);
xlabel("Freq");
ylabel("|X(k)|");
title("dft Magnitude 101806024")
subplot(2,2,2)
stem(t,angle(X));
xlabel("Freq");
ylabel("Phase(inv)");
title(" dft Phase 101806024")
subplot(2,2,3)
stem(t,r);
xlabel("Freq");
ylabel("|inv(X)|");
title("idft Magnitude 101806024")
subplot(2,2,4)
stem(t,angle(r))
xlabel("Freq");
ylabel("Phase");
title("idft Phase 101806024")
 EXP4
AIM:
For any two user input discrete sequences

a) Compute circular convolution

b) Compute linear convolution using circular convolution (as computed in (a))

c) Compute circular convolution using linear convolution and verify with result obtained in (a).

d) Compute circular convolution using DFT-IDFT method

CODE:
clc;
clear all;
close all;
X=input('Enter the First Sequence'); %[5 7 8 9 5]
H=input('Enter the Second Sequence');%[2 9]
N1=length(X);
N2=length(H);
x=[X,zeros(1,N2-1)];
h=[H,zeros(1,N1-1)];
N = max(N1, N2); %length of circular convolution
for m = 1:N
y(m) = 0;
for n = 1:N
y(m) = y(m) + x(n)*h((mod(m-n, N)+1));
end
end
circonv = cconv(x,h,N) %verification
y %Result
ii)

clc;
clear all;
close all;
X=input('Enter the First Sequence'); %[5 7 8 9 5]
H=input('Enter the Second Sequence'); %[2 9]
N1=length(X);
N2=length(H);
x=[X,zeros(1,N2-1)];
h=[H,zeros(1,N1-1)];
N = N1 + N2 -1; %length of linear convolution
for m = 1:N
y(m) = 0;
for n = 1:N
y(m) = y(m) + x(n)*h((mod(m-n, N)+1));
end
end
linconv = cconv(x,h,N) %verification
y %Result
iii)

clc;
clear all;
close all;
X=input('Enter the First Sequence'); %[7 8 9]
H=input('Enter the Second Sequence'); %[1 5 2 7]
N1=length(X);
N2=length(H);
NL = N1+N2-1; %Length of linear convolution
N = max(N1, N2);%Length of circular convolution
y = conv(X,H) %Linear convolution
y1 = y(1:N); %first part
y2 = [y(N+1:NL), zeros(1, NL-N)]; %second part
for i = 1:N
C(i) = y1(i) + y2(i); %Calculating Circular convolution
end
C %Result
circonv = cconv(X, H, N) %Verification
iv) clc;
clear all;
close all;
X=input('Enter the First Sequence');%[1 2 3 4]
H=input('Enter the Second Sequence'); %[5 8 9]
N1=length(X);
N2=length(H);
%NL = N1+N2-1; %Length of linear convolution
N = max(N1, N2); %Length of circular convolution
y1 = fft(X, N); %DFT of sequeance 1
y2 = fft(H, N); %DFT of sequence 2
Y = y1.*y2;
C = ifft(Y) %Result
circonv = cconv(X,H,N) %Verification
 EXP5
AIM:
Write a MATLAB program to determine the time domain and frequency domain response of fixed
windows.

CODE:

a)rectangular
%RishabdevDhawan
M=input('Enter the value of M: ');
N=input('Enter the value of N: ');
w1=linspace(0,pi,1000);
t1=(0:1:M-1);
A1=ones(1,M);
stem(t1,A1);
xlabel('------> n ------>');
ylabel('Amplitude ------>');
figure
[H1,w1]=freqz(A1,1,1024);
wN1=w1./(2*pi);
HN1=abs(H1)./max(abs(H1));
plot(wN1,20*log10(HN1),'b')
grid
xlabel('Normalizied frequency --------->')
ylabel('Normalized Magnitude response (dB) ----->')
title('Frequency response of rectangular window function')
hold on
t2=(0:1:N-1);
A2=ones(1,N);
[H2,w2]=freqz(A2,1,1024);
wN2=w2./(2*pi);
HN2=abs(H2)./max(abs(H2));
plot(wN2,20*log10(HN2),'r')
hold off
legend('M=25', 'N=49');
b)Blackman
%RishabdevDhawan
M=input('enter the value of M: ');
N=input('enter the value of N: ');
t1=(0:1:M-1);
A1=0.42 - 0.5.*cos((2.*pi.*t1)./(M-1)) + 0.08.*cos((4.*pi.*t1)./(M-1));
stem(t1,A1);
xlabel('------> n ------>');
ylabel('Amplitude ------>');
figure
[H1,w1]=freqz(A1,1,1024);
wN1=w1./(2*pi);
HN1=abs(H1)./max(abs(H1));
plot(wN1,20.*log10(HN1),'b')
grid
xlabel('Normalizied frequency --------->')
ylabel('Normalized Magnitude response (dB) ----->')
title('Frequency response of Blackman window function')
hold on
t2=(0:1:N-1);
A2=0.42 - 0.5.*cos((2.*pi.*t2)./(N-1)) + 0.08.*cos((4.*pi.*t2)./(N-1));
[H2,w2]=freqz(A2,1,1024);
wN2=w2./(2*pi);
HN2=abs(H2)./max(abs(H2));
plot(wN2,20.*log10(HN2),'r')
hold off;
legend('M=25', 'N=49');
c)Hamming
%RishabdevDhawan
M=input('enter the value of M: ');
N=input('enter the value of N: ');
t1=(0:1:M-1);
A1=0.54 - 0.46.*cos((2.*pi.*t1)./(M-1));
stem(t1,A1);
xlabel('------> n ------>');
ylabel('Amplitude ------>');
figure
[H1,w1]=freqz(A1,1,1024);
wN1=w1./(2*pi);
HN1=abs(H1)./max(abs(H1));
plot(wN1,20.*log10(HN1),'b')
grid
xlabel('Normalizied frequency --------->')
ylabel('Normalized Magnitude response (dB) ----->')
title('Frequency response of Hamming window function')
hold on
t2=(0:1:N-1);
A2=0.54 - 0.46.*cos((2.*pi.*t2)./(N-1));
[H2,w2]=freqz(A2,1,1024);
wN2=w2./(2*pi);
HN2=abs(H2)./max(abs(H2));
plot(wN2,20.*log10(HN2),'r')
hold off;
legend('M=25', 'N=49');
d)Hanning
%RishabdevDhawan
M=input('enter the value of M: ');
N=input('enter the value of N: ');
t1=(0:1:M-1);
A1=0.5.*(1-cos((2.*pi.*t1)./(M-1)));
stem(t1,A1);
xlabel('------> n ------>');
ylabel('Amplitude ------>');
figure
[H1,w1]=freqz(A1,1,1024);
wN1=w1./(2*pi);
HN1=abs(H1)./max(abs(H1));
plot(wN1,20.*log10(HN1),'b')
grid
xlabel('Normalizied frequency --------->')
ylabel('Normalized Magnitude response (dB) ----->')
title('Frequency response of Hanning window function')
hold on
t2=(0:1:N-1);
A2=0.5.*(1-cos((2.*pi.*t2)./(N-1)));
[H2,w2]=freqz(A2,1,1024);
wN2=w2./(2*pi);
HN2=abs(H2)./max(abs(H2));
plot(wN2,20.*log10(HN2),'r')
hold off;
legend('M=25', 'N=49');
e)Bartlett

%RishabdevDhawan
M=input('enter the value of M: ');
N=input('enter the value of N: ');
A1=zeros(1,M);
t1=(0:1:M-1);
for i=0:M-1;
if (t1(1,i+1)>=0 && t1(1,i+1)<=(M-1)/2)
A1(1,i+1)= (2.*i)./(M-1);
elseif (t1(1,i+1)>(M-1)/2 && t1(1,i+1)<=M-1)
A1(1,i+1)= 2 - ((2.*i)./(M-1));
else
A1(1,i+1)=0;
end
end
stem(t1,A1);
xlabel('------> n ------>');
ylabel('Amplitude ------>');
title('Bartlett window ');
figure
[H1,w1]=freqz(A1,1,1024);
wN1=w1./(2*pi);
HN1=abs(H1)./max(abs(H1));
plot(wN1,20.*log10(HN1),'b')
grid
xlabel('Normalizied frequency --------->')
ylabel('Normalized Magnitude response (dB) ----->')
title('Frequency response of Bartlett window function')
hold on
t2=(0:1:N-1);
A2=zeros(1,N);
for j=0:N-1;
if (t2(1,j+1)>=0 && t2(1,j+1)<=(N-1)/2)
A2(1,j+1)= (2.*j)./(N-1);
elseif (t2(1,j+1)>(N-1)/2 && t2(1,j+1)<=N-1)
A2(1,j+1)= 2 - ((2.*j)./(N-1));
else
A2(1,j+1)=0;
end
end
[H2,w2]=freqz(A2,1,1024);
wN2=w2./(2*pi);
HN2=abs(H2)./max(abs(H2));
plot(wN2,20.*log10(HN2),'r')
hold off;
legend('M=25', 'N=49');
 EXP6
AIM:
To design FIR Low Pass Filter using Rectangular and Triangular windows with user input
specifications for pass band edge frequency, transition width, stop band attenuation and
sampling frequency.

CODE:
a)Rectangular
%RECTANGULAR
%101806024
close all;
clear all;
fp=1500;
tw=500;
fsample=8000;
tw=tw/fsample;
fc=fp+tw/2;
N=floor(0.9/tw);
h=ones(1,N);
hd=ones(1,N);

for i=1:N
a(i)=(-(N-1)/2-1)+i;
end

rectangular=ones(1,N);
for i=1:N
wc=2*pi*fc;
h(i)=sin(a(i)*wc)/(a(i)*wc);
hd(i)=h(i)*rectangular(i);
end

subplot(4,2,1);
stem(a,hd);
title('LPF with Rectangular Window');
xlabel('t');
ylabel('hd');

[X2,w2]=freqz(hd,1,1024);
X2max=max(abs(X2));
subplot(4,2,2);
plot(w2/pi,20*log10(abs(X2)/X2max));
title('LPF with Rectangular Window Freq Response');
xlabel('f');
ylabel('hd');
b)Triangular
%TRIANGULAR
%101806024
close all;
clear all;
fp=1500;
tw=500;
fsample=8000;
tw=tw/fsample;
fc=fp+tw/2;
N=floor(0.9/tw);
a=[zeros(1,N)];

for i=1:N
a(i)=(-(N-1)/2-1)+i;
end
a(i)=a(i)+0.00001;
n=-(N-1)/2:(N-1)/2
triangular=(1-abs(n)/(N-1)/2);
for i=1:N
wc=2*pi*fc;
h(i)=2*fc*sin(a(i)*wc)/(a(i)*wc)
hd(i)=h(i)*triangular(i);
end

subplot(5,2,1);
stem(a,hd);
title('LPF with trinagulars Window');
xlabel('t');
ylabel('hd');

[X2,w2]=freqz(hd,1,1024);
X2max=max(abs(X2));
subplot(4,2,2);
plot(w2/2*pi,20*log10(abs(X2)/X2max));
title('LPF with triangular Window Freq Response');
xlabel('f');
ylabel('hd');
 EXP7
AIM:
To design FIR Low Pass Filter using Hamming and hanning windows with user input specifications
for pass band edge frequency, transition width, stop band attenuation and sampling frequency

CODE:

%RishabdevDhawan

clc;

clear all;

rp=input('enter the passband ripple:');%0.02

rs=input('enter the stopband attenuation:');%0.01

fp=input('enter the passband freq:'); %1500

fs=input('enter the stopband freq:');%2000

f=input('enter the sampling freq:');%6000

wp=2*fp/f;

ws=2*fs/f;

num=-20*log10(sqrt(rp*rs))-13;

dem=14.6*(fs-fp)/f;

n=ceil(num/dem);

n1=n+1;

if (rem(n,2)~=0)

n1=n;
n=n-1;

end

y1=hamming(n1);

y2=hann(n1);

% high-pass filter

%hamming

b=fir1(n,wp,'high',y1);

[h,o]=freqz(b,1,256);

m=20*log10(abs(h));

figure;plot(o/pi,m);

title('FIR Filter Response using hamming window ');

ylabel('Gain in dB');

xlabel('Normalised frequency');

%hanning

b=fir1(n,wp,'high',y2);

[h,o]=freqz(b,1,256);

m=20*log10(abs(h));

figure;plot(o/pi,m);

title('FIR Filter Response using Hanning window ');

ylabel('Gain in dB');

xlabel('Normalised frequency');
 EXP 8

AIM: Write a MATLAB program to design FIR BANDPASS filter using Kaiser windows
with the user input specifications for pass band range, transition width, passband ripple,
stop band attenuation and sampling frequency.

CODE:
clc
clear all
fp1=input('Enter passband edge frequency 1: ');
fp2=input('Enter passband edge frequency 2: ');
df1=input('Enter transition width: ');
fs=input('Enter sampling frequency: ');
att=input('Enter Stopband attenuation: ');
rip=input('Enter Passband ripple: ');
L=input('Enter L: ');
fp=fp1/fs;
fpp=fp2/fs;
df=df1/fs;
dp=(10.^(rip/20))-1;
ds=(10.^(-att/20));
d=min(dp,ds);
A=-20*log10(d);
if A<=21
beta=0;
elseif A>=21 && A<=50
beta=0.584*((A-21)^0.4)+0.07886*(A-21);
else
beta=0.1102*(A-8.7);
end
X=(A-7.95)/(14.36*df);
N=ceil(X);
if mod(N,2) == 0 N=N+1;
else N=N;
end
for n=-(N-1)/2:(N-1)/2
x(n+((N-1)/2)+1)=beta*((1-(2*n/(N-1))^2))^0.5;
x1(n+((N-1)/2)+1)=beta;
for k=1:L

y(k)=((((x(n+((N-1)/2)+1)/2).^k)./factorial(k)).^2);
end
Y(n+((N-1)/2)+1)=sum(y);
for k=1:L
y1(k)=((((x1(n+((N-1)/2)+1)/2).^k)./factorial(k)).^2);
end
Y1(n+((N-1)/2)+1)=sum(y1);
I(n+((N-1)/2)+1)=1+Y(n+((N-1)/2)+1);
I1(n+((N-1)/2)+1)=1+Y1(n+((N-1)/2)+1);
end
w=I./I1;
f1=fp+df/2;
f2=fpp+df/2;
h1=2*(f2-f1);
for n=1:(N-1)/2
h2(n)=(((2*f2)*sin(2*n*pi*f2))/(2*n*pi*f2))-
(((2*f1)*sin(2*n*pi*f1))/(2*n*pi*f1));
end
h3=fliplr(h2);
hd=[h3 h1 h2];
hh=hd.*w;
subplot(2,1,1);
[x2]=fft(w,1000);
a2=(20*log10(abs(x2)/max(x2))); y1=(0:1000-1).*(2*pi/fs);
plot(y1,a2)
title('Kaiser Window Response')
xlabel('Normalized Frequency')
ylabel('Magnitude in dB')
subplot(2,1,2);
[x3]=fft(hh,1000);
a3=(20*log10(abs(x3)/max(x3)));
plot(y1,a3)
title('Filter Response')
xlabel('Normalized Freuqency')
ylabel('Magnitude in dB')

Ente r pa ssband edge f requency I :

150

Ente r pa ssband edge I requency 2 :

Z50

Ente r t rans1t1on w1dth :

100

Ente r sanpl1ng f requen Cy:

 1200

Ente r Stopband attenuation :

20

Ence r Pa ssband ripple: D . a1

Ente r L :

25

OUTPUT:
 EXP 9
AIM: Write a MATLAB program to design digital LOWPASS filter using Butterworth
prototype with the user input specifications for pass band edge frequency, stopband
edge frequency, passband ripple, stop band attenuation. Use impulse invariant
method.

CODE:
function [b,a] = afd_butt (Wp,Ws,Rp,As) ;
% X Analog Lowpass Filter Design: Butterworth
% y
% [b,a] = afd-butt(Wp,Ws,Rp.As);
% % b = Numerator coefficients of Hab)
% X a = Denominator coefficients of Ha(d
% % Wp = Passband. edge frequency in rad/sec; Wp > 0
% X Ws = Stopband edge frequency in rad/sec; Ws > Wp > 0
% Rp = Passband ripple in +dB; (Rp > 0)
% As = Stopband attenuation in +dB; (As > 0)
% X
if Wp<=0
error('Passband edge must be larger than 0')
end
if Ws <= Wp
error('Stopband edge must be larger than Passband edge')
end
if (Rp <= 0) | (As < 0)
error('PB ripple and/or SB attenuation ust be larger than 0')
end
N =ceil((log10((10^(Rp/10)-1)/(10^(As/10)-1)))/(2*log10(Wp/Ws)));
disp('N')
disp(N)
OmegaC = Wp/((10^(Rp/10)-1)^(1/(2*N))) ;
[b,a]=U_buttap(N,OmegaC) ;
function [C,B,A] = dir2par(b,a);
% X Dim-form to PWLEL-form conversion
% y. S
% % CC,B,Al = dir2par(b,a)
% X C = Polynomial part when length(b1 >= length(a1
% X B = K by 2 matrix of real coefficients containing bk's;
% X A = K by 3 matrix of real coefficients containing ak's;
% X b = numerator polynomial coefficients of DIRECT form
% X a = denominator polynomial coefficients of DIRECT form
% X
M = length(b) ; N = length(a); [rl ,p1 ,C] = residuez(b,a) ;
p = cplxpair (p1,10000000*eps) ; I = cplxcomp(p1,p) ; r = rl(I);
K = floor(N/2); B = zeros(K,2); A = zeros(K,3);
if K*2 == N; %W even, order of A(z) odd, one factor is first
order for i=1:2:N-2
Brow = r(i:1:i+1,:);
Arow = p(i:1:i+1,:);
[Brow,Arow] = residuez(Brow,Arow,[]); B(fix((i+1)/2),:) = real(Brow) ;
A(fix((i+1)/2), :) = real(Arow); end [Brow,Arow] = residuez(r(N-1)
,p(N-1), []) ;
B(K,:) = [real(Brow) 0]; A(K,:) = [real(Arow) 0]; else
for i=1:2:N-1
Brow = r(i:1:i+1,:);
Arow = p(i:1:i+1,:);
[Brow Arow] = residuez(Brow, Arow, [] ) ; B(fix((i+1)/2), :) = real(Brow)
;
A(fix((i+1)/2), :) = real(Arow); end end
function [b,a] = imp_invr(c,d,T)
% X Impulse Invariance Transformation from Analog to Digital Filter
%X
% X [b,d = imp-invr(c,d,T)
% X b = Numerator polynomial in f(-1) of the digital filter
% X a = Denominator polynomial in 2-(-1) of the digital filter
% X c = Numerator polynomial in s of the analog filter
% X d = Denominator polynomial in s of the analog filter
% X T = Sampling (transformation) parameter
% X
[R,p,k] = residue(c,d);
p = exp(p*T);
[b a] = residuez(R,p,k) ;
b = real(b');
a = real(a');
function I = cplxcomp(p1,p2)
% I = cplxcomp(pl,p2)
% X Compares two complex pairs which contain the same scalar elements
% but (possibly) at diffenent indices. This routine should be
% used after CPLXPAIR routine for rearranging pole vector and its
% corresponding residue vector.
% p2 = cplxpair(p1) I=[] ;
for j=1: 1:length(p2) for i=1: 1: length(p1) if (abs(p1(i)-p2(j)) <
0.0001) I=[I,i];
end
end
end
I=I' ;
function [b,a] = U_buttap(N,Omegac) ;
% X Unnormalized Butterworth Analog Lowpass Filter Prototype
% X [b,al = U-buttap(N,Gmegac) ;
% % b = numerator polynomial coefficients of Hab)
% % a = denominator polynomial coefficients of Hab)
% % N = Order of the Butternorth Filter
% % hegac = Cutoff frequency in radians/sec
% X
[z p k]=buttap(N); p=p*Omegac; k=k*Omegac^N; B=real(poly(z)); b0=k;
b=k*B; a=real(poly(p));
% MAIN PROGRAM
clc;
clear all;
close all;
wp = 0.2*pi;
%digital Passband freq in rad ws = 0.3*pi;
%digital Stopband freq in rad Rp = 1;
%Passband ripple in dB
As = 15;
%Stopband attenuation in dB
% Analog Prototype Specifications:
T = 1;
%Set T=1 OmegaP = wp/T;
% Prototype Passband freq OmegaS = ws/T;
% Prototype Stopband freq
% Analog Prototype Order Calculation:
[cs,ds]=afd_butt(OmegaP,OmegaS,Rp, As);
[b,a] = imp_invr(cs,ds,T);
[C B A]=dir2par(b,a) [X2,w2] = freqz(b,a,1024);
X2max = max(abs(X2));
plot(w2/(pi),(abs(X2)),'r');
%Plotting the graph
%giving xlabel and ylabel and title
xlabel('-------- \bf\omega (Normalized w.r.t \pi)>');
ylabel('-------- \bf|H(w)| (Normalized w.r.t.
Max(h(w)))'); title('Magnitude Response'); grid
figure plot(w2/(pi),angle(X2)/(pi),'g');
xlabel('-------- \bf\omega (Normalized w.r.t \pi)>');
ylabel('-------- \bfphase (Normalized w.r.t. \pi)>');
title('Phase Response');

OUTPUT:
 EXP 10

AIM: Write a MATLAB program to generate following sequence

i) sinusoidal sequence of user input normalized frequency
ii) sum of two sinusoids with multiple normalized frequency.

Perform up sampling for both by factor 4 and down

sampling by factor 5 on sequence of length 51.

CODE:
(i)
N=51;
n=0:N-1;
N_Freq1=input('Enter the value of normalized frequency1: ');
x1=sin(2*pi*N_Freq1*n);
N_Freq2=input('Enter the value of normalized frequency2: ');
N_Freq3=input('Enter the value of normalized frequency3: ');
x2=sin(2*pi*N_Freq2*n)+sin(2*pi*N_Freq3*n);
M=input('Value of down sampling factor: ');
y1=downsample(x1,M);
L1=length(y1);
y2=downsample(x2,M);
L2=length(y2);
subplot(2,2,1);
stem(0:N-1,x1(1:N));
title('x1(n)');
subplot(2,2,2);
stem(0:N-1,x2(1:N));
title('x2(n)');
subplot(2,2,3);
stem(0:L1-1,y1(1:L1));
title('downsampled x1(n)');
subplot(2,2,4);
stem(0:L2-1,y2(1:L2));
title('downsampled x2(n)');
OUTPUT:

CODE:

(ii)
N=51;
n=0:N-1;
N_Freq1=input('Enter the value of normalized frequency1: ');
x1=sin(2*pi*N_Freq1*n);
N_Freq2=input('Enter the value of normalized frequency2: ');
N_Freq3=input('Enter the value of normalized frequency3: ');
x2=sin(2*pi*N_Freq2*n)+sin(2*pi*N_Freq3*n);
M=input('Value of up sampling factor: ');
y1=upsample(x1,M);
L1=length(y1);
y2=upsample(x2,M);
L2=length(y2);
subplot(2,2,1);
stem(0:N-1,x1(1:N));
title('x1(n)');
subplot(2,2,2);
stem(0:N-1,x2(1:N));
title('x2(n)');
subplot(2,2,3);
stem(0:L1-1,y1(1:L1));

title('upsampled x1(n)');
subplot(2,2,4);
stem(0:L2-1,y2(1:L2));
title('upsampled x2(n)');

OUTPUT: