DSP Lab Manual

Digital Signal Processing Lab Manual
Program No. 1
1. Verification of Sampling Theorem both in time and frequency domains.
% Verification of sampling theorem
tfinal=0.05;
t=0:0.00005:tfinal;
fd=input('enter the desired analog frequency');
xt=cos(2*pi*fd*t);
%Simulate condition for under sampling i.e.,fs1,2*fd
fs1=1.3*fd;
n1=0:1/fs1 :tfinal;
xn=cos(2*pi*n1*fd);
subplot(3,1,1);
plot(t,xt,'b',n1,xn,'r*-');
title('Undersampling Plot');
%Condition For Nyquist Sampling fs2=2*fd
fs2=2*fd;
n2=0:1/fs2 :tfinal;
xn=cos(2*pi*n2*fd);
subplot(3,1,2);
title('Nyquist Plot');
GCE, Ramanagaram Dept. of E&E

1
Digital Signal Processing Lab ManualDigital Signal Processing Lab Manual
%Condition For OverSampling fs2=2*fd
fs3=5*fd;
n3=0:1/fs3 :tfinal;
xn=cos(2*pi*n3*fd);
subplot(3,1,3);
title('Oversampling Plot');
xlabel('time');
ylabel('amplitude');
legend('analog','discrete');
Undersampling Plot
1
-1
0 0.005 0.01 0.015 0.02 0.025 0.03 0.035 0.04 0.045 0.05
Nyquist Plot
1
-1
0 0.005 0.01 0.015 0.02 0.025 0.03 0.035 0.04 0.045 0.05
Oversampling Plot
1
amplitude
analog
0 discrete
-1
0 0.005 0.01 0.015 0.02 0.025 0.03 0.035 0.04 0.045 0.05
time

2
Program No. 2
Evaluation of impulse response of a system
Aim: Write a program to generate an impulse response of a system using MATLAB
clc;
clear all;
close all;
% Difference equation of a second order system
% y(n) = x(n)+0.5x(n-1)+0.85x(n-2)+y(n-1)+y(n-2)
b=input('enter the coefficients of x(n),x(n-1)-----');
a=input('enter the coefficients of y(n),y(n-1)----');
N=input('enter the number of samples of imp response ');
[h,n]=impz(b,a,N);
stem(n,h);
title('plot of impulse response');
ylabel('amplitude');
xlabel('time index----->N');
disp(h);
grid on;
Output
enter the coefficients of x(n),x(n-1)-----[1 -2 -3 2]
enter the coefficients of y(n),y(n-1)----[2,-1,2,2]
enter the number of samples of imp response 10
h(n)={0.5000,-0.7500,-2.3750,0.0625,3.1563,3.8906,-1.2734,-7.6836,-
6.4590,5.7275}

3
plot of impulse response

6
0
amplitude
-2
-4
-6
-8
0 1 2 3 4 5 6 7 8 9
time index----->N

4
Program No. 3
Linear convolution of given sequences.
Aim: Write a program to perform linear convolution of given sequences using

MATLAB
clc;
x1=input('enter the first sequence ');
subplot(3,1,1);
stem(x1);
xlabel('n');
ylabel('x1');
title('plot of the first sequence x1');
x2=input('enter 2nd sequence ');
subplot(3,1,2);
stem(x2);
xlabel('n');
ylabel('x2');
title('plot of 2nd sequence x2');
f=conv(x1,x2);
disp('output of linear convolution of two sequences is y = ');
disp(f);
xlabel('n');
ylabel('f');
subplot(3,1,3);
stem(f);
title('linear conv of two sequences x1 & x2 ');

5
OUTPUT:
enter the first sequence [1,2,-1,-2]
enter 2nd sequence [1,-2,-3,2,4]
output of linear convolution of two sequences is
y = 1 0 -8 -4 15 12 -8 -8

6
Program No. 4. a).

Circular convolution of given sequences using convolution
summation formula.
Aim: Write a program to perform Circular convolution of given sequences using
convolution summation formula.
% circular Convolution using summation formula
xn=input('Enter The Input Sequence');
subplot(3,1,1);
stem(xn);
xlabel('n');
ylabel('x1');
title('plot of the first sequence xn');
hn=input('Enter The Impulse Response');
subplot(3,1,2);
stem(hn);
xlabel('n');
ylabel('hn');
title('plot of 2nd sequence hn');
N= max(length(xn),length(hn));
for n=0:N-1
y(n+1)=0;
for k=0:N-1
i=mod((n-k),N);
if i<0
i=i+N;
end
y(n+1)=y(n+1)+hn(k+1)*xn(i+1);
end
end
disp('Circular Convolution of x(n) & h(n) is y(n)=');
disp(y);
n1=0:N-1;
xlabel('n');
ylabel('y');
subplot(3,1,3);
stem(y);
title('Circular Convolution Output y(n)');

7
Output:
Enter The Input Sequence[1,2,3,4]
Enter The Impulse Response[4,3,2,1]
Circular Convolution of x(n) & h(n) is y(n)=
24 22 24 30

8
Program No. 4. b).

Circular Convolution Using Matrix Method:
Aim: Write a program to perform Circular convolution of given sequences using
Matrix Method:
clc;
close all;
x1=input('Enter The First sequence x(n)');
h1=input('Enter The Second sequence h(n)');
N1=length(x1);
N2=length(h1);
N=max(N1,N2);
if(N1>N2)
h1=[h1,zeros(1,N1-N2)];
x1=[x1,zeros(1,N2-N1)];
end;
x=transpose(x1);
h=transpose(h1);
temp=h;
for i=1:N-1;
temp=circshift(temp,1);
h=horzcat(h,temp);
end;
h
x
y=h*x
disp('Circular convolved output y[n]=');y
subplot(3,1,1);
stem(x1);
xlabel('N-->');

9
ylabel('Amplitude-->');
title('Input Sequence x[n]');
subplot(3,1,2);
stem(h1);
xlabel('N-->');
title('Second Sequence h[n]');
subplot(3,1,3);
stem(y);
xlabel('N-->');
title('Circular convolved output y[n]');
Output:
Enter The First sequence x(n)[1 2 3 4 ]
Enter The Second sequence h(n)[4 3 2 1 ]
h=
4 1 2 3
3 4 1 2
2 3 4 1
1 2 3 4
x=
1
2
3
4
Circular convolved output y[n]=[24 22 24 30]

10

11
Program No. 4. C).

Linear convolution using circular convolution with zero padding.
Aim: Write a program to perform linear convolution using circular convolution with
zero padding.
% Linear convolution using circular Convolution with zero padding

xn=input('Enter The Input Sequence');
subplot(3,1,1);
stem(xn);
xlabel('n');
ylabel('x1');
title('plot of the first sequence xn');
hn=input('Enter The Impulse Response');
subplot(3,1,2);
stem(hn);
xlabel('n');
ylabel('hn');
title('plot of 2nd sequence hn');
N= length(xn)+length(hn)-1;
xn=[xn zeros(1,N-length(xn))]
hn=[hn zeros(1,N-length(hn))]
for n=0:N-1
y(n+1)=0;
for k=0:N-1
i=mod((n-k),N);
if i<0
i=i+N;
end
y(n+1)=y(n+1)+hn(k+1)*xn(i+1);
end

12
end
disp('Circular Convolution of x(n) & h(n) is y(n)=');
disp(y);
n1=0:N-1;
xlabel('n1');
ylabel('y');
subplot(3,1,3);
stem(n1,y);
title('Circular Convolution Output y(n)');
output:
Enter The Input Sequence[1,2]
Enter The Impulse Response[1,3,5,7,2]
xn = 1 2 0 0 0 0
hn = 1 3 5 7 2 0
Circular Convolution of x(n) & h(n) is y(n)=
1 5 11 17 16 4

13
Program No. 5.
Computation of N – point DFT and to plot the magnitude and phase
spectrum.
Aim: Write a program to compute N – point DFT and to plot the magnitude and phase
spectrum.
N = input('Enter the the value of N ');

x = input('Enter the input sequence for which DFT is to be calculated ');
n=[0:1:N-1];
k=[0:1:N-1];
WN=exp(-1j*2*pi/N);
nk=n'*k;
WNnk=WN.^nk;
Xk=x*WNnk;
MagX=abs(Xk) % Magnitude of calculated DFT
PhaseX=angle(Xk)*180/pi % Phase of the calculated DFT
figure(1);
subplot(2,1,1);
stem(k,MagX);
xlabel('k');
ylabel('x(k)');
title('magnitude plot');
subplot(2,1,2);
stem(k,PhaseX);
xlabel('k');
ylabel('Q(k)');
title('Phase plot');

14
output;
Enter the the value of N 8
Enter the input sequence for which DFT is to be calculated [1,2,3,4,-1,-2,-3,-4]
MagX = Φ
Columns 1 through 5
0 14.5090 0.0000 5.4305 0.0000
Columns 6 through 8
5.4305 0.0000 14.5090
PhaseX =
Columns 1 through 5
0 -93.2732 45.0000 -27.2357 90.0000
Columns 6 through 8
27.2357 135.0000 93.2732

15
magnitude plot
15
10
x(n)
0
0 1 2 3 4 5 6 7
n
Phase plot
200
100
Q(n)
-100
0 1 2 3 4 5 6 7
n

16
Program No. 6.
LINEAR CONVOLUTION USING DFT AND IDFT
Aim: Write a program to perform linear convolution using DFT and IDFT.
clc;
clear all;
close all;
x=input('Enter x[n]:');
h=input('Enter h[n]:');
nx=length(x);
nh=length(h);
n=nx+nh-1;
xnew=[x zeros(1,n-nx)];
hnew=[h zeros(1,n-nh)];
xf=fft(xnew);
hf=fft(hnew);
zf=xf.*hf;
z=ifft(zf);
lx=0:1:nx-1;
lh=0:1:nh-1;
nz=0:1:length(z)-1;
subplot(3,1,1);
stem(lx,x);
xlabel('Time');ylabel('Amplitude');
title('Input sequence x[n]');
grid;
subplot(3,1,2);
stem(lh,h);

17
title(‘Impulse Response h[n]’);

grid;
subplot(3,1,3);
stem(nz,z);
title('Linear Convolution ');
grid;
%Verification
z1=conv(xnew,hnew)
INPUT:
Enter x[n]:[1 2 3 4]
Enter h[n]:[4 3 2 1]
OUTPUT:
z1 = 4 11 20 30 20 11 4 0 0 0 0 0 0

18

19
Program No. 7. a).

Aim: Write a program to find Impulse response for a given difference equation.
% To find Impulse response
N = input('Length Of Response required=');
b = [1];%x(n) coefficients
a = [1,-1,0.9]; %y(n) coefficients
%impulse input
x = [1,zeros(1,N-1)];
% time vector for plotting
n = 0:N-1;
% Impulse Response
h = filter(b,a,x);
subplot(2,1,1);
stem(n,x);
title('impulse input');
xlabel('n');
ylabel('?(n)');
subplot(2,1,2);
stem(n,h);
title('impulse response');
xlabel('n');
ylabel('h(n)');

20

21
Program No. 7. b).

Aim: Write a program to find a Step response for a given difference equation.
% To find Step response

%y[n]-0.8y [n-1]=x[n]
N = input('Length Of response required=');
b = [1];%x(n) coefficients
a = [1,-1,1]; %y(n) coefficients
x = [ones(1,N)]; %Step input
% time vector for plotting
n = 0:1:N-1;
% Step Response
y = filter(b,a,x);
subplot(2,1,1);
stem(n,x);
title('Step input');
xlabel('n');
ylabel('u(n)');
subplot(2,1,2);
stem(n,y);
title('Step response');
xlabel('n');
ylabel('y(n)');

22

23
Program No. 8.
Calculation of DFT and IDFT by FFT
Aim: Write a program to compute DFT and IDFT by FFT.
n = 0:6;
x = input('enter the i/p sequence ');
a = fft(x);
mag = abs(a);
pha = angle(a);
subplot(3,1,1);
stem(mag);
xlabel('k')
ylabel('|x(k)|');
grid on
title('Magnitude Response');
subplot(3,1,2);
stem(pha);
xlabel('k')
ylabel('?(k)');
grid on
title('phase Response');
xn= ifft(a);
subplot(3,1,3);
stem(xn);
xlabel('n')
ylabel('x(n)');
title('Input Sequence');

24
output:
enter the i/p sequence [1,2,3,1,-4,-5,-1,-2]
Magnitude Response
20
|x(k)|
10
0
1 2 3 4 5 6 7 8
k
phase Response
5
?(k)
-5
1 2 3 4 5 6 7 8
k
Input Sequence
5
x(n)
-5
1 2 3 4 5 6 7 8
n

25
Program No. 9. a).

Design of IIR Filter
a) Butterworth Filter
Aim: Write a program to design digital Butterworth low pass filter for a given
specification
To design digital Butterworth low pass filter clear all;
clc;
rp = input(‘Enter the pass band ripple freq’);
rs = input(‘Enter the stop ripple freq’);
fp = input(‘Enter the pass band freq’);
fs = input(‘Enter the stop band freq’);
f = input(‘Enter the sample freq’);
w1=2*fp/f;
w2=2*fs/f;
[n,wn]= buttord(w1,w2,rp,rs);
[b,a] = butter(n,wn);
w=0:.1:pi;
[h,p] = freqz(b,a,w);
g=20*log10(abs(h));
A=angle(h);
subplot (2,2,1); plot(p/pi,g);
ylabel(‘amp’);
xlabel(‘ferq’);
title(‘amp,freq’);
subplot (2,2,2); plot(p/pi,A);
xlabel(‘normal. freq’);
ylabel(‘phase’);
title(‘normal.freq,phase’);

26
Output:
Enter the pass band ripple freq= .5
Enter the stop band ripple freq= 40
Enter the pass band freq= 2000
Enter the stop band freq= 3000
sample freq= 10000

27
Program No. 9. b).

Aim: Write a program to design digital Butterworth high pass filter for a given
specification
%To design digital Butterworth high pass filter

clear all;
clc;
rp = input(' Enter the pass ripple freq');
rs = input(' Enter the stop ripple freq');
fp = input(' Enter the pass band freq');
fs = input(' Enter the stop band freq');
f = input(' Enter the sample freq');
w1= 2*fp/f;
w2= 2*fs/f;
[b,a] = butter(n,wn,'high');
w=0:.1:pi;
[h,p]= freqz(b,a,w);
g=20*log10(abs(h));
A=angle(h);
subplot (2,2,1);
plot(p/pi,g);
ylabel('amp');
xlabel('ferq');
title('amp,freq');
subplot (2,2,2);
plot(p/pi,A);
xlabel('normal. freq');
ylabel('phase');
title('normal.freq,phase');
Output
Enter the pass ripple freq .5
Enter the stop ripple freq 40
Enter the pass band freq 2000
Enter the stop band freq 3000
Enter the sample freq 10000

28
amp,freq
0
-100
amp
-200
-300
-400
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
ferq
normal.freq,phase
4
2
phase
-2
-4
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
normal. freq

29
Program No. 9. c).

Aim: Write a program to design digital Butterworth band pass filter for a given
specification
%To design digital Butterworth Band pass filter

clear all;
clc;
rp = input('Enter the pass ripple freq ');
rs = input('Enter the stop ripple freq ');
fp = input('Enter the pass band freq ');
fs = input('Enter the stop band freq ');
f = input('Enter the sample freq ');
w1= 2*fp/f;
w2= 2*fs/f;
[b,a] = butter(n,wn,'high');
w=0:.1:pi;
g=20*log10(abs(h));
A=angle(h);
subplot (2,1,1);
plot(p/pi,g);
ylabel('amp');
xlabel('ferq');
title('amp,freq');
subplot (2,1,2);
plot(p/pi,A);
ylabel('phase');

30
Output:
Enter the pass ripple freq= .5

Enter the stop ripple freq= 40
Enter the pass band freq= 2000
Enter the stop band freq= 3000
Enter the sample freq= 10000

31
Program No. 9. d).

Aim: Write a program to design digital Butterworth band stop filter for a given
specification
%To design digital Butterworth Band stop filter using MATLAB.

clear all;
clc;
rp = input('Enter the pass ripple freq ');
rs = input('Enter the stop ripple freq ');
fp = input('Enter the pass band freq ');
fs = input('Enter the stop band freq ');
f = input('Enter the sample freq ');
w1= 2*fp/f;
w2= 2*fs/f;
[n]= buttord(w1,w2,rp,rs);
wn = [w1,w2];
[b,a] = butter(n,wn,'stop');
w=0:.1:pi;
g=20*log10(abs(h));
A=angle(h);
subplot(2,1,1);
plot(p/pi,g);
ylabel('amp');
xlabel('ferq');
title('amp,freq');
subplot (2,1,2);
plot(p/pi,A);
ylabel('phase');

32
Output
Enter the pass ripple freq .3
Enter the stop ripple freq 30
Enter the pass band freq 2000
Enter the stop band freq 3000
Enter the sample freq 10000

33
Program No. 10.a).

Aim: Write a program to design low pass FIR FILTER for given specification.
clc;
clear all;
wc=input('enter the value of cut off frequency');
N=input('enter the order of filter');
alpha=(N-1)/2;
eps=0.001;
%Rectangular Window
n=0:1:N-1;
hd=sin(wc*(n-alpha+eps))./(pi*(n-alpha+eps));
hn=hd
w=0:0.01:pi;
h=freqz(hn,1,w);
plot(w/pi,abs(h));
hold on
%Hamming Window
n=0:1:N-1;
wh=0.54-0.46*cos((2*pi*n)/(N-1));
hn=hd.*wh
w=0:0.01:pi;
h=freqz(hn,1,w);
plot(w/pi,abs(h),'ms');
hold off;
hold on
%Hanning Window
n=0:1:N-1;
wh=0.5-0.5*cos((2*pi*n)/(N-1));
hn=hd.*wh
w=0:0.01:pi;
h=freqz(hn,1,w);
plot(w/pi,abs(h),'blue');
hold off;
hold on
%Blackman Window
n=0:1:N-1;

34
wh=0.42-0.5*cos((2*pi*n)/(N-1))+0.08*cos((4*pi*n)/(N-1));
hn=hd.*wh
w=0:0.01:pi;
h=freqz(hn,1,w);
plot(w/pi,abs(h),'green');
hold off;
Output:
enter the value of cut off frequency2
enter the order of filter10
hn =
Columns 1 through 6
0.0293 0.0596 -0.1222 0.0304 0.5361 0.5353
Columns 7 through 10
0.0295 -0.1220 0.0599 0.0290
hn =
Columns 1 through 6
0.0023 0.0112 -0.0562 0.0234 0.5212 0.5205
0.0227 -0.0561 0.0112 0.0023
hn =
Columns 1 through 5
0 0.0070 -0.0505 0.0228 0.5199
0.5192 0.0221 -0.0504 0.0070 0
hn =
Columns 1 through 5
-0.0000 0.0030 -0.0315 0.0191 0.5099
0.5092 0.0186 -0.0315 0.0030 -0.0000
1.4
1.2
0.8
0.6
0.4
0.2
0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1

35
Program No. 10.b).

Aim: Write a program to design high pass FIR FILTER for given specification using
window functions.
% DESIGN FOR HIGH PASS FILTER:

clc;
clear all;
wc=input('enter the value of cut off frequency');
N=input('enter the value of filter');
alpha=(N-1)/2;
eps=0.001;
%Rectangular Window
n=0:1:N-1;
hd=(sin(pi*(n-alpha+eps))-sin((n-alpha+eps)*wc))./(pi*(n-alpha+eps));
hn=hd
w=0:0.01:pi;
h=freqz(hn,1,w);
plot(w/pi,abs(h));
hold on
%Hamming Window
n=0:1:N-1;
wh=0.54-0.46*cos((2*pi*n)/(N-1));
hn=hd.*wh
w=0:0.01:pi;
h=freqz(hn,1,w);
plot(w/pi,abs(h),'ms');
hold off;
hold on
%Hanning Window
n=0:1:N-1;
wh=0.5-0.5*cos((2*pi*n)/(N-1));
hn=hd.*wh
w=0:0.01:pi;
h=freqz(hn,1,w);
plot(w/pi,abs(h),'blue');
hold off;
hold on

36
%Blackman Window
n=0:1:N-1;
wh=0.42-0.5*cos((2*pi*n)/(N-1))-0.08*cos((4*pi*n)/(N-1));
hn=hd.*wh
w=0:0.01:pi;
h=freqz(hn,1,w);
plot(w/pi,abs(h),'green');
hold off;
OUTPUT:
enter the value of cut off frequency2
enter the value of filter15
hn =
Columns 1 through 6
-0.0449 0.0284 0.0347 -0.0790 0.0302 0.1198
-0.2890 0.3634 -0.2899 0.1211 0.0291 -0.0784
0.0345 0.0285 -0.0452
hn =
Columns 1 through 6
-0.0036 0.0036 0.0088 -0.0346 0.0194 0.0991
-0.2758 0.3634 -0.2767 0.1001 0.0187 -0.0343
0.0087 0.0036 -0.0036
hn =
Columns 1 through 5
0 0.0014 0.0065 -0.0307 0.0185
0.0972 -0.2747 0.3634 -0.2755 0.0983
0.0178 -0.0305 0.0065 0.0014 0

37
hn =
Columns 1 through 5
0.0072 -0.0023 0.0044 -0.0301 0.0182
0.0898 -0.2371 0.3052 -0.2379 0.0908
0.0176 -0.0299 0.0044 -0.0023 0.0072
1.4
1.2
0.8
0.6
0.4
0.2
0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1

38
Program No. 10.c).

Aim: Write a program to design band pass FIR FILTER for given specification using
window functions.
% DESIGN FOR BAND PASS FILTER:

clc;
Wc1=input('enter the value of Wc1=');
Wc2=input('enter the value of Wc2=');
N=input('enter the value of N=');
alpha=(N-1)/2;
eps=0.001;
%Rectangular Window
n=0:1:N-1;
hd=(sin(Wc1*(n-alpha+eps))-sin(Wc2*(n-alpha+eps)*pi))./((n-alpha+eps)*pi);
hn=hd
W=0:0.01:pi;
h=freqz(hn,1,W);
plot(W/pi,abs(h));
hold on;
%Hamming Window
n=0:1:N-1;
Wn=0.54-0.46*cos((2*pi*n)/(N-1));
hn=hd.*Wn
W=0:0.01:pi;
h=freqz(hn,1,W);
plot(W/pi,abs(h),'green');
hold on;
%Hanning Window
n=0:1:N-1;
Wn=0.5-0.5*cos((2*pi*n)/(N-1));

39
hn=hd.*Wn
W=0:0.01:pi;
h=freqz(hn,1,W);
plot(W/pi,abs(h),'red');
hold off;
%Blackman Window
n=0:1:N-1;
wh=042-0.5*cos((2*pi*n)/(N-1))-0.08*cos((4*pi*n)/(N-1));
hn=hd.*wh
w=0:0.01:pi;
h=freqz(hn,1,w);
plot(W/pi,abs(h),'green');
hold off;
Output:
enter the value of Wc1=1
enter the value of Wc2=1.5
enter the value of N=12
hn =
Columns 1 through 6
-0.0816 -0.1194 0.0322 0.1668 0.0624 -0.1479
-0.1420 0.0609 0.1656 0.0326 -0.1189 -0.0819
hn =
Columns 1 through 6
-0.0065 -0.0183 0.0112 0.1010 0.0525 -0.1452
-0.1393 0.0512 0.1003 0.0114 -0.0182 -0.0066
hn =
Columns 1 through 6
0 -0.0095 0.0094 0.0953 0.0516 -0.1449
-0.1391 0.0504 0.0946 0.0095 -0.0094 0

40
hn =
Columns 1 through 5
-3.3807 -4.9610 1.3473 7.0313 2.6401
-6.2735 -6.0216 2.5781 6.9813 1.3647
-4.9405 -3.3920
40
35
30
25
20
15
10
0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1

41
Program No. 11.a).

Aim: Write a program to design low pass FIR filter using frequency sampling
method.
%Design a lowpass filter using frequency sampling method

M=63; Wp=0.5*pi;%the number of samples and the passband cutoff frequency
m=0:(M+1)/2; Wm=2*pi*m./(M+1);%the sampling points and the stopband cutoff
frequency
mtr=floor(Wp*(M+1)/(2*pi))+2;%round to negative part,i.e.floor(3.5)=3;floor(-
3.2)=-4
Ad=[Wm<=Wp];Ad(mtr)=0.38;
Hd=Ad.*exp(-j*0.5*M*Wm);%define frequency-domain sampling vector H(k)
Hd=[Hd conj(fliplr(Hd(2:(M+1)/2)))];
%fliplr is to realize the fliplr of matrix and conj
h=real(ifft(Hd));% h(n)=IDFT[H(k)
w=linspace(0,pi,1000);%get 1000 row vectors between 0 and pi
H=freqz(h,[1],w);%the amplitude -frequency characteristic diagram of the filter
figure(1)
plot(w/pi,20*log10(abs(H)));%parameters are respectively the normalized frequency
and amplitude
xlabel('the normailzed frequency');ylabel('gian/dB');title('The gain response of
lowpass filter');
axis([0 1 -50 0.5]);
f1=100;f2=300;f3=700;fs=2000;%the frequencies of sines signal that needs filtered
and the sample frequency
subplot(211)
t=0:1/fs:0.25;%define the time domain and steplength
s=sin(2*pi*f1*t)+sin(2*pi*f2*t)+sin(2*pi*f3*t);%signal before filtering
plot(t,s);%plot the diagram before filtering
xlabel('time/s');ylabel('amplitude');title('Time-domain diagram before filtering');

42
subplot(212)
Fs=fft(s,512); AFs=abs(Fs);%transform to the frequency domain
f=(0:255)*fs/512;%frequency sampling
plot(f,AFs(1:256));%plot the frequency domain diagram before filtering
xlabel('frequency/Hz');ylabel('amplitude');title('Frequency-domain diagram before
filtering');
figure(3)
sf=filter(h,1,s);%use function filter
subplot(211)
plot(t,sf)%plot the diagram after filtering
xlabel('time/s');ylabel('amplitude');title('Time-domain diagram after filtering')
axis([0.2 0.25 -2 2]);%set the range of image coordinates
subplot(212)
Fsf=fft(sf,512); AFsf=abs(Fsf);%frequency-domain and the amplitude diagram
plot(f,AFsf(1:256))%plot the frequency domain diagram before filtering
xlabel('frequency/Hz');ylabel('amplitude');title('Frequency-domain diagram after
filtering');

43
Time-domain diagram before filtering

2
1
amplitude
-1
-2
0 0.05 0.1 0.15 0.2 0.25
time/s
Frequency-domain diagram before filtering
300
200
amplitude
100
0
0 100 200 300 400 500 600 700 800 900 1000
frequency/Hz
Time-domain diagram after filtering

2
1
amplitude
-1
-2
0.2 0.205 0.21 0.215 0.22 0.225 0.23 0.235 0.24 0.245 0.25
time/s
Frequency-domain diagram after filtering
300
200
amplitude
100
0
0 100 200 300 400 500 600 700 800 900 1000
frequency/Hz

44
Program No. 11.b).

Aim: Write a program to design high pass FIR filter using frequency sampling
method.
M=32;%the number of samples

Wp=0.6*pi;%passband cutoff frequency
m=0:M/2;%the sampling points
Wm=2*pi*m./(M+1);%stopband cutoff frequency
mtr=ceil(Wp*(M+1)/(2*pi));%round to positive part,i.e.ceil(3.5)=4;ceil(-3.2)=-3;
Ad=[Wm>=Wp];
Ad(mtr)=0.28;
Hd=Ad.*exp(-j*0.5*M*Wm);%define frequency-domain sampling vector H(k))
Hd=[Hd conj(fliplr(Hd(2:M/2+1)))];
%fliplr is to realize the fliplr of matrix and conj is the conjugate
h=real(ifft(Hd));%h(n)=IDFT[H(k)]
w=linspace(0,pi,1000);%get 1000 row vectors between 0 and
piH=freqz(h,[1],w);%the amplitude -frequency characteristic diagram of the filter
figure(1)
plot(w/pi,20*log10(abs(H)));%parameters are respectively the noemalized frequency
and amplitude
xlabel('the normailzed frequency');ylabel('gian/dB');title('The gain response of
highpass filter');
axis([0 1 -50 0]);
f1=200;f2=700;f3=800;%the frequencies of sines signal that needs filtered
fs=2000;%the sample frequency
figure(2)
subplot(211)
t=0:1/fs:0.25;%define the time domain and steplength
s=sin(2*pi*f1*t)+sin(2*pi*f2*t)+sin(2*pi*f3*t);%signal before filtering

45
plot(t,s);%plot the diagram before filtering

xlabel('time/s');ylabel('amplitude');title('Time-domain diagram before filtering');
subplot(212)
Fs=fft(s,512);%transform to the frequency domain
AFs=abs(Fs);%the amplitude
plot(f,AFs(1:256));%plot the frequency domain diagram before filtering
xlabel('frequency/Hz');ylabel('amplitude');title('Frequency-domain diagram before
filtering');
figure(3)
sf=filter(h,1,s);%use function filter
subplot(211)
plot(t,sf)%plot the diagram after filtering
xlabel('time/s');ylabel('amplitude');title('Time-domain diagram after filtering')
axis([0.2 0.25 -2 2]);%set the range of image coordinates
subplot(212)
Fsf=fft(sf,512);AFsf=abs(Fsf);
plot(f,AFsf(1:256))%plot the frequency domain diagram before filtering
xlabel('frequency/Hz');ylabel('amplitude');title('Frequency-domain diagram after
filtering');

46
The gain response of highpass filter

0
-5
-10
-15
-20
gian/dB
-25
-30
-35
-40
-45
-50
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
the normailzed frequency
Time-domain diagram before filtering

4
2
amplitude
-2
-4
0 0.05 0.1 0.15 0.2 0.25
time/s
Frequency-domain diagram before filtering
300
200
amplitude
100
0
0 100 200 300 400 500 600 700 800 900 1000
frequency/Hz

47
Time-domain diagram after filtering

2
1
amplitude
-1
-2
0.2 0.205 0.21 0.215 0.22 0.225 0.23 0.235 0.24 0.245 0.25
time/s
Frequency-domain diagram after filtering
300
200
amplitude
100
0
0 100 200 300 400 500 600 700 800 900 1000
frequency/Hz

48
Program No. 12.a).

Aim: Write a program to Realize direct form I and II of digital filters.
%realization of Direct form one & two digital filters

function y= direct(typ,b,a,x);
x= input('enter the input sequence=');
b= input('enter the numerator polynomials=');
a= input('enter the denominator polynomials=');
typ=input('type of realization=');
p=length(a)-1;
q=length(b)-1;
pq=max(p,q);
a=a(2:p+1);
u=zeros(1,pq);
if(typ==1)
for i=1:length(x),unew=x(i)-sum(u(1:p).*a);
u=[unew,u];
y(i)=sum(u(1:q+1).*b);
u=u(1:pq);
end
if(typ==2)
fori=1:length(x),y(i)=b(1)*x(i)+u(1);
u=[u(2:pq),0];
u(1:q)=u(1:q)+b(2:q+1)*x(i);
u(1:p)=u(1:p)-a*y(i);
end
end

49
Output:
enter the input sequence=[1 1 2 2]
enter the numerator polynomials=[1 2 3 4 ]
enter the denominator polynomials=[4 3 2 1 ]
type of realization=1
ans = 1 0 5 -3

50
Program No. 12.b).

Aim: Write a program to Realize parallel form of digital filters.
%parallel realization of digital filters

function y= parallel(c,nsec,dsec,x);
x= input('enter the input sequence=');
b= input('enter the numerator polynomials=');
a= input('enter the denominator polynomials=');
c= input('enter the gain of the filter=');
[n,m]=size(a);a=a(:,2:3);
u=zeros(n,2);
for i=1:length(x),y(i)=c*x(i);
for k=1:n,
unew=x(i)-sum(u(k,:).*a(k,:));
u(k,:)=[unew,u(k,1)];
y(i)=y(i)+sum(u(k,:).*b(k,:));
end

51
DSP Lab - viva questions

1. What is MATLAB?
2. What are the applications of MATLAB?
3. State sampling theorem.
4. What is meant by Nyquist rate and Nyquist criteria?
5. Explain scaling and superposition properties of a system.
6. What is meant by linearity of a system and how it is related to scaling and
superposition?
7. What is impulse function?
8. What is meant by impulse response?
9. What is energy signal? How to calculate energy of a signal?
10. What is power signal? How to calculate power of a signal?
11. Differentiate between even and odd signals.
12. Explain time invariance property of a system with an example.
13. What is memory less system?
14. When a system is said to have memory?
15. What is meant by causality?
16. Explain linear convolution and circular convolution.
17. What is the length of linear and circular convolutions if the two sequences are having
the length n1 and n2?
18. What are Fourier series and Fourier transform?
19. What are the advantages and special applications of Fourier transform, Fourier series,
Z transform and Laplace transform?
20. Differentiate between DTFT and DFT. Why it is advantageous to use DFT in
computers rather than DTFT?
21. In DTFT, frequency appears to be continuous. But, in DFT, frequency is discrete. This
property is useful for computation in computers.

52
22. How to perform linear convolution using circular convolution?

If two signals x (n) and y (n) are of length n1 and n2, then the linear convoluted output
z (n) is of length n1+n2-1. Each of the input signals is padded with zeros to make it of
length n1+n2-1. Then circular convolution is done on zero padded sequences to get the
linear convolution of original input sequences x (n) and y (n).
23. What is meant by correlation?
Correlation is the measure of similarity between two signal/waveforms. It compares
the waveforms at different time instants.
24. What is auto-correlation?
It is a measure of similarity of similarity of a signal/waveform with itself.
25. What is cross-correlation?
26. What are the advantages of using autocorrelation and cross correlation properties in
signal processing fields?
27. How auto-correlation can be used to detect the presence of noise?
28. Differentiate between IIR filters and FIR filters.
Advantages
FIR IIR
 Stable
 Highly precise  cost lesser
 Finite duration impulse response  Faster computations
 Excellent phase response  Less hardware, computations
 The word-size effect such as round-  Easier to design
off noise and coefficient  Lower order required
quantization errors are much less
severe in FIR.

53
Disadvantages
FIR IIR
 Require higher order  Sensitive to data
 Increased hardware round off and cutoff
 More computations  Make become
 Larger input and unstable
output delays  Poor phase response
 Cost more
29. What is the procedure to design a digital Butterworth filter?

30. What is the difference between Butterworth, Chebyshev I and Chebyshev II filters?
31. What are difference equations and differential equations?
32. What is non real time processing?
Raw Refined
Collector Processor
Data Data
Signal
33. What is meant by real time processing?
 Ability to collect, analyze, and modify signals in real-time
 Real-Time: As these signals are occurring
 We can analyze and process signals while collecting them, not at a later time.
Real-Time
Signal Refined
Processor Data
Signal
34. What is a Digital Signal Processor (DSP)?

Microprocessor specifically designed to perform fast DSP operations (e.g., Fast Fourier
Transforms, inner products, Multiply & Accumulate)
 Good at arithmetic operations (multiplication/division)

54
 Mostly programmed with Assembly and C through Integrated Development

Environment (IDE)
35. Differentiate between RISC and CISC architectures.
RISC Emphasis Single- large Better C

on clock, code compilers
software reduced size
instruction
only
CISC Emphasis Includes Small Poor C

on multi-clock code compilers
hardware complex sizes
instructions
36. Differentiate between General purpose MPU(Micro Processor Unit) and DSP Processor
MPU are built for a range of general-purpose functions such as:
Data manipulation
Math calculations
Control systems
They run large blocks of software
They are used in real-time and in unreal-time systems
DSPs are single-minded, dedicated to:
Perform mathematical calculations
Small blocks of software
Have a predictable execution time
Real-time only
Could assist a general-purpose host MPU
Microprocessor DSP
General purpose Arithmetic
Fixed internal format Varying internal format
Single memory access Multiple memory access
General addressing mode Special addressing mode
Very large external memory Very large internal memory

55
37. What is pipelining?
Pipeline
Description
Stage
Generate program fetch address
PF
Read opcode
Route opcode to functional unit
D
Decode instruction
E Execute instruction
38. What is parallel processing?

39. What is MAC?
40. What is barrel shifter? Why it is advantageous to use it in DSP processor?
41. Differentiate between floating point DSP and fixed point DSP.
42. Fixed Point/Floating Point
 fixed point processor are :
i. cheaper
ii. smaller
iii. less power consuming
iv. Harder to program
1. Watch for errors: truncation, overflow, rounding
v.Limited dynamic range

vi.Used in 95% of consumer products
 floating point processors
i.have larger accuracy
ii.are much easier to program
iii.can access larger memory
iv. It is harder to create an efficient program in C on a fixed point processors than on
floating point processors

56
References
Digital Signal Processing – Principles, Jhon G. Proakis 4th Edition,
Pearson
Algorithms, and Applications Dimitris G. Manolakis 2007.
st
1 Edition,
Digital Signal Processing Sanjith K Mitra McGraw Hill
2005
nd
2 Edition,
Digital Signal Processing A.NagoorKani McGraw Hill
2012
nd
2 Edition,
Digital Signal Processing Shaila D. Apte Wiley
2009
st
1 Edition,
Digital Signal Processing Ashok Amberdar Cengage
2007
st
1 Edition,
Digital Signal Processing Tarun Kumar Rawat Oxford
2015
Dr. Ganesh Rao & Vineeta P. Gejji. Texas instruments materials.

57

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% Verification of sampling theorem

fd=input('enter the desired analog frequency');

%Simulate condition for under sampling i.e.,fs1,2*fd

%Condition For Nyquist Sampling fs2=2*fd

GCE, Ramanagaram Dept. of E&E

%Condition For OverSampling fs2=2*fd

GCE, Ramanagaram Dept. of E&E

Aim: Write a program to generate an impulse response of a system using MATLAB

GCE, Ramanagaram Dept. of E&E

plot of impulse response

GCE, Ramanagaram Dept. of E&E

Aim: Write a program to perform linear convolution of given sequences using

GCE, Ramanagaram Dept. of E&E

GCE, Ramanagaram Dept. of E&E

Program No. 4. a).

GCE, Ramanagaram Dept. of E&E

GCE, Ramanagaram Dept. of E&E

Program No. 4. b).

GCE, Ramanagaram Dept. of E&E

Circular convolved output y[n]=[24 22 24 30]

GCE, Ramanagaram Dept. of E&E

GCE, Ramanagaram Dept. of E&E

Program No. 4. C).

% Linear convolution using circular Convolution with zero padding

GCE, Ramanagaram Dept. of E&E

GCE, Ramanagaram Dept. of E&E

N = input('Enter the the value of N ');

GCE, Ramanagaram Dept. of E&E

0 14.5090 0.0000 5.4305 0.0000

5.4305 0.0000 14.5090

0 -93.2732 45.0000 -27.2357 90.0000

27.2357 135.0000 93.2732

GCE, Ramanagaram Dept. of E&E

GCE, Ramanagaram Dept. of E&E

GCE, Ramanagaram Dept. of E&E

title(‘Impulse Response h[n]’);

GCE, Ramanagaram Dept. of E&E

GCE, Ramanagaram Dept. of E&E

Program No. 7. a).

% To find Impulse response

N = input('Length Of Response required=');

a = [1,-1,0.9]; %y(n) coefficients

% time vector for plotting

GCE, Ramanagaram Dept. of E&E

GCE, Ramanagaram Dept. of E&E

Program No. 7. b).

% To find Step response

GCE, Ramanagaram Dept. of E&E

GCE, Ramanagaram Dept. of E&E

GCE, Ramanagaram Dept. of E&E

GCE, Ramanagaram Dept. of E&E

Program No. 9. a).

To design digital Butterworth low pass filter clear all;

GCE, Ramanagaram Dept. of E&E

GCE, Ramanagaram Dept. of E&E

Program No. 9. b).

%To design digital Butterworth high pass filter

GCE, Ramanagaram Dept. of E&E

GCE, Ramanagaram Dept. of E&E

Program No. 9. c).

%To design digital Butterworth Band pass filter

GCE, Ramanagaram Dept. of E&E