CONTENTS

S.No

Experiment Name

Page No.

LIST OF EXPERIMENTS:
1. Basic operations on matrices.
2. Generation on various signals and Sequences
(periodic and aperiodic), such as unit impulse, unit step,
square, sawtooth, triangular, sinusoidal, ramp, sinc.
3. Operations on signals and sequences such as addition,
multiplication, scaling, shifting, folding, computation of
energy and average power.
4. Finding the even and odd parts of signal/sequence
and real and imaginary part of signal.
5. Convolution between signals and sequences.
6. Auto correlation and cross correlation between
signals and sequences.
7. Verification of linearity and time invariance
properties of a given continuous /discrete system.
8. Computation of unit sample, unit step and sinusoidal
response of the given LTI system and verifying its
physical Realizability and stability properties.
9. Gibbs phenomenon.
10. Finding the Fourier transform of a given
signal and plotting its magnitude and phase spectrum.
11. Waveform synthesis using Laplace Transform.
12. Locating the zeros and poles and plotting the
pole zero maps in s-plane and z-plane for the given
transfer function.
13. Generation of Gaussian Noise(real and complex),
computation of its mean, M.S. Value and its skew,
kurtosis, and PSD, probability distribution function.
14. Sampling theorem verification.
15. Removal of noise by auto correlation/cross correlation.
16. Extraction of periodic signal masked by noise
using correlation.
17. Verification of Weiner-Khinchine relations.
18. Checking a random process for stationarity in wide sense.

1
6
18
22
26
29
33
40
43
48
53
54
57
66
70
73

EXP.NO: 1

BASIC OPERATIONS ON MATRICES

Aim: To generate matrix and perform basic operation on matrices Using MATLAB
Software.
EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software
MATLAB on Matrices
MATLAB treats all variables as matrices. For our purposes a matrix can be thought of as
an array, in fact, that is how it is stored.
Vectors are special forms of matrices and contain only one
row OR one column.
Scalars are matrices with only one row AND one column.A matrix with only one
row AND one column is a scalar. A scalar can be reated in MATLAB as follows:
a_value=23
a_value =23
A matrix with only one row is called a row vector. A row vector can be created in
MATLAB as follows :
rowvec = [12 , 14 , 63]
rowvec =
12 14 63
A matrix with only one column is called a column vector. A column vector can be
created in MATLAB as follows:
colvec = [13 ; 45 ; -2]
colvec =
13
45
-2
A matrix can be created in MATLAB as follows:
matrix = [1 , 2 , 3 ; 4 , 5 ,6 ; 7 , 8 , 9]
matrix =
123
456
789
Extracting a Sub-Matrix
A portion of a matrix can be extracted and stored in a smaller matrix by specifying
the names of both matrices and the rows and columns to extract. The syntax is:
sub_matrix = matrix ( r1 : r2 , c1 : c2 ) ;
Where r1 and r2 specify the beginning and ending rows and c1 and c2 specify the
beginning and ending columns to be extracted to make the new matrix.

 A column vector can beextracted from a matrix.

As an example we create a matrix below:
matrix=[1,2,3;4,5,6;7,8,9]
matrix =
123
456
789
Here we extract column 2 of the matrix and make a column vector:
col_two=matrix( : , 2)
col_two =
258
A row vector can be extracted from a matrix.
As an example we create a matrix below:
matrix=[1,2,3;4,5,6;7,8,9]
matrix =
123
456
789
Here we extract row 2 of the matrix and make a row vector. Note that the 2:2
specifies the second row and the 1:3 specifies which columns of the row.
rowvec=matrix(2 : 2 , 1 :3)
rowvec =4 5 6
a=3;
b=[1, 2, 3;4, 5, 6]
b=
123
456
c= b+a % Add a to each element of b
c=
456
789
Scalar - Matrix Subtraction
a=3;
b=[1, 2, 3;4, 5, 6]
b=
123
456
c = b - a %Subtract a from each element of b
c=
-2 -1 0
123
Scalar - Matrix Multiplication
a=3;
b=[1, 2, 3; 4, 5, 6]
b=
123

456
c = a * b % Multiply each element of b by a
c=
369
12 15 18
Scalar - Matrix Division
a=3;
b=[1, 2, 3; 4, 5, 6]
b=
123
456
c = b / a % Divide each element of b by a
c=
0.3333 0.6667 1.0000
1.3333 1.6667 2.0000
a = [1 2 3 4 6 4 3 4 5]
a=
1

b=a+2
b=
3

A = [1 2 0; 2 5 -1; 4 10 -1]
A=
1

-1

10

-1

10

-1

-1

B = A'
B=

C = A* B
C=

12

24

12

30

59

24

59 117

Instead of doing a matrix multiply, we can multiply the corresponding elements of

two matrices or vectors using the .* operator.
C = A .* B
C=
1

25 -10

0 -10

Let's find the inverse of a matrix

X = inv(A)
X=
5

-2

-2

-1

-2

and then illustrate the fact that a matrix times its inverse is the identity matrix.
I = inv(A) * A
I=
1

to obtain eigenvalues
eig(A)
ans =
3.7321

0.2679
1.0000
as the singular value decomposition.
svd(A)
ans =

12.3171
0.5149
0.1577

CONCLUSION: Inthis experiment basic operations on matrices Using MATLAB

have been demonstrated.

3.perform following operations on any two matrices

A+B
A-B
A*B
A.*B
A/B
A./B
A\B
A.\B
A^B,A.^B,A',A.

EXP.NO: 2

GENERATION ON VARIOUS SIGNALS AND SEQUENCES

(PERIODIC AND APERIODIC), SUCH AS UNIT IMPULSE, UNIT
STEP, SQUARE, SAWTOOTH, TRIANGULAR, SINUSOIDAL,
RAMP, SINC.
Aim: To generate different types of signals Using MATLAB Software.
EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software
THEORY :
UNIT IMPULSE:

a) Continous signal:

(t )

And

t0
t0

(t ) dt 1

Also called unit impulse function. The value of delta function can also be defined
in the sense of generalized function
(t): Test Function
b) Unit Sample sequence: (n)={ 1,
0,

n=0
n0

i.e

Matlab program:
%unit impulse generation
clc
close all
n1=-3;
n2=4;
n0=0;
n=[n1:n2];
x=[(n-n0)==0]
stem(n,x)

2)Unit Step Function u(t):

u (t ) (t ) dt

(t ) dt

1
0

u (t )

b)Unit Step Sequence u(n): )=

1,
0,

n0
n< 0

% unit step generation

n1=-4;
n2=5;
n0=0;
[y,n]=stepseq(n0,n1,n2);
stem(n,y);
xlabel('n')

t0
t0

ylabel('amplitude');
title('unit step');

Square waves: Like sine waves, square waves are described in terms of period,
frequency and amplitude:

Peak amplitude, Vp , and peak-to-peak amplitude, Vpp , are measured as you might
expect. However, the rms amplitude, Vrms , is greater than that of a sine wave. Remember
that the rms amplitude is the DC voltage which will deliver the same power as the signal.
If a square wave supply is connected across a lamp, the current flows first one way and
then the other. The current switches direction but its magnitude remains the same. In other
words, the square wave delivers its maximum power throughout the cycle so that Vrms is
equal to Vp . (If this is confusing, don't worry, the rms amplitude of a square wave is not
something you need to think about very often.)
Although a square wave may change very rapidly from its minimum to maximum
voltage, this change cannot be instaneous. The rise time of the signal is defined as the
time taken for the voltage to change from 10% to 90% of its maximum value. Rise times
are usually very short, with durations measured in nanoseconds (1 ns = 10-9 s), or
microseconds (1 s = 10-6 s), as indicated in the graph

% square wave wave generator

fs = 1000;
t = 0:1/fs:1.5;
x1 = sawtooth(2*pi*50*t);
x2 = square(2*pi*50*t);
subplot(2,2,1),plot(t,x1), axis([0 0.2 -1.2 1.2])
xlabel('Time (sec)');ylabel('Amplitude'); title('Sawtooth Periodic Wave')
subplot(2,2,2),plot(t,x2), axis([0 0.2 -1.2 1.2])
xlabel('Time (sec)');ylabel('Amplitude'); title('Square Periodic Wave');
subplot(2,2,3),stem(t,x2), axis([0 0.1 -1.2 1.2])
xlabel('Time (sec)');ylabel('Amplitude');

SAW TOOTH:
The sawtooth wave (or saw wave) is a kind of non-sinusoidal waveform. It is
named a sawtooth based on its resemblance to the teeth on the blade of a saw. The
convention is that a sawtooth wave ramps upward and then sharply drops. However, there
are also sawtooth waves in which the wave ramps downward and then sharply rises. The
latter type of sawtooth wave is called a 'reverse sawtooth wave' or 'inverse sawtooth
wave'. As audio signals, the two orientations of sawtooth wave sound identical. The
piecewise linear function based on the floor function of time t, is an example of a
sawtooth wave with period 1.

A more general form, in the range 1 to 1, and with period a, is

This sawtooth function has the same phase as the sine function. A sawtooth wave's
sound is harsh and clear and its spectrum contains both even and odd harmonics of the

fundamental frequency. Because it contains all the integer harmonics, it is one of the best
waveforms to use for synthesizing musical sounds, particularly bowed string instruments
like violins and cellos, using subtractive synthesis.
Applications
The sawtooth and square waves are the most common starting points used to
create sounds with subtractive analog and virtual analog music synthesizers.
The sawtooth wave is the form of the vertical and horizontal deflection signals
used to generate a raster on CRT-based television or monitor screens. Oscilloscopes also
use a sawtooth wave for their horizontal deflection, though they typically use electrostatic
deflection.
On the wave's "ramp", the magnetic field produced by the deflection yoke drags
the electron beam across the face of the CRT, creating a scan line.
On the wave's "cliff", the magnetic field suddenly collapses, causing the electron
beam to return to its resting position as quickly as possible.
The voltage applied to the deflection yoke is adjusted by various means
(transformers, capacitors, center-tapped windings) so that the half-way voltage on the
sawtooth's cliff is at the zero mark, meaning that a negative voltage will cause deflection
in one direction, and a positive voltage deflection in the other; thus, a center-mounted
deflection yoke can use the whole screen area to depict a trace. Frequency is 15.734 kHz
on NTSC, 15.625 kHz for PAL and SECAM)

% sawtooth wave generator

fs = 10000;
t = 0:1/fs:1.5;
x = sawtooth(2*pi*50*t);
subplot(1,2,1);
plot(t,x), axis([0 0.2 -1 1]);
xlabel('t'),ylabel('x(t)')
title('sawtooth signal');
N=2; fs = 500;n = 0:1/fs:2;
x = sawtooth(2*pi*50*n);
subplot(1,2,2);
stem(n,x), axis([0 0.2 -1 1]);
xlabel('n'),ylabel('x(n)')
title('sawtooth sequence');

10

Triangle wave
A triangle wave is a non-sinusoidal waveform named for its triangular shape.A
bandlimited triangle wave pictured in the time domain (top) and frequency domain
(bottom). The fundamental is at 220 Hz (A2).Like a square wave, the triangle wave
contains only odd harmonics. However, the higher harmonics roll off much faster than in
a square wave (proportional to the inverse square of the harmonic number as opposed to
just the inverse).It is possible to approximate a triangle wave with additive synthesis by
adding odd harmonics of the fundamental, multiplying every (4n1)th harmonic by 1 (or
changing its phase by ), and rolling off the harmonics by the inverse square of their
relative frequency to the fundamental.This infinite Fourier series converges to the triangle
wave:

11

To generate a trianguular pulse

A=2; t = 0:0.0005:1;
x=A*sawtooth(2*pi*5*t,0.25); %5 Hertz wave with duty cycle 25%
plot(t,x);
grid
axis([0 1 -3 3]);

%%To generate a trianguular pulse

fs = 10000;t = -1:1/fs:1;
x1 = tripuls(t,20e-3); x2 = rectpuls(t,20e-3);
subplot(211),plot(t,x1), axis([-0.1 0.1 -0.2 1.2])
xlabel('Time (sec)');ylabel('Amplitude'); title('Triangular Aperiodic Pulse')
subplot(212),plot(t,x2), axis([-0.1 0.1 -0.2 1.2])
xlabel('Time (sec)');ylabel('Amplitude'); title('Rectangular Aperiodic Pulse')
set(gcf,'Color',[1 1 1]),

12

%%To generate a rectangular pulse

t=-5:0.01:5;
pulse = rectpuls(t,2); %pulse of width 2 time units
plot(t,pulse)
axis([-5 5 -1 2]);
grid

13

Sinusoidal Signal Generation

The sine wave or sinusoid is a mathematical function that describes a smooth repetitive
oscillation. It occurs often in pure mathematics, as well as physics, signal processing,
electrical engineering and many other fields. Its most basic form as a function of time (t)
is:
where:

A, the amplitude, is the peak deviation of the function from its center position.
, the angular frequency, specifies how many oscillations occur in a unit time
interval, in radians per second
, the phase, specifies where in its cycle the oscillation begins at t = 0.

A sampled sinusoid may be written as:

14

where f is the signal frequency, fs is the sampling frequency, is the phase and A is the
amplitude of the signal. The program and its output is shown below:
Note that there are 64 samples with sampling frequency of 8000Hz or sampling time
of 0.125 mS (i.e. 1/8000). Hence the record length of the signal is 64x0.125=8mS.
There are exactly 8 cycles of sinewave, indicating that the period of one cycle is 1mS
which means that the signal frequency is 1KHz.

% sinusoidal signal
N=64; % Define Number of samples
n=0:N-1; % Define vector n=0,1,2,3,...62,63
f=1000; % Define the frequency
fs=8000; % Define the sampling frequency
x=sin(2*pi*(f/fs)*n); % Generate x(t)
plot(n,x); % Plot x(t) vs. t
title('Sinewave [f=1KHz, fs=8KHz]');
xlabel('Sample Number');
ylabel('Amplitude');

% RAMP
clc
close all
n=input('enter the length of ramp');
t=0:n;

15

plot(t);
xlabel('t');
ylabel('amplitude');
title ('ramp')

SINC FUNCTION:
The sinc function computes the mathematical sinc function for an input vector or
matrix x. Viewed as a function of time, or space, the sinc function is the inverse Fourier
transform of the rectangular pulse in frequency centered at zero of width 2 and height 1.
The following equation defines the sinc function:

The sinc function has a value of 1 whenx is equal to zero, and a value of

16

% sinc
x = linspace(-5,5);
y = sinc(x);
subplot(1,2,1);plot(x,y)
xlabel(time);
ylabel(amplitude);
title(sinc function);
subplot(1,2,2);stem(x,y);
xlabel(time);
ylabel(amplitude);
title(sinc function);

CONCLUSION: In this experiment various signals have been generated Using

MATLAB
Exersize questions:generate following signals using MATLAB

1.x(t)=e-t
2.x(t)= t 2 / 2
3.generate rectangular pulse function

4.generate signum sunction sinc(t)= 1 t > 0

17

t=0

-1

t<0

5. generate complex exponential signal x(t)= e st for different values of and

EXP.NO: 3

OPERATIONS ON SIGNALS AND SEQUENCES SUCH AS ADDITION,

MULTIPLICATION, SCALING, SHIFTING, FOLDING,
COMPUTATION OF ENERGY AND AVERAGE POWER
Aim: To perform arithmetic operations different types of signals Using MATLAB
Software.
EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software
THEORY :

Basic Operation on Signals:

Time shifting: y(t)=x(t-T)The effect that a time shift has on the appearance of a signal If T is a
positive number, the time shifted signal, x (t -T ) gets shifted to the right, otherwise it gets shifted left.
Signal Shifting and Delay:

. Shifting : y(n)={x(n-k)} ;

Time reversal:

m=n-k; y=x;

Y(t)=y(-t) Time reversal _ips the signal about t = 0 as seen in Figure 1.

Signal Addition and Substraction :

Addition: any two signals can be added to form a third signal,
z (t) = x (t) + y (t)

Signal Amplification/Attuation :

18

Multiplication/Divition : of two signals, their product is also a signal.

folding:

y(n)={x(-n)} ;

y=fliplr(x); n=-fliplr(n);

%plot the 2 Hz sine wave in the top panel

t = [0:.01:1];
% independent (time) variable
A = 8;
% amplitude
f1 = 2;
% create a 2 Hz sine wave lasting 1 sec
s1 = A*sin(2*pi*f1*t);
f2 = 6;
% create a 4 Hz sine wave lasting 1 sec
s2 = A*sin(2*pi*f2*t);
figure
subplot(4,1,1)
plot(t, s1)
title('1 Hz sine wave')
ylabel('Amplitude')
%plot the 4 Hz sine wave in the middle panel
subplot(4,1,2)
plot(t, s2)
title('2 Hz sine wave')
ylabel('Amplitude')
%plot the summed sine waves in the bottom panel
subplot(4,1,3)
plot(t, s1+s2)
title('Summed sine waves')
ylabel('Amplitude')
xlabel('Time (s)')
xmult=s1.*s2;
subplot(4,1,4);
plot(xmult);

19

z (t) = x (t) y (t)

title('multiplication');
ylabel('Amplitude')
xlabel('Time (s)')

%signal folding
clc; clear all
t=0:0.1:10;
x=0.5*t;
lx=length(x);
nx=0:lx-1;
xf=fliplr(x);
nf=-fliplr(nx);
subplot(2,1,1);
stem(nx,x);
xlabel('nx');
ylabel('x(nx)');
title('original signal');
subplot(2,1,2);
stem(nf,xf);
xlabel('nf');
ylabel('xf(nf)');
title('folded signal');

20

%plot the 2 Hz sine wave scalling

t = [0:.01:1];
% independent (time) variable
A = 8;
% amplitude
f1 = 2;
% create a 2 Hz sine wave lasting 1 sec
s1 = A*sin(2*pi*f1*t);
subplot(3,2,1)
plot(s1);
xlabel('t');
ylabel('amplitude');
s2=2*s1;
subplot(3,2,2)
plot(s2);
xlabel('t');
ylabel('amplitude');
s3=s1/2;
subplot(3,2,3)
plot(s3);

21

xlabel('t');
ylabel('amplitude');
subplot(3,2,4)
stem(s1);
xlabel('t');
ylabel('amplitude');
s2=2*s1;
subplot(3,2,5)
stem(s2);
xlabel('t');
ylabel('amplitude');
s3=s1/2;
subplot(3,2,6)
stem(s3);
xlabel('t');
ylabel('amplitude');

Excersize questions: Sketch the following questions using MATLAB

1.
2.
3.
4.
5.

x(t)= u(-t+1)
x(t)=3r(t-1)
x(t)=U(n+2-u(n-3)
x(n)=x1(n)+x2(n)where x1(n)={1,3,2,1},x2(n)={1,-2,3,2}
x(t)=r(t)-2r(t-1)+r(t-2)

22

6. x(n)=2(n+2)-2(n-4), -5 n 5.
7. X(n)={1,2,3,4,5,6,7,6,5,4,2,1} determine and plot the following sequence
a. x1(n)=2x(n-5-3x(n+4))
b. x2(n)=x(3-n)+x(n)x(n-2)

CONCLUSION: Inthis experiment the various oprations on signals have been

performedUsing MATLAB have been demonstrated
EXP.NO: 4

FINDING THE EVEN AND ODD PARTS OF SIGNAL/SEQUENCE

AND REAL AND IMAGINARY PART OF SIGNAL
Aim: program for finding even and odd parts of signals Using MATLAB Software.
EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software
THEORY :

Even and Odd Signal

One of characteristics of signal is symmetry that may be useful for signal analysis. Even
signals are symmetric around vertical axis, and Odd signals are symmetric about origin.
Even Signal: A signal is referred to as an even if it is identical to its time-reversed
counterparts; x(t) = x(-t).
Odd Signal: A signal is odd if x(t) = -x(-t).
An odd signal must be 0 at t=0, in other words, odd signal passes the origin.
Using the definition of even and odd signal, any signal may be decomposed into a sum of
its even part, xe(t), and its odd part, xo(t), as follows:

It is an important fact because it is relative concept of Fourier series. In Fourier series, a

periodic signal can be broken into a sum of sine and cosine signals. Notice that sine
function is odd signal and cosine function is even signal.

%even and odd signals program:

23

t=-4:1:4;
h=[ 2 1 1 2 0 1 2 2 3 ];
subplot(3,2,1)
stem(t,h);
xlabel('time'); ylabel('amplitude');
title('signal');
n=9;
for i=1:9
x1(i)=h(n);
n=n-1;
end
subplot(3,2,2)
stem(t,x1);
xlabel('time'); ylabel('amplitude');
title('folded signal');
z=h+x1
subplot(3,2,3);
stem(t,z);
xlabel('time'); ylabel('amplitude');
title('sum of two signal');
subplot(3,2,4);
stem(t,z/2);
xlabel('time'); ylabel('amplitude');
title('even signal');
a=h-x1;
subplot(3,2,5);
stem(t,a);
xlabel('time'); ylabel('amplitude');
title('difference of two signal');
subplot(3,2,6);
stem(t,a/2);
xlabel('time'); ylabel('amplitude');
title('odd signal');

24

ENERGY AND POWER SIGNAL:A signal can

be categorized into energy signal or power

signal: An energy signal has a finite energy, 0 < E < . In other words, energy signals
have values only in the limited time duration. For example, a signal having only one
square pulse is energy signal. A signal that decays exponentially has finite energy, so, it is
also an energy signal. The power of an energy signal is 0, because of dividing finite
energy by infinite time (or length).

On the contrary, the power signal is not limited in time. It always exists from beginning
to end and it never ends. For example, sine wave in infinite length is power signal. Since
the energy of a power signal is infinite, it has no meaning to us. Thus, we use power
(energy per given time) for power signal, because the power of power signal is finite, 0 <
P < .

% energy
clc;
close all;

25

clear all;
x=[1,2,3];
n=3
e=0;
for i=1:n;
e=e+(x(i).*x(i));
end
% energy
clc;
close all;
clear all;
N=2
x=ones(1,N)
for i=1:N
y(i)=(1/3)^i.*x(i);
end
n=N;
e=0;
for i=1:n;
e=e+(y(i).*y(i));
end
% power
clc;
close all;
clear all;
N=2
x=ones(1,N)
for i=1:N
y(i)=(1/3)^i.*x(i);
end
n=N;
e=0;
for i=1:n;
e=e+(y(i).*y(i));
end
p=e/(2*N+1);
% power
N=input('type a value for N');
t=-N:0.0001:N;
x=cos(2*pi*50*t).^2;
disp('the calculated power p of the signal is');
P=sum(abs(x).^2)/length(x)
plot(t,x);
axis([0 0.1 0 1]);

26

disp('the theoretical power of the signal is');

P_theory=3/8

CONCLUSION: In this experiment even and odd parts of various signals and energy
and power of signals have been calculated Using MATLAB
exersize questions: find even and odd parts of the following signals

1.x(n)= e(-0.1+j0.3)n ,-

10n10

2.calculate the following signals energies

1..x(t)=u(t)-u(t-15)
2.cos(10t)u(t)u(2-t-5)
3 calculate the following signals energies
1.
2.
3.

x(t)=

j(2t+/4)

x(t)=cos(t)
x(t)=cos(/4 n)

27

EXP.NO: 5
LINEAR CONVOLUTION
Aim: To find the out put with linear convolution operation Using MATLAB Software.
EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software
Theory:
Linear Convolution involves the following operations.
1. Folding
2. Multiplication
3. Addition
4. Shifting
These operations can be represented by a Mathematical Expression as follows:

x[ ]= Input signal Samples

h[ ]= Impulse response co-efficient.
y[ ]= Convolution output.
n = No. of Input samples
h = No. of Impulse response co-efficient.
Example : X(n)={1 2 -1 0 1}, h(n)={ 1,2,3,-1}
Program:
clc;
close all;
clear all;
x=input('enter input sequence');
h=input('enter impulse response');
y=conv(x,h);
subplot(3,1,1);
stem(x);
xlabel('n');ylabel('x(n)');
title('input signal')
subplot(3,1,2);
stem(h);
xlabel('n');ylabel('h(n)');
title('impulse response')
subplot(3,1,3);
stem(y);
xlabel('n');ylabel('y(n)');
title('linear convolution')

28

disp('The resultant signal is');

disp(y)
linear convolution
output:
enter input sequence[1 4 3 2]
enter impulse response[1 0 2 1]
The resultant signal is
1 4 5 11 10 7 2

29

CONCLUSION: In this experiment convolution of various signals have been

performed Using MATLAB
Applications:Convolution is used to obtain the response of an LTI system to an arbitrary
input signal.It is used to find the filter response and finds application in speech processing
and radar signal processing.
Excersize questions: perform convolution between the following signals
1. X(n)=[1 -1 4 ], h(n) = [ -1 2 -3 1]
2. perform convolution between the. Two periodic sequences
x1(t)=e-3t{u(t)-u(t-2)} , x2(t)= e -3t for 0 t 2

30

EXP.NO: 6

6. AUTO CORRELATION AND CROSS CORRELATION BETWEEN

SIGNALS AND SEQUENCES.

Aim: To compute auto correlation and cross correlation between signals and

sequences
EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software
Theory:
Correlations of sequences:
It is a measure of the degree to which two sequences are similar. Given two realvalued sequences x(n) and y(n) of finite energy,
Convolution involves the following operations.
1. Shifting
2. Multiplication
3. Addition
These operations can be represented by a Mathematical Expression as follows:
Crosscorrelation

rx , y (l )

x ( n) y ( n l )

The index l is called the shift or lag parameter

Autocorrelation
rx , x (l )

x (n) x (n l )

The special case: y(n)=x(n)

% Cross Correlation
clc;
close all;
clear all;
x=input('enter input sequence');
h=input('enter the impulse suquence');
subplot(3,1,1);

31

stem(x);
xlabel('n');
ylabel('x(n)');
title('input signal');
subplot(3,1,2);
stem(h);
xlabel('n');
ylabel('h(n)');
title('impulse signal');
y=xcorr(x,h);
subplot(3,1,3);
stem(y);
xlabel('n');
ylabel('y(n)');
disp('the resultant signal is');
disp(y);
title('correlation signal');

32

% auto correlation
clc;
close all;
clear all;
x = [1,2,3,4,5]; y = [4,1,5,2,6];
subplot(3,1,1);
stem(x);
xlabel('n');
ylabel('x(n)');
title('input signal');
subplot(3,1,2);
stem(y);
xlabel('n');
ylabel('y(n)');
title('input signal');
z=xcorr(x,x);
subplot(3,1,3);
stem(z);
xlabel('n');
ylabel('z(n)');
title('resultant signal signal');

33

CONCLUSION: In this experiment correlation of various signals have been

performed Using MATLAB
Applications:it is used to measure the degree to which the two signals are similar and it is
also used for radar detection by estimating the time delay.it is also used in Digital
communication, defence applications and sound navigation
Excersize questions: perform convolution between the following signals
1. X(n)=[1 -1 4 ], h(n) = [ -1 2 -3 1]
2. perform convolution between the. Two periodic sequences
x1(t)=e-3t{u(t)-u(t-2)} , x2(t)= e -3t for 0 t 2

34

EXP.NO: 7

VERIFICATION OF LINEARITY AND TIME INVARIANCE

PROPERTIES OF A GIVEN CONTINUOUS /DISCRETE SYSTEM.
Aim: To compute linearity and time invariance properties of a given

continuous /discrete system

EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software
THEORY:

LINEARITY PROPERTY

Program1:
clc;
clear all;
close all;
n=0:40; a=2; b=1;
x1=cos(2*pi*0.1*n);
x2=cos(2*pi*0.4*n);
x=a*x1+b*x2;
y=n.*x;
y1=n.*x1;
y2=n.*x2;

35

yt=a*y1+b*y2;
d=y-yt;
d=round(d)
if d
disp('Given system is not satisfy linearity property');
else
disp('Given system is satisfy linearity property');
end
subplot(3,1,1), stem(n,y); grid
subplot(3,1,2), stem(n,yt); grid
subplot(3,1,3), stem(n,d); grid

Program2:
clc;
clear all;
close all;
n=0:40; a=2; b=-3;
x1=cos(2*pi*0.1*n);
x2=cos(2*pi*0.4*n);
x=a*x1+b*x2;
y=x.^2;
y1=x1.^2;
y2=x2.^2;
yt=a*y1+b*y2;

36

d=y-yt;
d=round(d);
if d
disp('Given system is not satisfy linearity property');
else
disp('Given system is satisfy linearity property');
end
subplot(3,1,1), stem(n,y); grid
subplot(3,1,2), stem(n,yt); grid
subplot(3,1,3), stem(n,d); grid

Program
clc;
close all;
clear all;
x=input('enter the sequence');
N=length(x);
n=0:1:N-1;
y=xcorr(x,x);
subplot(3,1,1);

37

stem(n,x);
xlabel(' n----->');ylabel('Amplitude--->');
title('input seq');
subplot(3,1,2);
N=length(y);
n=0:1:N-1;
stem(n,y);
xlabel('n---->');ylabel('Amplitude----.');
title('autocorr seq for input');
disp('autocorr seq for input');
disp(y)
p=fft(y,N);
subplot(3,1,3);
stem(n,p);
xlabel('K----->');ylabel('Amplitude--->');
title('psd of input');
disp('the psd fun:');
disp(p)

LINEAR TIME INVARIENT SYSTEMS(LTI):

Program1:
clc;
close all;
clear all;
n=0:40;
D=10;
x=3*cos(2*pi*0.1*n)-2*cos(2*pi*0.4*n);
xd=[zeros(1,D) x];

38

y=n.*xd(n+D);
n1=n+D;
yd=n1.*x;
d=y-yd;
if d
disp('Given system is not satisfy time shifting property');
else
disp('Given system is satisfy time shifting property');
end
subplot(3,1,1),stem(y),grid;
subplot(3,1,2),stem(yd),grid;
subplot(3,1,3),stem(d),grid;

Program2:
clc;
close all;
clear all;
n=0:40;

39

D=10;
x=3*cos(2*pi*0.1*n)-2*cos(2*pi*0.4*n);
xd=[zeros(1,D) x];
x1=xd(n+D);
y=exp(x1);
n1=n+D;
yd=exp(xd(n1));
d=y-yd;
if d
disp('Given system is not satisfy time shifting property');
else
disp('Given system is satisfy time shifting property');
end
subplot(3,1,1),stem(y),grid;
subplot(3,1,2),stem(yd),grid;
subplot(3,1,3),stem(d),grid;

40

CONCLUSION: In this experiment Linearity and Time invariance property of given

system has bees verified performed Using MATLAB
Applications:it is used to measure the degree to which the two signals are similar and it is
also used for radar detection by estimating the time delay.it is also used in Digital
communication defence applications and sound navigation
Excersize questions: perform convolution between the following signals
1. X(n)=[1 -1 4 ], h(n) = [ -1 2 -3 1]
2. perform convolution between the. Two periodic sequences
x1(t)=e-3t{u(t)-u(t-2)} , x2(t)= e -3t for 0 t 2

41

EXP.NO:8

COMPUTATION OF UNIT SAMPLE, UNIT STEP AND SINUSOIDAL

RESPONSE OF THE GIVEN LTI SYSTEM AND VERIFYING ITS
PHYSICAL REALIZABILITY AND STABILITY PROPERTIES.
Aim: To Unit Step And Sinusoidal Response Of The Given LTI System And Verifying Its
Physical Realizability And Stability Properties.
EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software
A discrete time system performs an operation on an input signal based on predefined
criteria to produce a modified output signal. The input signal x(n) is the system excitation,
and y(n) is the system response. The transform operation is shown as,

If the input to the system is unit impulse i.e. x(n) = (n) then the output of the system is
known as impulse response denoted by h(n) where,
h(n) = T[(n)]
we know that any arbitrary sequence x(n) can be represented as a weighted sum of
discrete impulses. Now the system response is given by,

For linear system (1) reduces to

%given difference equation y(n)-y(n-1)+.9y(n-2)=x(n);

42

%calculate and plot the impulse response and step response

b=[1];
a=[1,-1,.9];
x=impseq(0,-20,120);
n = [-20:120];
h=filter(b,a,x);
subplot(3,1,1);stem(n,h);
title('impulse response');
xlabel('n');ylabel('h(n)');
=stepseq(0,-20,120);
s=filter(b,a,x);
s=filter(b,a,x);
subplot(3,1,2);
stem(n,s);
title('step response');
xlabel('n');ylabel('s(n)')
t=0:0.1:2*pi;
x1=sin(t);
%impseq(0,-20,120);
n = [-20:120];
h=filter(b,a,x1);
subplot(3,1,3);stem(h);
title('sin response');
xlabel('n');ylabel('h(n)');
figure;
zplane(b,a);

43

44

CONCLUSION: In this experiment computation of unit sample, unit step and

sinusoidal response of the given lti system and verifying its physical
realizability and stability properties Using MATLAB

45

EXP.NO: 9

GIBBS PHENOMENON
Aim: To verify the Gibbs Phenomenon.
EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software
the Gibbs phenomenon, the Fourier series of a piecewise continuously differentiable
periodic function behaves at a jump discontinuity.the n the approximated function shows
amounts of ripples at the points of discontinuity. This is known as the Gibbs Phenomina .
partial sum of the Fourier series has large oscillations near the jump, which might
increase the maximum of the partial sum above that of the function itself. The overshoot
does not die out as the frequency increases, but approaches a finite limit
The Gibbs phenomenon involves both the fact that Fourier sums overshoot at a jump
discontinuity, and that this overshoot does not die out as the frequency increases
Gibbs Phenomina Program :
t=0:0.1:(pi*8);
y=sin(t);
subplot(5,1,1);
plot(t,y);
xlabel('k');
ylabel('amplitude');
title('gibbs phenomenon');
h=2;
%k=3;
for k=3:2:9
y=y+sin(k*t)/k;
subplot(5,1,h);
plot(t,y);
xlabel('k');
ylabel('amplitude');
h=h+1;
end

46

CONCLUSION: In this experiment Gibbs phenomenon have been demonstrated

Using MATLAB

47

EXP.NO: 10.

FINDING THE FOURIER TRANSFORM OF A GIVEN SIGNAL AND

PLOTTING ITS MAGNITUDE AND PHASE SPECTRUM
Aim: to find the fourier transform of a given signal and plotting its

magnitude and phase spectrum

EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software

Fourier Transform TheoremsL:

We may use Fourier series to motivate the Fourier transform as follows. Suppose that is
a function which is zero outside of some interval [L/2, L/2]. Then for any T L we may
expand in a Fourier series on the interval [T/2,T/2], where the "amount" of the wave
e2inx/T in the Fourier series of is given by
By definition

PROGRAM:

48

Aim: To compute N-point FFT

EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software

Theory:
DFT of a sequence
X[K] =

N 1

x n e

j 2 Kn
N

K 0

Where N= Length of sequence.

K= Frequency Coefficient.
n = Samples in time domain.
FFT : -Fast Fourier transformer .
There are Two methods.
1.Decimation in time (DIT FFT).
2. Decimation in Frequency (DIF FFT).

Program:
clc;
close all;
clear all;
x=input('enter the sequence');
N=length(x);
n=0:1:N-1;
y=fft(x,N)
subplot(2,1,1);
stem(n,x);
title('input sequence');
xlabel('time index n----->');
ylabel('amplitude x[n]----> ');
subplot(2,1,2);
stem(n,y);
title('output sequence');

49

xlabel(' Frequency index K---->');

ylabel('amplitude X[k]------>');

FFT magnitude and Phase plot:

clc
close all
x=[1,1,1,1,zeros(1,4)];
N=8;
X=fft(x,N);
magX=abs(X),phase=angle(X)*180/pi;
subplot(2,1,1)
plot(magX);
grid
xlabel('k')

50

ylabel('X(K)')
subplot(2,1,2)
plot(phase);
grid
xlabel('k')
ylabel('degrees')

Applications:
The no of multiplications in DFT = N2.
The no of Additions in DFT = N(N-1).
For FFT.
The no of multiplication = N/2 log 2N.
The no of additions

= N log2 N.

51

CONCLUSION: In this experiment the fourier transform of a given signal and

plotting its magnitude and phase spectrum have been demonstrated using matlab

Exp:11
LAPLECE TRNASFORMS
Aim: To perform waveform synthesis using Laplece Trnasforms of a given signal

Bilateral Laplace transform :

When one says "the Laplace transform" without qualification, the unilateral or one-sided
transform is normally intended. The Laplace transform can be alternatively defined as the
bilateral Laplace transform or two-sided Laplace transform by extending the limits of
integration to be the entire real axis. If that is done the common unilateral transform
simply becomes a special case of the bilateral transform where the definition of the
function being transformed is multiplied by the Heaviside step function.
The bilateral Laplace transform is defined as follows:

Inverse Laplace transform

The inverse Laplace transform is given by the following complex integral, which is
known by various names (the Bromwich integral, the Fourier-Mellin integral, and
Mellin's inverse formula):

Program for Laplace Transform:

f=t
syms f t;
f=t;
laplace(f)

Program for nverse Laplace Transform

f(s)=24/s(s+8) invese LT
syms F s
F=24/(s*(s+8));

52

ilaplace(F)
y(s)=24/s(s+8) invese LT poles and zeros

Signal synthese using Laplace Tnasform:

clear all
clc
t=0:1:5
s=(t);
subplot(2,3,1)
plot(t,s);
u=ones(1,6)
subplot(2,3,2)
plot(t,u);
f1=t.*u;
subplot(2,3,3)
plot(f1);
s2=-2*(t-1);
subplot(2,3,4);
plot(s2);
u1=[0 1 1 1 1 1];
f2=-2*(t-1).*u1;
subplot(2,3,5);
plot(f2);
u2=[0 0 1 1 1 1];
f3=(t-2).*u2;
subplot(2,3,6);
plot(f3);
f=f1+f2+f3;
figure;
plot(t,f);
% n=exp(-t);
% n=uint8(n);
% f=uint8(f);
% R = int(f,n,0,6)
laplace(f);

53

54

CONCLUSION: In this experiment the Triangular signal synthesised using

Laplece Trnasforms using MATLAB

Applications of laplace transforms:

1. Derive the circuit (differential) equations in the time domain, then transform these
ODEs to the s-domain;
2. Transform the circuit to the s-domain, then derive the circuit equations in the sdomain (using the concept of "impedance").

The main idea behind the Laplace Transformation is that we can solve an equation
(or system of equations) containing differential and integral terms by transforming
the equation in "t-space" to one in "s-space". This makes the problem much easier
to solve

EXP.NO: 12

55

LOCATING THE ZEROS AND POLES AND PLOTTING THE POLE

ZERO MAPS IN S-PLANE AND Z-PLANE FOR THE GIVEN
TRANSFER FUNCTION.
Aim: To locating the zeros and poles and plotting the pole zero maps in s-

plane and z-plane for the given transfer function

EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software

Z-transforms
the Z-transform converts a discrete time-domain signal, which is a sequence of real or
complex numbers, into a complex frequency-domain representation.The Z-transform, like
many other integral transforms, can be defined as either a one-sided or two-sided
transform.

Bilateral Z-transform
The bilateral or two-sided Z-transform of a discrete-time signal x[n] is the function X(z)
defined as

Unilateral Z-transform
Alternatively, in cases where x[n] is defined only for n 0, the single-sided or unilateral
Z-transform is defined as

In signal processing, this definition is used when the signal is causal.

The roots of the equation P(z) = 0 correspond to the 'zeros' of X(z)

The roots of the equation Q(z) = 0 correspond to the 'poles' of X(z)
The ROC of the Z-transform depends on the convergence of the

56

clc;
close all
clear all;
%b= input('enter the numarator cofficients')
%a= input('enter the dinomi cofficients')
b=[1 2 3 4]
a=[1 2 1 1 ]
zplane(b,a);

Applications :Z-Transform is used to find the system responses

CONCLUSION: In this experiment the zeros and poles and plotting the pole
zero maps in s-plane and z-plane for the given transfer function
using MATLAB

57

EXP.NO: 13

13. Gaussian noise

%Estimation of Gaussian density and Distribution Functions
%% Closing and Clearing all
clc;
clear all;
close all;
%% Defining the range for the Random variable
dx=0.01; %delta x
x=-3:dx:3;
[m,n]=size(x);
%% Defining the parameters of the pdf
mu_x=0;
% mu_x=input('Enter the value of mean');
sig_x=0.1; % sig_x=input('Enter the value of varience');
%% Computing the probability density function
px1=[];
a=1/(sqrt(2*pi)*sig_x);
for j=1:n
px1(j)=a*exp([-((x(j)-mu_x)/sig_x)^2]/2);
end
%% Computing the cumulative distribution function
cum_Px(1)=0;
for j=2:n
cum_Px(j)=cum_Px(j-1)+dx*px1(j);
end
%% Plotting the results
figure(1)
plot(x,px1);grid
axis([-3 3 0 1]);
title(['Gaussian pdf for mu_x=0 and sigma_x=', num2str(sig_x)]);
xlabel('--> x')
ylabel('--> pdf')
figure(2)
plot(x,cum_Px);grid
axis([-3 3 0 1]);
title(['Gaussian Probability Distribution Function for mu_x=0 and sigma_x=',
num2str(sig_x)]);
title('\ite^{\omega\tau} = cos(\omega\tau) + isin(\omega\tau)')
xlabel('--> x')
ylabel('--> PDF')

58

59

EXP.NO: 14

14. Sampling theorem verification

Aim: To detect the edge for single observed image using sobel edge detection and canny
edge detection.
EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software
Sampling Theorem:
\A bandlimited signal can be reconstructed exactly if it is sampled at a rate atleast twice
the maximum frequency component in it." Figure 1 shows a signal g(t) that is
bandlimited.

Figure 1: Spectrum of bandlimited signal g(t)

The maximum frequency component of g(t) is fm. To recover the signal g(t) exactly from
its samples it has to be sampled at a rate fs 2fm.
The minimum required sampling rate fs = 2fm is called ' Nyquist rate
Proof: Let g(t) be a bandlimited signal whose bandwidth is fm
(wm = 2fm).

60

Figure 2: (a) Original signal g(t) (b) Spectrum G(w)

(t) is the sampling signal with fs = 1/T > 2fm.

Figure 3: (a) sampling signal (t) ) (b) Spectrum (w)

Let gs(t) be the sampled signal. Its Fourier Transform Gs(w) isgiven by

Figure 4: (a) sampled signal gs(t) (b) Spectrum Gs(w)

61

To recover the original signal G(w):

1. Filter with a Gate function, H2wm(w) of width 2wm
Scale it by T.

Figure 5: Recovery of signal by filtering with a fiter of width 2wm

Aliasing
{ Aliasing is a phenomenon where the high frequency components of the sampled signal
interfere with each other because of inadequate sampling ws < 2wm.

Figure 6: Aliasing due to inadequate sampling

62

Aliasing leads to distortion in recovered signal. This is the

reason why sampling frequency should be atleast twice thebandwidth of the signal.
Oversampling
{ In practice signal are oversampled, where fs is signi_cantly
higher than Nyquist rate to avoid aliasing.

Figure 7: Oversampled signal-avoids aliasing

t=-10:.01:10;
T=4;
fm=1/T;
x=cos(2*pi*fm*t);
subplot(2,2,1);
plot(t,x);
xlabel('time');ylabel('x(t)')
title('continous time signal')
grid;
n1=-4:1:4
fs1=1.6*fm;
fs2=2*fm;
fs3=8*fm;
x1=cos(2*pi*fm/fs1*n1);
subplot(2,2,2);
stem(n1,x1);
xlabel('time');ylabel('x(n)')
title('discrete time signal with fs<2fm')
hold on
subplot(2,2,2);
plot(n1,x1)
grid;
n2=-5:1:5;
x2=cos(2*pi*fm/fs2*n2);
subplot(2,2,3);
stem(n2,x2);
xlabel('time');ylabel('x(n)')
title('discrete time signal with fs=2fm')
hold on
subplot(2,2,3);
plot(n2,x2)
grid;

63

n3=-20:1:20;
x3=cos(2*pi*fm/fs3*n3);
subplot(2,2,4);
stem(n3,x3);
xlabel('time');ylabel('x(n)')
title('discrete time signal with fs>2fm')
hold on
subplot(2,2,4);
plot(n3,x3)
grid;

CONCLUSION: In this experiment the sampling theorem have been verified

using MATLAB

64

EXP.No:15

REMOVAL OF NOISE BY AUTO CORRELATION/CROSS

CORRELATION
Aim: removal of noise by auto correlation/cross correlation
EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software
Detection of a periodic signal masked by random noise is of greate importance .The
noise signal encountered in practice is a signal with random amplitude variations. A signal
is uncorrelated with any periodic signal. If s(t) is a periodic signal and n(t) is a noise
signal then
T/2
Lim 1/T S(t)n(t-T) dt=0
T-- -T/2

for all T

Qsn(T)= cross correlation function of s(t) and n(t) Then Qsn(T)=0

Detection by Auto-Correlation:
S(t)=Periodic Signal mixed with a noise signal n(t).Then f(t) is [s(t ) + n(t) ]
Let Qff(T) =Auto Correlation Function of f(t)
Qss(t) = Auto Correlation Function of S(t)
Qnn(T) = Auto Correlation Function of n(t)
T/2
Qff(T)= Lim 1/T f(t)f(t-T) dt
T--
-T/2
T/2
= Lim 1/T [s(t)+n(t)][s(t-T)+n(t-T)] dt
T--
-T/2
=Qss(T)+Qnn(T)+Qsn(T)+Qns(T)
The periodic signal s(t) and noise signal n(t) are uncorrelated
Qsn(t)=Qns(t)=0 ;Then Qff(t)=Qss(t)+Qnn(t)
The Auto correlation function of a periodic signal is periodic of the same frequency and
the Auto correlation function of a non periodic signal is tends to zero for large value of T
since s(t) is a periodic signal and n(t) is non periodic signal so Qss(T) is a periodic where
as aQnn(T) becomes small for large values of T Therefore for sufficiently large values of
T Qff(T) is equal to Qss(T).

65

Detection by Cross Correlation:

f(t)=s(t)+n(t)
c(t)=Locally generated signal with same frequencyas that of S(t)
Qfc (t)

T/2
= Lim 1/T [s(t)+n(t)] [ c(t-T)] dt
T--
-T/2

= Qsc(T)+Qnc(T)
C(t) is periodic function and uncorrelated with the random noise signal n(t). Hence
Qnc(T0=0) Therefore Qfc(T)=Qsc(T)

a)auto correlation
clear all
clc
t=0:0.1:pi*4;
s=sin(t);
k=2;
subplot(6,1,1)
plot(s);
title('signal s');
xlabel('t');
ylabel('amplitude');
n = randn([1 126]);
f=s+n;
subplot(6,1,2)
plot(f);
title('signal f=s+n');
xlabel('t');
ylabel('amplitude');
as=xcorr(s,s);
subplot(6,1,3)
plot(as);
title('auto correlation of s');
xlabel('t');
ylabel('amplitude');
an=xcorr(n,n);
subplot(6,1,4)
plot(an);
title('auto correlation of n');
xlabel('t');
ylabel('amplitude');
cff=xcorr(f,f);
subplot(6,1,5)

66

plot(cff);
title('auto correlation of f');
xlabel('t');
ylabel('amplitude');
hh=as+an;
subplot(6,1,6)
plot(hh);
title('addition of as+an');
xlabel('t');
ylabel('amplitude');

67

B)CROSS CORRELATION :
clear all
clc
t=0:0.1:pi*4;
s=sin(t);
k=2;
%sk=sin(t+k);
subplot(7,1,1)
plot(s);
title('signal s');xlabel('t');ylabel('amplitude');
c=cos(t);
subplot(7,1,2)
plot(c);
title('signal c');xlabel('t');ylabel('amplitude');
n = randn([1 126]);
f=s+n;
subplot(7,1,3)
plot(f);
title('signal f=s+n');xlabel('t');ylabel('amplitude');
asc=xcorr(s,c);
subplot(7,1,4)
plot(asc);
title('auto correlation of s and c');xlabel('t');ylabel('amplitude');
anc=xcorr(n,c);
subplot(7,1,5)
plot(anc);
title('auto correlation of n and c');xlabel('t');ylabel('amplitude');
cfc=xcorr(f,c);
subplot(7,1,6)
plot(cfc);
title('auto correlation of f and c');xlabel('t');ylabel('amplitude');
hh=asc+anc;
subplot(7,1,7)
plot(hh);
title('addition of asc+anc');xlabel('t');ylabel('amplitude');

68

CONCLUSION: in this experiment the removal of noise by auto

correlation/cross correlation have been verified using MATLAB

69

EXP.No:16

EXTRACTION OF PERIODIC SIGNAL MASKED BY NOISE USING

CORRELATION
Extraction Of Periodic Signal Masked By Noise Using Correlation

clear all;
close all;
clc;
n=256;
k1=0:n-1;
x=cos(32*pi*k1/n)+sin(48*pi*k1/n);
plot(k1,x)
%Module to find period of input signl
k=2;
xm=zeros(k,1);
ym=zeros(k,1);
hold on
for i=1:k
[xm(i) ym(i)]=ginput(1);
plot(xm(i), ym(i),'r*');
end
period=abs(xm(2)-xm(1));
rounded_p=round(period);

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m=rounded_p
% Adding noise and plotting noisy signal
y=x+randn(1,n);
figure
plot(k1,y)
% To generate impulse train with the period as that of input signal
d=zeros(1,n);
for i=1:n
if (rem(i-1,m)==0)
d(i)=1;
end
end
%Correlating noisy signal and impulse train
cir=cxcorr1(y,d);

%plotting the original and reconstructed signal

m1=0:n/4;
figure
plot(m1,x(m1+1),'r',m1,m*cir(m1+1));

71

72

Application
The theorem is useful for analyzing linear time-invariant systems, LTI systems, when the
inputs and outputs are not square integrable, so their Fourier transforms do not exist. A
corollary is that the Fourier transform of the autocorrelation function of the output of an
LTI system is equal to the product of the Fourier transform of the autocorrelation function
of the input of the system times the squared magnitude of the Fourier transform of the
system impulse response. This works even when the Fourier transforms of the input and
output signals do not exist because these signals are not square integrable, so the system
inputs and outputs cannot be directly related by the Fourier transform of the impulse
response. Since the Fourier transform of the autocorrelation function of a signal is the
power spectrum of the signal, this corollary is equivalent to saying that the power
spectrum of the output is equal to the power spectrum of the input times the power
transfer function.
This corollary is used in the parametric method for power spectrum estimation.

CONCLUSION: In this experiment the Weiner-Khinchine Relation have been

verified using MATLAB

73

EXP.No:17

VERIFICATION OF WIENERKHINCHIN RELATION

AIM: verification of wienerkhinchin relation
EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software
The WienerKhinchin theorem (also known as the WienerKhintchine theorem and
sometimes as the WienerKhinchinEinstein theorem or the KhinchinKolmogorov
theorem) states that the power spectral density of a wide-sense-stationary random
process is the Fourier transform of the corresponding autocorrelation function.[1][2][3]
Continuous case:

Where

is the autocorrelation function defined in terms of statistical expectation, and where is the
power spectral density of the function . Note that the autocorrelation function is defined
in terms of the expected value of a product, and that the Fourier transform of does not
exist in general, because stationary random functions are not square integrable.
The asterisk denotes complex conjugate, and can be omitted if the random process is realvalued.
Discrete case:

Where

74

and where is the power spectral density of the function with discrete values . Being a
sampled and discrete-time sequence, the spectral density is periodic in the frequency
domain.
PROGRAM:
clc
clear all;
t=0:0.1:2*pi;
x=sin(2*t);
subplot(3,2,1);
plot(x);
au=xcorr(x,x);
subplot(3,2,2);
plot(au);
v=fft(au);
subplot(3,2,3);
plot(abs(v));
fw=fft(x);
subplot(3,2,4);
plot(fw);
fw2=(abs(fw)).^2;
subplot(3,2,5);
plot(fw2);

75

76

EXP18.
CHECKING A RANDOM PROCESS FOR STATIONARITY IN WIDE
SENSE.
AIM:Checking a random process for stationarity in wide sense.
EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software

Theory:
a stationary process (or strict(ly) stationary process or strong(ly) stationary process)
is a stochastic process whose joint probability distribution does not change when shifted in
time or space. As a result, parameters such as the mean and variance, if they exist, also do
not change over time or position..

Definition
Formally, let Xt be a stochastic process and let
represent the
cumulative distribution function of the joint distribution of Xt at times t1..tk. Then, Xt
is said to be stationary if, for all k, for all , and for all t1..tk

Weak or wide-sense stationarity

A weaker form of stationarity commonly employed in signal processing is known as
weak-sense stationarity, wide-sense stationarity (WSS) or covariance stationarity.
WSS random processes only require that 1st and 2nd moments do not vary with respect to
time. Any strictly stationary process which has a mean and a covariance is also WSS.
So, a continuous-time random process x(t) which is WSS has the following restrictions on
its mean function

and autocorrelation function

77

The first property implies that the mean function mx(t) must be constant. The second
property implies that the correlation function depends only on the difference between t1
and t2 and only needs to be indexed by one variable rather than two variables. Thus,
instead of writing,

we usually abbreviate the notation and write

This also implies that the autocovariance depends only on

= t1 t2, since

When processing WSS random signals with linear, time-invariant (LTI) filters, it is
helpful to think of the correlation function as a linear operator. Since it is a circulant
operator (depends only on the difference between the two arguments), its eigenfunctions
are the Fourier complex exponentials. Additionally, since the eigenfunctions of LTI
operators are also complex exponentials, LTI processing of WSS random signals is highly
tractableall computations can be performed in the frequency domain. Thus, the WSS
assumption is widely employed in signal processing algorithms.

Applicatons: Stationarity is used as a tool in time series analysis, where the raw data
are often transformed to become stationary, for example, economic data are often
seasonal and/or dependent on the price level. Processes are described as trend stationary
if they are a linear combination of a stationary process and one or more processes
exhibiting a trend. Transforming these data to leave a stationary data set for analysis is
referred to as de-trending

Stationary and Non Stationary Random Process:

A random X(t) is stationary if its statistical properties are unchanged by a time shift in the
time origin.When the auto-Correlation function Rx(t,t+T) of the random X(t) varies with

78

time difference T and the mean value of the random variable X(t1) is independent of the
choice of t1,then X(t) is said to be stationary in the wide-sense or wide-sense stationary .
So a continous- Time random process X(t) which is WSS has the following properties
1) E[X(t)]=X(t)= X(t+T)
2) The Autocorrelation function is written as a function of T that is
3) RX(t,t+T)=Rx(T)
If the statistical properties like mean value or moments depends on time then the
random process is said to be non-stationary.
When dealing wih two random process X(t) and Y(t), we say that they are jointly
wide-sense stationary if each pocess is stationary in the wide-sense.
Rxy(t,t+T)=E[X(t)Y(t+T)]=Rxy(T).
MATLAB PROGRAM:
clear all
clc
y = randn([1 40])
my=round(mean(y));
z=randn([1 40])
mz=round(mean(z));
vy=round(var(y));
vz=round(var(z));
t = sym('t','real');
h0=3;
x=y.*sin(h0*t)+z.*cos(h0*t);
mx=round(mean(x));
k=2;
xk=y.*sin(h0*(t+k))+z.*cos(h0*(t+k));
x1=sin(h0*t)*sin(h0*(t+k));
x2=cos(h0*t)*cos(h0*(t+k));
c=vy*x1+vz*x1;
%if we solve "c=2*sin(3*t)*sin(3*t+6)" we get c=2cos(6)
%which is a costant does not depent on variable 't'
% so it is wide sence stationary

79

1. Define Signal
2. Define determistic and Random Signal
3. Define Delta Function
4. What is Signal Modeling
5. Define Periodic and a periodic Signal
6. Define Symetric and Anti-Symmetric Signals
7. Define Continuous and Discrete Time Signals
8. What are the Different types of representation of discrete time signals
9. What are the Different types of Operation performed on signals
10. What is System
11. What is Causal Signal
12. What are the Different types of Systems
13. What is Linear System
14. What is Time Invariant System
15. What is Static and Dynamic System
16. What is Even Signal
17. What is Odd Signal
18. Define the Properties of Impulse Signal
19. What is Causality Condition of the Signal
20. What is Condition for System Stability
21. Define Convolution
22. Define Properties of Convolution
23. What is the Sufficient condition for the existence of F.T
24. Define the F.T of a signal
25. State Paesevals energy theorem for a periodic signal
26. Define sampling Theorem
27. What is Aliasing Effect
28. what is Under sampling
29. What is Over sampling
30. Define Correlation
31. Define Auto-Correlation
32. Define Cross-Correlation
33. Define Convolution
34. Define Properties of Convolution
35. What is the Difference Between Convolution& Correlation
36. What are Dirchlet Condition
37. Define Fourier Series
38. What is Half Wave Symmetry
39. What are the properties of Continuous-Time Fourier Series
40. Define Laplace-Transform
41. What is the Condition for Convergence of the L.T
42. What is the Region of Convergence(ROC)

80

43. State the Shifting property of L.T

44. State convolution Property of L.T
45. Define Transfer Function
46. Define Pole-Zeros of the Transfer Function
47. What is the Relationship between L.T & F.T &Z.T
48. Fined the Z.T of a Impulse and step
49. What are the Different Methods of evaluating inverse z-T
50. Explain Time-Shifting property of a Z.T
51. what are the ROC properties of a Z.T
52. Define Initial Value Theorem of a Z.T
53. Define Final Value Theorem of a Z.T
54. Define Sampling Theorem
55. Define Nyquist Rate
56. Define Energy of a Signal
57. Define Power of a signal
58. Define Gibbs Phenomena
59. Define the condition for distortionless transmission through the system
60. What is signal band width
61. What is system band width
62. What is Paley-Winer criterion?
63. Derive relationship between rise time and band width
64.

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