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CONTENTS

S.No Experiment Name Page No.

LIST OF EXPERIMENTS:

1. Basic operations on matrices. 1
2. Generation on various signals and Sequences 6
(periodic and aperiodic), such as unit impulse, unit step,
square, sawtooth, triangular, sinusoidal, ramp, sinc.
3. Operations on signals and sequences such as addition, 18
multiplication, scaling, shifting, folding, computation of
energy and average power.
4. Finding the even and odd parts of signal/sequence 22
and real and imaginary part of signal.
5. Convolution between signals and sequences. 26
6. Auto correlation and cross correlation between 29
signals and sequences.
7. Verification of linearity and time invariance 33
properties of a given continuous /discrete system.
8. Computation of unit sample, unit step and sinusoidal 40
response of the given LTI system and verifying its
physical Realizability and stability properties.
9. Gibbs phenomenon. 43
10. Finding the Fourier transform of a given
signal and plotting its magnitude and phase spectrum.
11. Waveform synthesis using Laplace Transform. 48
12. Locating the zeros and poles and plotting the
pole zero maps in s-plane and z-plane for the given 53
transfer function.
13. Generation of Gaussian Noise(real and complex), 54
computation of its mean, M.S. Value and its skew,
kurtosis, and PSD, probability distribution function.
14. Sampling theorem verification. 57
15. Removal of noise by auto correlation/cross correlation.
16. Extraction of periodic signal masked by noise 66
using correlation.
17. Verification of Weiner-Khinchine relations. 70
18. Checking a random process for stationarity in wide sense. 73



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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 =
- 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 =
1 2 3
4 5 6
7 8 9
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.



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- 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 =
1 2 3
4 5 6
7 8 9

Here we extract column 2 of the matrix and make a column vector:
» col_two=matrix( : , 2)
col_two =
2 5 8
- 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 =
1 2 3
4 5 6
7 8 9
- 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 =
1 2 3
4 5 6
» c= b+a % Add a to each element of b
c =
4 5 6
7 8 9
- Scalar - Matrix Subtraction
» a=3;
» b=[1, 2, 3;4, 5, 6]
b =
1 2 3
4 5 6
» c = b - a %Subtract a from each element of b
c =
-2 -1 0
1 2 3
- Scalar - Matrix Multiplication
» a=3;
» b=[1, 2, 3; 4, 5, 6]
b =
1 2 3
4 5 6



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» c = a * b % Multiply each element of b by a
c =
3 6 9
12 15 18
- Scalar - Matrix Division
» a=3;
» b=[1, 2, 3; 4, 5, 6]
b =
1 2 3
4 5 6
» 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 2 3 4 6 4 3 4 5

b = a + 2

b =

3 4 5 6 8 6 5 6 7

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

A =

1 2 0

2 5 -1

4 10 -1

B = A'

B =

1 2 4

2 5 10

0 -1 -1

C = A * B

C =



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5 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



4



0
4 25 -10
0 -10 1

Let's find the inverse of a matrix …


X = inv(A)

X =

5 2 -2

-2 -1 1

0 -2 1

and then illustrate the fact that a matrix times its inverse is the identity matrix.


I = inv(A) * A

I =

1 0 0
0 1 0
0 0 1

to obtain eigenvalues

eig(A)

ans =
3.7321



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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.



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´




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:

And
o (t) =
¦ 0

¹
·

·
t = 0
t = 0
}
÷·
o (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: δ(n)={ 1, n=0
0, n≠0
i.e



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+
u(t)|(t)dt =
+
|(t)dt
0


t > 0
t < 0





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):




¦
1
u (t ) =
´

¹
0




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

1,

n ≥ 0

0, n < 0









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



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




















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



















Peak amplitude, V
p
, and peak-to-peak amplitude, V
pp
, are measured as you might
expect. However, the rms amplitude, V
rms
, 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
V
rms
is equal to V
p
. (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 µs = 10
-6
s), as indicated in the graph



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% 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



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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');



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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 (4n−1)th harmonic by −1 (or
changing its phase by π), 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:



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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]),



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%%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



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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.
- ω, the angular frequency, specifies how many oscillations occur in a unit time
interval, in radians per second
- υ, the phase, specifies where in its cycle the oscillation begins at t = 0.

A sampled sinusoid may be written as:



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where f is the signal frequency, fs is the sampling frequency, θ 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;



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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 2π 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



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% 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



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0 t=0

-1 t<0

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





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:



% program for time shifting of a signal
clc
clear all
close all
t = [0:.01:pi]; %
independent (time) variable A = 8; %
amplitude
f1 = 2;
s1 = A*sin(2*pi*f1*t);
subplot(3,1,1)
plot(t,s1)
xlabel('t');
ylabel('amplitude');
title('input signal')
subplot(3,1,2)
plot(t+10,s1)
xlabel('t');
ylabel('amplitude');
title('right shifted signal')
subplot(3,1,3)



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plot(t-10,s1) xlabel('t');
ylabel('amplitude');
title('left shifted signal')
















































% program for time shifting of a sequence
clc
clear all
close all
n=0:1:4;
h=[1,1,1,-1,2];
subplot(3,1,1)



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stem(n,h);
xlabel('n'); ylabel('h(n)'); title('impulse sequence');
subplot(3,1,2)
stem(n+2,h);
xlabel('n') ; ylabel('h(n)'); title('folded sequence');
subplot(3,1,3)
stem(n-2,h)
xlabel('n') ; ylabel('h(n)'); title('shifted sequence');
subplot(3,1,3)



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









Multiplication :
Multiplication of two signals can be obtained by multiplying their values at
every instants . z (t) = x (t) y (t)














%Program for addition and multiplication of signals

t = [0:.01:1]; % independent (time) variable
A = 8; % amplitude
f1 = 2;
s1 = A*sin(2*pi*f1*t); % create a 2 Hz sine wave
f2 = 6;
s2 = A*sin(2*pi*f2*t); % create a 4 Hz sine wave
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



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



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%Program for addition and multiplication & addition of sequences



n1=1:1:9
s1 = [1 2 3 4 5 6 2 3 1];
subplot(4,1,1)
stem(n, s1)
xlabel('n1')



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title('input sequence')
ylabel('Amplitude')
n2=-2:1:6;
s2=[1 1 1 0 2 3 1 1 0 ]
subplot(4,1,2) ;stem(n2, s2)
title('secong sequence') ;
ylabel('Amplitude')
xlabel('n2')
s3=s1+s2;
subplot(4,1,3) ;stem(n2,s3)
title('Summed squence')
ylabel('Amplitude')
xlabel('n2') s4=s1.*s2;
subplot(4,1,4);stem(n2,s4);
title('multiplication');
ylabel('Amplitude')
xlabel('n2')



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Time reversal:
Time reversal of a signal x(t) can be obtained by folding the signal about
t=0. Y(t)=y(-t)



% Folding for signal

Clc
Close all
Clear all
t = [0:.01:1]; % independent
(time) variable A = 8; % amplitude
f1 = 2; % create a 2 Hz sine wave lasting 1 sec
x = A*sin(2*pi*f1*t);
lx=length(x);
nx=0:lx-1;
xf=fliplr(x); nf=-
fliplr(nx);
subplot(2,1,1);
stem(nx,x);
xlabel('nx');
ylabel('amplitude)');
title('original signal');
subplot(2,1,2);
stem(nf,xf);
xlabel('nf');
ylabel('amplitude');
title('folded signal');



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%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)');



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title('folded signal');













































Signal Amplification/Attuation : Y(n)=ax(n) if a < 1 attnuation
a >1 amplification






% Amplitude scalling



t = [0:.01:1]; % independent
(time) variable A = 8; % amplitude
f1 = 2; % create a 2 Hz sine wave lasting 1 sec



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s1 = A*sin(2*pi*f1*t);
subplot(3,1,1)
plot(s1);
xlabel('t');
ylabel('amplitude');
title('input signal');
s2=2*s1;
subplot(3,1,2)
plot(s2);
xlabel('t');
ylabel('amplitude');
title('amplified input signal');
s3=s1/2;
subplot(3,1,3)
plot(s3);
xlabel('t');
ylabel('amplitude');
title('atteniatedinput signal');



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% Amplitude scalling for sequences
n=0:1:6
s1 = [1 2 3 3 1 1 1]
subplot(3,1,1)
stem(n,s1);
xlabel('n');
ylabel('amplitude');
title('input signal');
s2=4*s1;
subplot(3,1,2)
stem(n,s2);
xlabel('t');
ylabel('amplitude');
title('amplified input signal');
s3=s1/4;
subplot(3,1,3)
stem(n,s3);



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xlabel('t');
ylabel('amplitude');
title('atteniatedinput signal');







































Time scalling:

The Time scalling of a signal x(t) can be accomplished by replacing t by at
where t is a scalling factor.
Y(t)=x(at) : a= arbotorary constant

If a<1 (a=1/2) i e x(t/2)---- the signal y(t) is expanded by 2
If a>1 (a=2) i e x(2 t )---- the signal y(t) is compressed 2





% Time scalling

t = [0:.05:pi]; % independent (time) variable



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A = 8; % amplitude
f1 = 2; % create a 2 Hz sine wave lasting 1 sec
s1 = A*sin(2*pi*f1*t);
subplot(3,1,1)
plot(s1); xlabel('t');
ylabel('amplitude');
title ('sine signal')
s2=A*sin(2*2*pi*f1*t); % scalling by a=4
subplot(3,1,2)
plot(s2); xlabel('t');
ylabel('amplitude');
title ('scaled sine signal with a=4')
s3=A*sin(.5*pi*f1*t); % scalling by a=1/4
subplot(3,1,3);
plot(s3);
xlabel('t');
ylabel('amplitude');
title ('scaled sine signal with a=1/4')



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CONCLUSION: Inthis experiment the various oprations on signals have been
performedUsing MATLAB have been demonstrated




Excersize questions: Sketch the following questions using MATLAB



1. x(t)= u(-t+1)
2. x(t)=3r(t-1)
3. x(t)=U(n+2-u(n-3)
4. x(n)=x
1
(n)+x
2
(n)where x
1
(n)={1,3,2,1},x
2
(n)={1,-2,3,2}
5. x(t)=r(t)-2r(t-1)+r(t-2)
6. x(n)=2δ(n+2)-2δ(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)



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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, x
e
(t), and its odd part, x
o
(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.

close all; clear
all;
t=0:.005:4*pi;
x=sin(t)+cos(t); % x(t)=sint(t)+cos(t)
subplot(2,2,1)
plot(t,x)
xlabel('t');
ylabel('amplitude')
title('input signal')
y=sin(-t)+cos(-t) % y=x(-t)



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subplot(2,2,2)
plot(t,y)
xlabel('t');
ylabel('amplitude')
title('input signal with t=-t')
z=x+y
subplot(2,2,3)
plot(t,z/2)
xlabel('t');
ylabel('amplitude')
title('even part of the signal')
p=x-y
subplot(2,2,4)
plot(t,p/2)
xlabel('t');
ylabel('amplitude')
title('odd part of the signal')



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%even and odd signals program:

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');



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ENERGY AND POWER SIGNA L:A signal can be categorized into energy signal
or power signal:

energy signal has a finite energy, 0 < E < ∞. And Power=0

Energy signals have values only in the limited time duration.

Example: a signal having only one square pulse is energy signal.
A signal that decays exponentially has finite energy

Energy of Contineous Signals






Energy of Discrete Signals



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power signal has a finite energy, 0 < P < ∞. And Energy =∞
Example: sine
Power of Contonous Signals





Power of Discrete Signals






% energy
clc;
close all;
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



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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]);
disp('the theoretical power of the signal is');
P_theory=3/8



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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:












If x(n)=h(n) [ impulse ) then output y(n) is known as impulse response of the system.

x(n)=δ(n)

y(n)=T[x(n)]=h(n) similarly δ (n-k)= h(n-k)
x(n) cab represented as weighted sum of impulses such as




y(n)=T[x(n)]


=
δ (n-k)= h(n-k)


--- Linear Convolution equation

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



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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')
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



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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]



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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

















































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).



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MATLAB Software

Theory:
Correlations of sequences:
It is a measure of the degree to which two sequences are similar. Given two real-
valued 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

r
x, y
(l) =

¿
x(n) y(n ÷ l)
n=÷·
The index l is called the shift or lag parameter

Autocorrelation

r
x, x
(l) =



¿
x(n)x(n ÷ l)
n=÷·

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);
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);



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subplot(3,1,3);
stem(y);
xlabel('n');
ylabel('y(n)');
disp('the resultant signal is');
disp(y);
title('correlation signal');











































% 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');



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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');






































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



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x1(t)=e
-3t
{u(t)-u(t-2)} , x2(t)= e
-3t
for 0 ≤ t ≤ 2



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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 :
Any system is said to be linear if it satisfies the superposition principal
superposition principal state that Response to a weigted sumn of input
signal equal to the corresponding weighted sum of the outputs of the
system to each of the individual input signals





X(n)-----------input signal
Y(n) --------- output signal

Y(n)=T[x(n)]

Y1(n)=T[X1(n)] : Y2(n)=T[X2(n)]
x3=[a X1(n)] +b [X2(n) ]
Y3(n)= T [x3(n)]

= T [a X1(n)] +b [X2(n) ] = a Y1(n)+ b [X2(n) ]= Z 3(n)
Let a [Y1(n)]+ b [X2(n) ] =Z 3(n)
If Y3(n )- Z 3(n)=0 then the system is stable other wise it is nit stable



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Program 1:
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;
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:



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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;
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



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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);
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)



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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):

A system is called time invariant if its input – output characteristics do not
change with time

X(t)---- input : Y(t) ---output
X(t-T) -----delay input by T seconds : Y(t-T) ------ Delayed output by T seconds




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];
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;



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Program2:
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];
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



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subplot(3,1,1),stem(y),grid;
subplot(3,1,2),stem(yd),grid;
subplot(3,1,3),stem(d),grid;













































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



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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) = δ(n) then the output of the system is
known as impulse response denoted by h(n) where,
h(n) = T[δ(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);



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%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);



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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



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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



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CONCLUSION: In this experiment Gibbs phenomenon have been demonstrated
Using MATLAB



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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:

the Fourier transform as follows. Suppose that ƒ is a function which is zero outside of
some interval [−L/2, L/2]. Then for any T ≥ L we may expand ƒ in a Fourier series on the
interval [−T/2,T/2], where the "amount" of the wave e
2πinx/T
in the Fourier series of ƒ is
given by

By definition Fourier Transform of signal f(t) is defined as






Iverse Fourier Transform of signal F(w) is defined as



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PROGRAM:

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
¿
K = 0

x |n |e

÷ j 2 [ Kn
N



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');



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xlabel('time index n----->');
ylabel('amplitude x[n]----> ');
subplot(2,1,2);
stem(n,y);
title('output sequence');
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);



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magX=abs(X),phase=angle(X)*180/pi;
subplot(2,1,1)
plot(magX);
grid
xlabel('k')
ylabel('X(K)')
subplot(2,1,2)
plot(phase);
grid
xlabel('k')
ylabel('degrees')












































Applications:
The no of multiplications in DFT = N
2.




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The no of Additions in DFT = N(N-1).

For FFT.

The no of multiplication = N/2 log
2
N.
The no of additions = N log
2
N.

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)



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Program for nverse Laplace Transform

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



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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 s-
domain (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



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EXP.NO: 12

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)



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The ROC of the Z-transform depends on the convergence of the














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



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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')



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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 = 2πfm).



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Figure 2: (a) Original signal g(t) (b) Spectrum G(w)
δ (t) is the sampling signal with fs = 1/T > 2fm.









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



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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



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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;



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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



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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 for all T
T--∞ -T/2

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

Detection of noise by Auto-Correlation:

S(t)=Periodic Signal (Transmitted) , mixed with a noise signal n(t).
Then f(t) is received signal is [s(t ) + n(t) ]
Let Q
ff
(T) =Auto Correlation Function of f(t)
Q
ss
(t) = Auto Correlation Function of S(t)
Q
nn
(T) = Auto Correlation Function of n(t)

T/2
Q
ff
(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
=Q
ss
(T)+Q
nn
(T)+Q
sn
(T)+Q
ns
(T)

The periodic signal s(t) and noise signal n(t) are uncorrelated
Q
sn
(t)=Q
ns
(t)=0 ;
Then Q
ff
(t)=Q
ss
(t)+Q
nn
(t)



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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 Q
ss
(T) is a periodic where
as aQ
nn
(T) becomes small for large values of T Therefore for sufficiently large values of
T Q
ff
(T) is equal to Q
ss
(T).

Detection by Cross Correlation:

f(t)=s(t)+n(t)
c(t)=Locally generated signal with same frequencyas that of S(t)
T/2
Q
fc
(t) = Lim 1/T ∫ [s(t)+n(t)] [ c(t-T)] dt
T--∞ -T/2

= Q
sc
(T)+Q
nc
(T)
C(t) is periodic function and uncorrelated with the random noise signal n(t). Hence
Q
nc
(T0=0) Therefore Q
fc
(T)=Q
sc
(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);



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subplot(6,1,4)
plot(an);
title('auto correlation of n');
xlabel('t');
ylabel('amplitude');
cff=xcorr(f,f);
subplot(6,1,5)
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');



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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);



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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(' correlation of s and c');xlabel('t');ylabel('amplitude');
anc=xcorr(n,c);
subplot(7,1,5)
plot(anc);
title(' correlation of n and c');xlabel('t');ylabel('amplitude');
cfc=xcorr(f,c);
subplot(7,1,6)
plot(cfc);
title(' correlation of f and c');xlabel('t');ylabel('amplitude');
hh=asc+anc;
subplot(7,1,7)
plot(hh);
title('addition of sc+nc');xlabel('t');ylabel('amplitude');



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Applications:detection of radar and sonal signals ,detection of cyclical
component in brain analysis meterology etc.

CONCLUSION: in this experiment the removal of noise by auto
correlation/cross correlation have been verified using MATLAB



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EXP.No:16

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);
m=rounded_p
% Adding noise and plotting noisy signal



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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));



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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



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EXP.No:17

VERIFICATION OF WIENER–KHINCHIN RELATION

AIM: verification of wiener–khinchin relation

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



The Wiener–Khinchin theorem (also known as the Wiener–Khintchine theorem and
sometimes as the Wiener–Khinchin–Einstein theorem or the Khinchin–Kolmogorov
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 real-
valued.

Discrete case:







Where



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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);



DEPT.OF ECE
JNTUWORLD - 90 -

www.jntuworld.com
MADHIRA INSTITUTE OF TECHNOLOGY & SCIENCES,KODAD









DEPT.OF ECE
JNTUWORLD - 91 -

www.jntuworld.com
MADHIRA INSTITUTE OF TECHNOLOGY & SCIENCES,KODAD













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 X
t
be a stochastic process and let represent the
cumulative distribution function of the joint distribution of X
t
at times t
1
…..t
k
. Then, X
t

is said to be stationary if, for all k, for all τ, and for all t
1
…..t
k






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



DEPT.OF ECE
JNTUWORLD - 92 -

www.jntuworld.com
MADHIRA INSTITUTE OF TECHNOLOGY & SCIENCES,KODAD


















The first property implies that the mean function m
x
(t) must be constant. The second
property implies that the correlation function depends only on the difference between t
1
and t
2
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 τ = t
1
− t
2
, 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 tractable—all
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:



DEPT.OF ECE
JNTUWORLD - 93 -

www.jntuworld.com
MADHIRA INSTITUTE OF TECHNOLOGY & SCIENCES,KODAD







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
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)]=µX(t)= µX(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



DEPT.OF ECE
JNTUWORLD - 94 -

www.jntuworld.com
MADHIRA INSTITUTE OF TECHNOLOGY & SCIENCES,KODAD


















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 Paeseval’s 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



DEPT.OF ECE
JNTUWORLD - 95 -

www.jntuworld.com
MADHIRA INSTITUTE OF TECHNOLOGY & SCIENCES,KODAD







41. What is the Condition for Convergence of the L.T
42. What is the Region of Convergence(ROC)
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.



DEPT.OF ECE
JNTUWORLD - 96 -

www.jntuworld.com
MADHIRA INSTITUTE OF TECHNOLOGY & SCIENCES,KODAD



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