Basic Simulation Lab

Experiment No-1

BASIC OPERATIONS ON MATRICES

AIM: Generate a matrix and perform basic operation on matrices using MATLAB software.
Software Required: MATLAB software
Theory:
MATLAB treats all variables as matrices. Vectors are special forms of matrices and contain
only one row or one column. Whereas scalars are special forms of matrices and contain only
one row and one column. A matrix with one row is called row vector and a matrix with
single column is called column vector.
The first one consists of convenient matrix building functions, some of which are
given below.
1. eye - identity matrix
2. zeros - matrix of zeros
3. ones - matrix of ones
4. diag - extract diagonal of a matrix or create diagonal matrices
5. triu - upper triangular part of a matrix
6. tril - lower triangular part of a matrix
7. rand - ran
commands in the second sub-category of matrix functions are
1. size- size of a matrix
2. det -determinant of a square matrix
3. inv- inverse of a matrix
4. rank- rank of a matrix
5. rref- reduced row echelon form
6. eig- eigen values and eigenvectors
7. poly- characteristic polynomialdomly generated matrix
Program:
%Creating a column vector
>> a=[1;2;3]
a=
1
2
3

% Creating a row vector

>> b=[1 2 3]
b=
123

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% Creating a matrix
>>m=[1 2 3;4 6 9;2 6 9]
m=
123
469
269
% Extracting sub matrix from matrix
>> sub_m=m(2:3,2:3)
sub_m =
69
69
% Extracting column vector from matrix
>> c=m (:,2)
c=
2
6
6
% Extracting row vector from matrix
>> d=m (3,:)
d=
269
% creation of two matrices a and b
>> a=[2 4 -1;-2 1 9;-1 -1 0]
a=
2 4 -1
-2 1 9
-1 -1 0
>> b=[0 2 3;1 0 2;1 4 6]
b=
023
102
146

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% matrix multiplication
>> x1=a*b
x1 =
3 0 8
10 32 50
-1 -2 -5
% element to element multiplication
>> x2=a.*b
x2 =
0 8 -3
-2 0 18
-1 -4 0
% matrix addition
>> x3=a+b
x3 =
262
-1 1 11
036
% matrix subtraction
>> x4=a-b
x4 =
2 2 -4
-3 1 7
-2 -5 -6
% matrix division
>> x5=a/b
x5 =
-9.0000 -3.5000 5.5000
12.0000 3.7500 -5.7500
3.0000 0.7500 -1.7500
% element to element division
>> x6=a./b
Warning: Divide by zero.

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x6 =
Inf 2.0000 -0.3333
-2.0000 Inf 4.5000
-1.0000 -0.2500 0
% inverse of matrix a
>> x7=inv(a)
x7 =
-0.4286 -0.0476 -1.7619
0.4286 0.0476 0.7619 -
0.1429 0.0952 -0.4762
% transpose of matrix a
>> x8=a'
x8 =
2 -2 -1
4 1 -1
-1 9 0
%cube of matrix a( power of the matrix)
>> x9=a^3

x9 =
-66 -33 120
-9 -87 -129
18 3 -45

%vector power 3 of matrix a

>> x10=a.^3

x10 =

8 64 -1
-8 1 729
-1 -1 0

RESULT: Matrix operations are performed using Matlab software.

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Experiment No-2
Generation of signals and sequences

AIM: Generate various signals and sequences (Periodic and aperiodic), such as Unit Impulse,
Unit Step, Square, Saw tooth, Triangular, Sinusoidal, Ramp, Sinc.
Software Required: Matlab software
Theory: If the amplitude of the signal is defined at every instant of time then it is called
continuous time signal. If the amplitude of the signal is defined at only at some instants of
time then it is called discrete time signal. If the signal repeats itself at regular intervals then it
is called periodic signal. Otherwise they are called aperiodic signals.
EX: ramp,Impulse,unit step, sinc- Aperiodic signals
square,sawtooth,triangular sinusoidal – periodic signals.

Ramp sinal: The ramp function is a unitary real function, easily computable as the mean of
the independent variable and its absolute value.This function is applied in engineering. The
name ramp function is derived from the appearance of its graph.
t when

r(t)=
0 else

Unit impulse signal: One of the more useful functions in the study of linear systems is the
"unit impulse function."An ideal impulse function is a function that is zero everywhere but
at the origin, where it isinfinitely high. However, the area of the impulse is finite
Y(t)= 1 when t=0
=0 other wise
Unit step signal: The unit step function and the impulse function are considered to be
fundamental functions in engineering, and it is strongly recommended that the reader
becomes very familiar with both of these functions.
0 if t < 0
u(t)= 1 if t > 0
If t=0

Sinc signal: There is a particular form that appears so frequently in communications

engineering, that we give it its own name. This function is called the "Sinc function”.
The Sinc function is defined in the following manner:
if and sinc(0) =1
The value of sinc(x) is defined as 1 at x = 0, since

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PROCEDURE:-

Open MATLAB


Open new M-file


Type the program


Save in current directory


Compile and Run the program



For the output see command window\ Figure window

PROGRAM:

%Generation of signals and sequences %

clc;
clear all;
close all;
%~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
%generation of unit impulse signal
t1=-1:0.01:1;
y1=(t1==0);
Subplot(2,2,1);
plot(t1,y1);
xlabel('time');
ylabel('amplitude');
title('unit impulse signal');
%generation of impulse sequence
subplot(2,2,2);
stem(t1,y1);
xlabel('n');
ylabel('amplitude');
title('unit impulse sequence');
%~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

%generation of unit step signal

t2=-10:1:10;
y2=(t2>=0);
subplot(2,2,3);
plot(t2,y2);
xlabel('time');
ylabel('amplitude');
title('unit step signal');
%generation of unit step sequence
subplot(2,2,4);
stem(t2,y2);
xlabel('n');
ylabel('amplitude');
title('unit step sequence');
%~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

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%generation of square wave signal

t=0:0.002:0.1;
y3=square(2*pi*50*t)
figure;
subplot(2,2,1);
plot(t,y3);
axis([0 0.1 -2 2]);
xlabel('time');
ylabel('amplitude');
title('square wave signal');
%generation of square wave sequence
subplot(2,2,2);
stem(t,y3);
axis([0 0.1 -2 2]);
xlabel('n');
ylabel('amplitude');
title('square wave sequence');

%~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

%generation of sawtooth signal

y4=sawtooth(2*pi*50*t);
subplot(2,2,3);
plot(t,y4);
axis([0 0.1 -2 2]);
xlabel('time');
ylabel('amplitude');
title('sawtooth wave signal');
%generation of sawtooth sequence
subplot(2,2,4);
stem(t,y4);
axis([0 0.1 -2 2]);
xlabel('n');
ylabel('amplitude');
title('sawtooth wave sequence');
%~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

%generation of triangular wave

signal y5=sawtooth(2*pi*50*t,.5);
figure;
subplot(2,2,1);
plot(t,y5);
axis([0 0.1 -2 2]);
xlabel('time');
ylabel('amplitude');
title(' triangular wave signal');
%generation of triangular wave sequence
subplot(2,2,2);
stem(t,y5);
axis([0 0.1 -2 2]);
xlabel('n');
ylabel('amplitude');
title('triangular wave sequence');
%~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

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%generation of sinsoidal wave signal

y6=sin(2*pi*40*t);
subplot(2,2,3);
plot(t,y6);
axis([0 0.1 -2 2]);
xlabel('time');
ylabel('amplitude');
title(' sinsoidal wave signal');
%generation of sin wave sequence
subplot(2,2,4);
stem(t,y6);
axis([0 0.1 -2 2]);
xlabel('n');
ylabel('amplitude');
title('sin wave sequence');
%~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

%generation of ramp signal

y7=t;
figure;
subplot(2,2,1);
plot(t,y7);
xlabel('time');
ylabel('amplitude');
title('ramp signal');
%generation of ramp sequence
subplot(2,2,2);
stem(t,y7);
xlabel('n');
ylabel('amplitude');
title('ramp sequence');
%~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

%generation of sinc signal

t3=linspace(-5,5);
y8=sinc(t3);
subplot(2,2,3);
plot(t3,y8);
xlabel('time');
ylabel('amplitude');
title(' sinc signal');
%generation of sinc sequence
subplot(2,2,4);
stem(y8);
xlabel('n');
ylabel('amplitude');
title('sinc sequence');

Result: Various signals & sequences generated using Matlab software.

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output:

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Experiment No-3
Basic Operations on Signals and sequences
AIM: perform the operations on signals and sequences such as addition, multiplication,
scaling, shifting, folding and also compute energy and power.
Software Required: Matlab software.

Theory:
Signal Addition
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)

Time reversal/Folding:

Time reversal of a signal x(t) can be obtained by folding the signal about t=0.
Y(t)=y(-t)
Signal Amplification/Scaling : Y(n)=ax(n) if a < 1 attnuation
a >1 amplification

Time shifting: The time shifting of x(n) obtained by delay or advance the signal in time by
using y(n)=x(n+k)
If k is a positive number, y(n) shifted to the right i e the shifting delays the signal
If k is a negative number, y(n ) it gets shifted left. Signal Shifting advances the signal

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Energy:
E=sum(abs(x).^2)

Average power:
P=(sum(abs(x).^2))/length(x)

Program:
clc;
clear all;
close all;
%generating two input signals%
t=0:.01:1;
x1=sin(2*pi*4*t);
x2=sin(2*pi*8*t);
subplot(2,2,1);
plot(t,x1);
xlabel('time');
ylabel('amplitude');
title('input signal 1');
subplot(2,2,2);
plot(t,x2);
xlabel('time');
ylabel('amplitude');
title('input signal 2');

%addition of signals%
y1=x1+x2;
subplot(2,2,3);
plot(t,y1);
xlabel('time');
ylabel('amplitude');
title('addition of two signals');

%multiplication of signals%
y2=x1.*x2;
subplot(2,2,4);
plot(t,y2);
xlabel('time');
ylabel('amplitude');
title('multiplication of two signals');

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%scaling of a signal1
A=2;
y3=A*x1;
figure;
subplot(2,2,1);
plot(t,x1);
xlabel('time');
ylabel('amplitude');
title('input signal')
subplot(2,2,2);
plot(t,y3);
xlabel('time');
ylabel('amplitude');
title('amplified input signal');

% folding of a signal1
h=length(x1); nx=0:h-1;
subplot(2,2,3);
plot(nx,x1);
xlabel('nx');
ylabel('amplitude');
title('input signal')
y4=fliplr(x1);
nf=-fliplr(nx);
subplot(2,2,4);
plot(nf,y4);
xlabel('nf');
ylabel('amplitude');
title('folded signal');

%shifting of a signal 1
figure;
subplot(3,1,1);
plot(t,x1);
xlabel('time t');
ylabel('amplitude');
title('input signal');
subplot(3,1,2);
plot(t+2,x1);
xlabel('t+2');
ylabel('amplitude');
title('right shifted signal');
subplot(3,1,3);
plot(t-2,x1);
xlabel('t-2');
ylabel('amplitude');
title('left shifted signal');
%~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

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%operations on sequences
n1=1:1:9;
s1=[1 2 3 0 5 8 0 2 4];
figure;
subplot(2,2,1);
stem(n1,s1);
xlabel('n1');
ylabel('amplitude');
title('input sequence1');
s2=[1 1 2 4 6 0 5 3 6];
subplot(2,2,2);
stem(n1,s2);
xlabel('n2');
ylabel('amplitude');
title('input sequence2');

% addition of sequences
s3=s1+s2; subplot(2,2,3);
stem(n1,s3); xlabel('n1');
ylabel('amplitude');
title('sum of two
sequences');

% multiplication of sequences
s4=s1.*s2;
subplot(2,2,4);
stem(n1,s4);
xlabel('n1');
ylabel('amplitude');
title('product of two sequences');
%~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

% program for energy of a sequence

z1=input('enter the input sequence');
e1=sum(abs(z1).^2);
disp('energy of given sequence is');e1

%program for energy of a signal

t=0:pi:10*pi;
z2=cos(2*pi*50*t).^2;
e2=sum(abs(z2).^2);
disp('energy of given signal is');e2

% program for power of a sequence

p1= (sum(abs(z1).^2))/length(z1);
disp('power of given sequence is');p1

% program for power of a signal

p2=(sum(abs(z2).^2))/length(z2);
disp('power of given signal
is');p2

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OUTPUT:
enter the input sequence[1 3 2 4 1]
energy of given sequence is
e1 = 31
energy of given signal is

e2 = 4.0388
power of given sequence is
p1 = 6.2000
power of given signal is
p2 = 0.3672

Result: Various operations on signals and sequences are performed.

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Experiment No-4
Even and odd parts of signal and sequence & Real and imaginary parts

AIM: Finding even and odd part of the signal and sequence and also find real and imaginary
parts of signal.

Software Required: Matlab software

Theory: 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

Even and odd part of a signal: Any signal x(t) can be expressed as sum of even and odd
components I e

X(t)=xe(t)+xo(t)

Program:
%%even add odd parts of signals
clc;
close all;
clear all;
t=0:0.001:10;
x1=sin(t)+cos(t);
subplot(2,2,1);
plot(t,x1);
grid;
xlabel('time');
ylabel('amplitude');
title('first signal');

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x2=sin(-t)+cos(-t);
subplot(2,2,2);
plot(t,x2);
grid;
xlabel('time');
ylabel('amplitude');
title('second signal');
xe=1/2*(x1+x2);
subplot(2,2,3);
plot(t,xe);
grid;
xlabel('time');
ylabel('amplitude');
title('even part of the signal');
xo=1/2*(x1-x2);
subplot(2,2,4);
plot(t,xo);
grid;
xlabel('time');
ylabel('amplitude');
title('odd part of the signal');

%real and imaginary parts of signals and sequences

clc;
clear all;
close all;
%real and imag parts of signals
t=-10:0.01:10;
j=sqrt(-1);
z=cos(t)+j*sin(t);
subplot(2,2,1);
plot(t,real(z));
xlabel('time');
ylabel('amplitude');
title('Real part of complex signal');
subplot(2,2,2);
plot(t,imag(z));
xlabel('time');
ylabel('amplitude');
title('Imag part of complex signal');

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%real and imag parts of sequences

z1=[0,2+1j*4,-3+1j*2,5-1j*1,-2-1j*4,-1j*3,0];
n=-3:3;
subplot(2,2,3);
stem(n,real(z1));
xlabel('n');
ylabel('amplitude');
title('real part of seq ');
subplot(2,2,4);
stem(n,imag(z1));
xlabel('n');
ylabel('amplitude');
title('imaginary part of seq ');

RESULT: Even and odd part of the signal and sequence, real and imaginary parts of
signal are computed.

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Experiment No-5
Convolution between signals& sequences

Aim: Write the program for convolution between two signals and also between two
sequences.

Software Required: 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

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)]

T[δ(n)=h(n) then T[ δ (n-k)] = h(n-k)

Thus output of LTI system is geven by weighted sum of

time-shifted impulse responses ---------------- Linear Convolution equation

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Program:

%Convolution of signals
clc;
close all;
clear all;
t=-10:0.001:10;
x1=sin(t);
subplot(3,1,1);
plot(t,x1);
grid;
xlabel('time');
ylabel('amplitude');
title('input signal');
x2=cos(t);
subplot(3,1,2);
plot(t,x2);
grid;
xlabel('time');
ylabel('amplitude');
title('impulse signal ');
y=conv2(x1,x2);
subplot(3,1,3);
plot(y);
grid;
xlabel('time');
ylabel('amplitude');
title('convolution of signal');
%Convolution of sequences
clc;
close all;
clear all;
x=[2 3 4];
h=[5 6 7];
y=conv2(x,h);
subplot(3,1,1);
stem(x);
grid;
xlabel('n');
ylabel('x(n)');
title('input sequence');

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subplot(3,1,2);
stem(h);
grid;
xlabel('n');
ylabel('h(n)');
title('impulse sequence ');
subplot(3,1,3);
stem(y);
grid;
xlabel('n');
ylabel('y(n)');
title('convolution sequence');
display('the convolution sequence is:');
display(y);

RESULT: convolution between signals and sequences is computed.

Output:

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Experiment No-6
Auto correlation and Cross correlation

Aim: To compute Auto correlation and Cross correlation between signals and sequences.

Software Required: Mat lab 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:

Cross correlation +∞

rx, y (l) = ∑x(n) y(n − l)

n=−∞

The index l is called the shift or lag parameter

Autocorrelation
+∞

rx,x (l) = ∑x(n)x(n − l)

n =−∞

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Program:

%Auto and cross correlation of signals%

clc;
close all;
clear all;
t=-10:0.001:10;
x1=sin(t);
subplot(4,1,1);
plot(t,x1);
grid;
xlabel('time');
ylabel('amplitude');
title('input signal');
x2=cos(t);
subplot(4,1,2);
plot(t,x2);
grid;
xlabel('time');
ylabel('amplitude');
title('impulse signal ');
%auto correlation of signal
y=xcorr(x1,x1);
subplot(4,1,3);
plot(y);
grid;
xlabel('time');
ylabel('amplitude');
title('auto correlation of signal');
%cross correlation of signal
z=xcorr(x1,x2);
subplot(4,1,4);
plot(z);
grid;
xlabel('time');
ylabel('amplitude');
title('cross correlation of signal');

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%Auto and cross correlation of Sequences%

clc;
close all;
clear all;
x=[1 2 5 7];
subplot(4,1,1);
stem(x);
grid;
xlabel('n');
ylabel('x(n)');
title('input sequence');
h=[2 6 0 5 3];
subplot(4,1,2);
stem(h);
grid;
xlabel('n');
ylabel('h(n)');
title('impulse sequence ');
%auto correlation of sequence
y=xcorr(x,x);
subplot(4,1,3);
stem(y);
grid;
xlabel('n');
ylabel('y(n)');
title('auto correlation of sequence');

%cross correlation of sequence

z=xcorr(x,h);
subplot(4,1,4);
stem(z);
grid;
xlabel('n');
ylabel('z(n)');
title('cross correlation of sequence');

Result: Auto correlation and Cross correlation between signals and sequences is computed.
Output: enter input sequence [1 2 5 7]
enter the impulse sequence [2 6 0 5 3]

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Experiment No. -7(a)

VERIFICATION OF LINEARITY OF A DISCRETE SYSTEM
AIM: Verify the Linearity of a given Discrete System.

Software Required:
Mat lab software 7.0 and above
Theory:

LINEARITY PROPERTY:
Any system is said to be linear if it satisfies the superposition principal. superposition
principal state that Response to a weighted sum 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 [Y Let a

[Y1(n)]+ b [X2(n) ] =Z 3(n)

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Program:

clc;
clear all;
close all;
%entering two input sequences and impulse sequence
x1 = input ('type the samples of x1');
x2 = input ('type the samples of x2');
if(length(x1)~=length(x2))
disp('error: Lengths of x1 and x2 are different');
return;
end;
h = input ('type the samples of h');
%length of output sequence
N = length(x1)+length(h)-1;
disp('length of the output signal will be'); disp(N);
%entering scaling factors
a1 = input ('The scale factor a1 is');
a2 = input ('The scale factor a2 is');
x= a1*x1 + a2*x2;
%response of x and x1
y = conv(x,h);
y1 = conv(x1,h);
%scaled response of response of x2
y11 = a1 * y1;
y2 = conv(x2,h);
%scaled response of
y22 = a2*y2;
y33 = y11 + y22;
disp ('Input signal x1 is');
disp(x1);
disp ('Input signal x2 is');
disp(x2);
disp ('Output Sequence y1 is '); disp(y1);
disp ('Output Sequence y33 is '); disp(y33);
if ( y1 == y33 )
disp(' y1 = y33. Hence the LTI system is LINEAR ');
end;

Result: The Linearity of a given Discrete System is verified.

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Output:

Type the samples of x1 [1 5 6 7]

Type the samples of x2 [2 3 4 8]
Type the samples of h [2 6 5 4]
Length of the output signal will be
7

The scale factor a1 is 2

The scale factor a2 is 3
Input signal x1 is
1 5 6 7

Input signal x2 is
2 3 4 8

Output Sequence y is
16 86 202 347 424 286 152

Output Sequence y33 is

16 86 202 347 424 286 152

y= y33. Hence the LTI system is LINEAR

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Experiment No. -7(b)

VERIFICATION OF TIME INVARIANCE OF A DISCRETE
SYSTEM

AIM: Verify the Time Invariance of a given Discrete System.

Software Required:
Mat lab software 7.0 and above
Theory:

TIME INVARIENT SYSTEMS(TI):

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

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

seconds seconds : Y(t-T) ------ Delayed output by T

Program:

clc;
clear all;
close all;
%entering two input sequences%
x = input('Type the samples of
signal x(n)');
h = input('Type the samples of
signal h(n)');
%original response%
y = conv(x,h);
display('Enter a POSITIVE
number for delay');
%disp(or)display
d = input('Desired delay of the
signal is');
%delayed input
xd = [zeros(1,d), x];
nxd = 0 : length(xd)-1;

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%delayed output%
yd = conv(xd,h);
nyd = 0:length(yd)-1;
disp('Original Input Signal x(n) is');disp(x);
disp('Delayed Input Signal xd(n) is');disp(xd);
%disp(or)display
disp('Original Output Signal y(n) is');disp(y);
disp('Delayed Output Signal yd(n) is');disp(yd);
xp = [x ,zeros(1,d)];
subplot(2,1,1);
stem(nxd,xp);
grid;
xlabel( 'Time Index n' );
ylabel( 'x(n)' );
title( 'Original Input Signal x(n)' );
subplot(2,1,2);
stem(nxd,xd);
grid;
xlabel( 'Time Index n');
ylabel( 'xd(n)' );
title( 'Delayed Input Signal xd(n)');
yp = [y ,zeros(1,d)];
figure;
subplot(2,1,1);
stem(nyd,yp);
grid;
xlabel('Time Index n');
ylabel('y(n)');
title('Original Output Signal y(n)');
subplot(2,1,2);
stem(nyd,yd);
grid;
xlabel('Time Index n');
ylabel('yd(n)');
title('Delayed Output Signal yd(n)');

Result: The Time Invariance of a given Discrete System is verified.

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Output:

Type the samples of signal x(n) [2 3 4 6]

Type the samples of signal h(n) [1 2 3 8]
Enter a POSITIVE number for delay
Desired delay of the signal is 5
Original Input Signal x(n) is
2 3 4 6

Delayed Input Signal xd(n) is

0 0 0 0 0 2 3 4 6

Original Output Signal y(n) is

2 7 16 39 48 50 48

Delayed Output Signal yd(n) is

0 0 0 0 0 2 7 16 39 48 50 48

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Experiment No. -8
UNIT SAMPLE, UNIT STEP AND SINUSOIDAL RESPONSE OF THE
GIVEN LTI SYSTEM AND VERIFYING ITS STABILITY

AIM: Compute the Unit sample, unit step and sinusoidal response of the given LTI
system and verifying its stability

Software Required:
Mat lab software 7.0 and above

Theory:
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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Program:

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

clc;
clear all;
close all;
a=[1,-1,0.9];
b=[1];
n =0:3:100;
%generating impulse signal
x1=(n==0);
%impulse response
h1=filter(b,a,x1);
subplot(3,1,1);
stem(n,h1);
xlabel('n');
ylabel('h(n)');
title('impulse response');
%generating step signal
x2=(n>0);
%step response
s=filter(b,a,x2);
subplot(3,1,2);
stem(n,s);
xlabel('n');
ylabel('s(n)')
title('step response');
%generating sinusoidal signal
t=0:0.1:2*pi;
x3=sin(t);
%sinusoidal response
h2=filter(b,a,x3);
subplot(3,1,3);
stem(h2);
xlabel('n');
ylabel('h(n)');
title('sin response');
%verifing stability
figure;
zplane(b,a);

Result: The Unit sample, unit step and sinusoidal response of the given LTI system is
computed and its stability verified.

Hence all the poles lie inside the unit circle, so system is stable.

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Output:

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Experiment No-9
GIBBS PHENOMENON

AIM: Verify the Gibbs phenomenon.

Software Required:
Matlab software

Theory:
The Gibbs phenomenon, the Fourier series of a piecewise continuously
differentiable periodic function behaves at a jump discontinuity. Then the approximated
function shows amounts of ripples at the points of discontinuity. This is known as the
Gibbs Phenomena . 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.

Program:

clc;
clear all;
close all;
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
%Fourier series of sin signal
y=y+sin(k*t)/k;
subplot(5,1,h);
plot(t,y);
xlabel('k');
ylabel('amplitude');
h=h+1;
end

Result: Gibbs phenomenon is verified.

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Output:

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Experiment 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.

Software Required:
Matlab software

Theory:

Fourier Transform :
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 e2πinx/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

Program:

clc;
clear all;
close all;
fs=1000;
N=1024;
%length of fft sequence
t=[0:N-1]*(1/fs);

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%input signal
x=0.8*cos(2*pi*100*t);
subplot(3,1,1);
plot(t,x);
axis([0 0.05 -1 1]);
grid;
xlabel('t');
ylabel('amplitude');
title('input signal');
%magnitude spectrum
x1=fft(x);
k=0:N-1;
Xmag=abs(x1);
subplot(3,1,2);
plot(k,Xmag);
grid;
xlabel('t');
ylabel('amplitude');
title('magnitude of fft signal') ;
%phase spectrum
Xphase=angle(x1)*(180/pi);
subplot(3,1,3);
plot(k,Xphase);
grid;
xlabel('t');
ylabel('angleindegrees');
title('phase of fft signal');

Result: Magnitude and phase spectrum of FFT of a given signal is plotted.

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Output:

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Experiment No-11(a)
LAPLACE TRANSFORM & INVERSE LAPLACE TRANSFORM

AIM: Finding the Laplace transform & Inverse Laplace transform of some signals.

Software Required:
Matlab software

Theory:

Bilateral Laplace transforms:

The Laplace transform of a signal f(t) can be defined as follows:

Inverse Laplace transform

The inverse Laplace transform is given by the following formula :

Program:

clc;
clear all;
close all;
%representation of symbolic variables
syms f t w s;
%laplace transform of t
f=t;
z=laplace(f);
disp('the laplace transform of f = ');
disp(z);

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%laplace transform of sin signal

f1=sin(w*t);
f2=sin(w*(t-1));
v1=laplace(f1);
disp('the laplace transform of f1 = ');
disp(v1);
v2=laplace(f2);
disp('the laplace transform of f2 = ');
disp(v2);
simplify(v2)
%inverse laplace transform
y1=ilaplace(z);
disp('the inverse laplace transform of z = ');
disp(y1);
y2=ilaplace(v1);
disp('the inverse laplace transform of v1 = ');
disp(y2);
y3=ilaplace(v2);
disp('the inverse laplace transform of v2 = ');
disp(y3);
x1=11*s^3+52*s^2+86*s+72;
x2=4*s*(s+2)^2*(s+3);
x3=x1/x2;
y4=ilaplace(x3);
disp('the inverse laplace transform of x3 =');
disp(y4);

Result: The Laplace transform & Inverse Laplace transform of signals by using matlab
software is verified.

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Output:

the laplace transform of f =

1/s^2

the laplace transform of f1 =

w/(s^2 + w^2)

the laplace transform of f2 =

(w*cos(w) - s*sin(w))/(s^2 + w^2)

ans =

(w*cos(w) - s*sin(w))/(s^2 + w^2)

the inverse laplace transform of z =

t

the inverse laplace transform of v1 =

(w*sin(t*(w^2)^(1/2)))/(w^2)^(1/2)

the inverse laplace transform of v2 =

(exp(t*(-w^2)^(1/2))*(w*cos(w) - sin(w)*(-w^2)^(1/2)))/(2*(-w^2)^(1/2)) -
(w*cos(w) + sin(w)*(-w^2)^(1/2))/(2*exp(t*(-w^2)^(1/2))*(-w^2)^(1/2))

the inverse laplace transform of x3 =

5/(4*exp(3*t)) - (5*t)/(2*exp(2*t)) + 3/2

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Experiment No-11(b)
WAVEFORM SYNTHESIS USING LAPLACE TRANSFORMS

AIM: Perform the waveform synthesis using Laplace transform of a given signal.

Software Required:
Matlab software

Theory:
Consider the square wave of period ‘T’which can be represented with unit
step functions as
f(t)=u(t)-2u(t-T/2)+2u(t-T)-2u(t-3T/2)+………..

Program:

clc;
clear all;
close all;
syms s;
T=1;
%representation of laplace transform
F1=1-exp(-T*s/2);
F2=s*(1+exp(-T*s/2));
F=F1/F2;
%inverse laplace transform
f=ilaplace(F);
disp('inverse laplace transform of F=');
disp(f);
%easy plot for 'f'
ezplot(f);

Result: The waveform synthesis using Laplace transform of a given signal is plotted.

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Output:

inverse laplace transform of F=

heaviside(t - 1/2)*((-1)^floor(1 - 2*t)/2 - 1/2) - (-1)^floor((-2)*t)/2 + 1/2

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Experiment No-12
LOCATING POLES AND ZEROS IN S-PLANE & Z-PLANE

AIM: Write the program for locating poles and zeros and plotting pole-zero maps in s-
plane and z-plane for the given transfer function.

Software Required:
Matlab software

Theory:

Z-transforms

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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Example:

Program:

clc;
clear all;
close all;
%enter the numerator and denamenator cofficients in square brackets
num=input('enter the numerator cofficients');
den=input('enter the denamenator cofficients');
%find the transfer function using built-in function 'filt'
H=filt(num,den);
%find locations of zeros
z=zero(H);
disp('zeros are at');
disp(z);
%find residues,pole locations and gain constant of H(z)
[r p k]=residuez(num,den);
disp('poles are at');
disp(p);
%plot the pole zero map in z-plane
zplane(num,den);
title('pole-zero map of LTI system in z-plane');
% ploe-zero plot in s-plane
H1=tf(num,den); % find transfer function H(s)%
[p1,z1]=pzmap(H1); % find the locations of poles and zeros%
disp('poles ar at ');
disp(p1);
disp('zeros ar at ');
disp(z1);
figure;
%plot the pole-zero map in s-plane
pzmap(H1);
disp(H1);
title('pole-zero map of LTI system in s-plane');

Result :Pole-zero maps are plotted in s-plane and z-plane for the given transfer
function.

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Output:

enter the numerator cofficients[1 -1 4 3.5]

enter the denamenator cofficients[2 3 -2.5 6]

Transfer function:
1 - z^-1 + 4 z^-2 + 3.5 z^-3
------------------------------
2 + 3 z^-1 - 2.5 z^-2 + 6 z^-3

zeros are at
0.8402 + 2.1065i
0.8402 - 2.1065i
-0.6805

poles are at
-2.4874
0.4937 + 0.9810i
0.4937 - 0.9810i

Transfer function:
s^3 - s^2 + 4 s + 3.5
-------------------------
2 s^3 + 3 s^2 - 2.5 s + 6

poles ar at
-2.4874
0.4937 + 0.9810i
0.4937 - 0.9810i

zeros ar at
0.8402 + 2.1065i
0.8402 - 2.1065i
-0.6805

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Experiment No-13
GENERATION OF GAUSSIAN NOISE

AIM: Write the program for generation of Gaussian noise and computation of its mean,
mean square value, skew, kurtosis and probability distribution function.

Software Required:
Matlab software

Theory:

Program:
clc;
clear all;
close all;
%generates first set of 2000 samples of Gaussian distributed
random numbers%
x1=randn(1,2000);
%generates second set of 2000 samples of Gaussian distributed
random numbers%
x2=randn(1,2000);
%plot the joint distribution of both the sets using dot%
plot(x1,x2,'.');
title('scatter plot of gaussian distributed random numbers');
%generates two sets of 2000 samples of uniform distributed
random numbers%
x3=rand(1,2000);
x4=rand(1,2000);
figure;
plot(x3,x4,'.');
title('scatter plot of uniform distributed random numbers');
figure;
subplot(2,1,1);
%plot the histogram graph of x2
hist(x2);
title('gaussian distribution');
subplot(2,1,2);
%plot the histogram graph of x4
hist(x4);
title('uniform distribution');
ymu=mean(x2);disp('ymu');
ymsq=sum(x2.^2)/length(x2);
ysigma=std(x2);
yvar=var(x2);
yskew=skewness(x2);
ykurt=kurtosis(x2);

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Output:

ymu =

0.0172
ymsq =

0.9528
ysigma =

0.9762
yvar =

0.9530
yskew =

-0.0041

ykurt =

2.9381

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Experiment No-14
SAMPLING THEOREM VERIFICATION

AIM: Verify the sampling theorem.

Software Required:
Matlab software

Theory:

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)

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

To recover the original signal G(w):

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

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

Aliasing ws < 2wm.

Figure 6: Aliasing due to inadequate sampling

Aliasing leads to distortion in recovered signal. This is the reason why sampling frequency
should be at least twice the bandwidth of the signal.

Oversampling ws >2wm. This condition avoids aliasing.

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Figure 7: Oversampled signal-avoids aliasing

Program:

clc;
clear all;
close all;
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;

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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;
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;

Result: Sampling theorem is verified.

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OUTPUT:

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Experiment No-15
REMOVAL OF NOISE BY AUTO CORRELATION/CROSS
CORRELATION

AIM: Write the program for Removal of noise by using auto correlation.

Software Required:
Matlab software

Theory:
Detection of a periodic signal masked by random noise is of great 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
Qsn(T)= cross correlation function of s(t) and n(t) Then Qsn(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 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/
=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)

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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 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).

Program:
clc;
clear all;
close all;
t=0:0.2:pi*8;
%input signal
s=sin(t);
subplot(6,1,1);
plot(s);
title('signal s');
xlabel('t');
ylabel('amplitude');
%generating noise
n = randn([1 126]);
%noisy signal
f=s+n;
subplot(6,1,2)
plot(f);
title('signal f=s+n');
xlabel('t');
ylabel('amplitude');
%aucorrelation of input signal
as=xcorr(s,s);
subplot(6,1,3);
plot(as);
title('auto correlation of s');
xlabel('t');
ylabel('amplitude');
%aucorrelation of noise signal
an=xcorr(n,n);
subplot(6,1,4)
plot(an);
title('auto correlation of n');
xlabel('t');
ylabel('amplitude');

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%aucorrelation of transmitted signal

cff=xcorr(f,f);
subplot(6,1,5)
plot(cff);
title('auto correlation of f');
xlabel('t');
ylabel('amplitude');
%aucorrelation of received signal
hh=as+an;
subplot(6,1,6)
plot(hh);
title('addition of as+an');
xlabel('t');
ylabel('amplitude');

Result: Removal of noise using autocorrelation is performed.

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Experiment No-16
EXTRACTION OF PERIODIC SIGNAL MASKED BY NOISE USING
CORRELATION

AIM: Write the program for extraction of periodic signal using correlation.

Software Required:
Matlab software

Theory:
A signal is masked by noise can be detected either by correlation techniques or by
filtering.Actually, the two techniques are equivalent. The correlation technique is a
measure of extraction of a given signal in the time domain whereas filtering achieves
exactly the same results in frequency domain.

Program:

clear all;
close all;
clc;
t=0:0.1:pi*4;
%input signal1
s=sin(t);
subplot(7,1,1)
plot(s);
title('signal s');
xlabel('t');
ylabel('amplitude');
%input signal2
c=cos(t);
subplot(7,1,2)
plot(c);
title('signal c');
xlabel('t');
ylabel('amplitude');
%generating noise signal
n = randn([1 126]);
%signal+noise
f=s+n;
subplot(7,1,3);
plot(f);
title('signal f=s+n');
xlabel('t');
ylabel('amplitude');

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%crosscorrelation of signal1&signal2
asc=xcorr(s,c);
subplot(7,1,4)
plot(asc);
title(' correlation of s and c');
xlabel('t');
ylabel('amplitude');
%crosscorrelation of noise&signal2
anc=xcorr(n,c);
subplot(7,1,5)
plot(anc);
title(' correlation of n and c');
xlabel('t');
ylabel('amplitude');
%crosscorrelation of f&signal2
cfc=xcorr(f,c);
subplot(7,1,6)
plot(cfc);
title(' correlation of f and c');
xlabel('t');
ylabel('amplitude');
%extracting periodic signal
hh=asc+anc;
subplot(7,1,7)
plot(hh);
title('addition of sc+nc');
xlabel('t');
ylabel('amplitude');

Result: Periodic signal is extracted by using correlation.

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Output:

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Experiment No-17
VERIFICATION OF WIENER-KHINCHIN RELATION

AIM: Verification of wiener-khinchin relation

Software Required:
Matlab software

Theory:
The wiener-khinchin theorem states that the power spectral density of a wide sense
stationary random process is the Fourier transform of the corresponding autocorrelation
function.

Continuous case:
Sxx(f)=∫rxx(T)e-j2πft dT
Where
rxx(T)=E[x(t)x*(t-T)]

Discrete case:

Sxx(f)=∑rxx[k]e-j2πkf
Where
rxx[k]=E[x(n)x*(n-k)]

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Program:
clc;
clear all;
close all;
t=0:0.1:2*pi;
%input signal
x=sin(2*t);
subplot(3,1,1);
plot(x);
xlabel('time');
ylabel('amplitude');
title('input signal');
%autocorrelation of input signal
xu=xcorr(x,x);
%fft of autocorrelation signal
y=fft(xu);
subplot(3,1,2);
plot(abs(y));
xlabel('f');
ylabel('amplitude');
title('fft of autocorrelation of input signal');
%fourier transform of input signal
y1=fft(x);
%finding the power spectral density
y2=(abs(y1)).^2;
subplot(3,1,3);
plot(y2);
xlabel('f');
ylabel('magnitude');
title('PSD of input signal');

Result: wiener-khinchin relation is verified.

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Output::

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18. CHECKING A RANDOM PROCESS FOR STATIONARITY IN WIDE SENSE.

AIM: Checking a random process for stationarity in wide sense.

EQUIPMENTS: 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 letrepresent 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 τ, 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

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,

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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:
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

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 E*X(t)+=μX(t)= μX(t+T)
 The Autocorrelation function is written as a function of T that is
 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).

PROGRAM:

clc;
clear all;
close all;
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

RESULT: Checking a random process for stationarity in wide sense is verified

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