Lab 2 Elementary Signals

Objective:
To study and plot the models of some elementary signals.
Theory:
A signal can be defined as a function of one or more variables that conveys information about some
physical phenomenon. Signals are modeled by mathematical functions. In the study of signals and
systems, several elementary signals feature prominently. These include the step function, the impulse
function, exponential signals, and sinusoidal signals. Not only that these signals occur naturally in a
wide range of physical systems, they also serve as building blocks for constructing more complex
signals.
In this lab, we will study such signals and learn to model these signals using Matlab. We will limit
signals to having one independent variable only. For continuous-time signals, the independent
variable will be time t, and for discrete-time signals, the independent variable will be the sample
number n. We will be modeling the following signals:
The Unit Step Function
The unit step function in continuous-time is defined as:
0, 0
( ) .
1, 0
t
u t
t
<
=

>

Notice that the function has a discontinuity at 0 t = .
In discrete-time, the unit step function is defined as:
0, 0
[ ] .
1, 0
n
u n
n
<
=

>

The Impulse Function
The continuous-time unit-impulse function is defined as:
1, 0
( ) .
0, 0
t
t
t
o
=
=

=

The unit impulse signal in discrete-time is given as:
1, 0
[ ] .
0, 0
n
n
n
o
=
=

It should be noted that in case of continuous-time impulse, the value of ( ) t o specifies the weight
rather than the amplitude of the impulse.
The Real Exponential
A continuous-time real exponential signal is defined as:
( ) ,
at
x t Ae =

where A is the amplitude of the signal at 0 t = and o is a real number. For 1 o < , the signal decays
while for 1 o > , the signal grows exponentially.
The discrete-time real exponential signal is expressed as:
[ ] ,
n
x n Ae
o
=

where A and o have same meanings as in the previous equation. .
The Sinusoidal Signal
In its most general form, the continuous-time sinusoidal signal can be given as:
0
( ) cos( ) x t A t e | = +

where A is the amplitude,
0
e is the angular frequency in radians per second, and | is the phase
angle in radians.
The continuous-time sinusoidal signal is a periodic signal with time period given by
0
2 / T t e = .
The discrete-time sinusoidal signal can be expressed as:
[ ] cos( ) x n A n | = O +

If n is taken to be dimensionless, then both O and | have units of radians. It must be noted that a
discrete-time sinusoidal signal may or may not be periodic. For it to be periodic with a period of N
samples, Omust be a rational multiple of 2t .
Modeling Elementary Signals
Now, we will model the above signals using Matlab. In general, these signal models are defined for
all t and all n, however, we will be modeling the signals within a small time-span around 0 n = or
0 t = . It should also be noted that Matlab works on discrete-time signals. Hence, we will only be
approximating continuous-time signals by keeping a very small sampling interval and interpolating
the samples using plot function of Matlab. Type and run the following programs in you m-file editor
and run the programs.
The output of the program can be seen in Figure 2.1.
-10 -8 -6 -4 -2 0 2 4 6 8 10
0
0.2
0.4
0.6
0.8
1
Sample Number
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Discrete-time Unit Impulse Function

Figure 2. 1. Output of Lab 2.1
%% Lab 2.1 - Generating Unit Impulse Function

%% Initializing
close all
clear all
clc

N=10;% Number of sample on both sides of n = 0
n=[-N:1:N];

%% Generating the Impulse
imp=zeros(1,length(n));%Generating a vector of zeros
imp(n==0)=1;%Setting the sample at n=0 to 1

%% Plotting the function
stem(n,imp,'LineWidth',2);
axis([n(1) n(length(n)) min(imp)-0.1 max(imp)+0.1]);
xlabel('Sample Number');
ylabel('Magnitude');
title('Discrete-time Unit Impulse Function');
grid on;


Notice the sampling interval of 0.0011 seconds to generate the time vector. This sampling interval
avoids the signal to be sampled at t = 0 at which the function is undefined. The output of the
program is shown in Figure 2.2.
%% Lab 2.2 - Generating Unit Step Function

%% Initializing
close all
clear all
clc

%% Discrete-time
N=10;% Number of sample on both sides of n = 0
n=[-N:1:N];

%% Generating the Step Function
DiscreteStep=zeros(1,length(n));%Generating a vector of zeros
DiscreteStep(n>=0)=1;%Setting the sample at n>=0 to 1

%% Plotting the function
figure
subplot(211);
stem(n,DiscreteStep,'LineWidth',2);
axis([n(1) n(length(n)) min(DiscreteStep)-0.1 max(DiscreteStep)+0.1]);
xlabel('Sample Number');
ylabel('Magnitude');
title('Discrete-time Unit Step Function');
grid on;

%% Continuous-time
T=2;% Time on both sides of t = 0
t=[-T:0.0011:T]; %

%% Generating the Step Function
ContStep=zeros(1,length(t));%Generating a vector of zeros
ContStep(t>0)=1;%Setting the sample at n=0 to 1

%% Plotting the function
subplot(212)
plot(t,ContStep,'LineWidth',2);
axis([t(1) t(length(t)) min(ContStep)-0.1 max(ContStep)+0.1]);
xlabel('Time');
ylabel('Magnitude');
title('Continuous-time Unit Step Function');
grid on;

-10 -8 -6 -4 -2 0 2 4 6 8 10
0
0.2
0.4
0.6
0.8
1
Sample Number
M
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Discrete-time Unit Step Function
-2 -1.5 -1 -0.5 0 0.5 1 1.5
0
0.2
0.4
0.6
0.8
1
Time
M
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Continuous-time Unit Step Function

Figure 2. 2. Output of Lab 2.2
%% Lab 2.3 - Generating Exponential Signals

%% Initializing
close all
clear all
clc

%% Discrete-time
N=10;% Number of sample on both sides of n = 0
n=[-N:1:N];
alpha = 0.1;
A=1;

DiscreteExp1=A*exp(alpha*n);
DiscreteExp2=A*exp(-alpha*n);



The output can be seen in Figure 2.3.
%% Plotting the signals
figure
subplot(221);
stem(n,DiscreteExp1,'LineWidth',2);
axis([n(1) n(length(n)) min(DiscreteExp1)-0.1 max(DiscreteExp1)+0.1]);
xlabel('Sample Number');
ylabel('Magnitude');
grid on

subplot(222);
stem(n,DiscreteExp2,'LineWidth',2);
axis([n(1) n(length(n)) min(DiscreteExp2)-0.1 max(DiscreteExp2)+0.1]);
xlabel('Sample Number');
ylabel('Magnitude');
grid on;

%% Continuous-time
T=2;% Number of sample on both sides of n = 0
t=[-T:0.001:T];
alpha = 0.5;
A=1;

ContExp1=A*exp(alpha*t);
ContExp2=A*exp(-alpha*t);

%% Plotting the signals
subplot(223);
plot(t,ContExp1,'LineWidth',2);
axis([t(1) t(length(t)) min(ContExp1)-0.1 max(ContExp1)+0.1]);
xlabel('Time');
ylabel('Magnitude');
grid on

subplot(224);
plot(t,ContExp2,'LineWidth',2);
axis([t(1) t(length(t)) min(ContExp2)-0.1 max(ContExp2)+0.1]);
xlabel('Time');
ylabel('Magnitude');
grid on;


-10 -5 0 5 10
0.5
1
1.5
2
2.5
Sample Number
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-10 -5 0 5 10
0.5
1
1.5
2
2.5
Sample Number
M
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-2 -1 0 1 2
0.5
1
1.5
2
2.5
Time
M
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-2 -1 0 1 2
0.5
1
1.5
2
2.5
Time
M
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Figure 2. 3. Output of Lab 2.3

%% Lab 2.4 - Generating Sinusoidal Signals

%% Initializing
close all
clear all
clc

%% Continuous-time
T=2;
t=[-T:0.001:T];
A=2;
omega=2*pi;
phi=pi/4;

x=A*sin(omega*t+phi);


The output of the program can be seen in Figure 2.4.
-2 -1.5 -1 -0.5 0 0.5 1 1.5 2
-2
-1
0
1
2
Sinusoidal Signal
Sample Number
M
a
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-20 -15 -10 -5 0 5 10 15 20
-2
-1
0
1
2
Time
M
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u
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Figure 2. 4. Output of Lab 2.4
%% Plotting
subplot(211);
plot(t,x,'LineWidth',2);
title('Sinusoidal Signal');
axis([-T T min(x)-0.1 max(x)+0.1]);
xlabel('Sample Number');
ylabel('Magnitude');
grid on

%% Discrete-time
N=20;
n=[-N:1:N];
A=2;
omega=pi/8;
phi=pi/4;
y=A*sin(omega*n+phi);

%% Plotting
subplot(212);
stem(n,y,'LineWidth',2);
axis([-N N min(y)-0.1 max(y)+0.1]);
xlabel('Time');
ylabel('Magnitude')
grid on


Exercises

Write Matlab code to generate and plot the following signals:
1. ( )
t
x t Ae
o
= where 1 A= . Try different values of o and observe the effect of changing the
values.


2. For the continuous-time signal in Lab 2.4, what is the time delay corresponding to the phase
angle of / 4 t radians. Change the value of | to 0 and / 2 t and comment on the result.

3. For the discrete-time signal in Lab 2.4, change the values of | to / 2 t and / 3 t and
comment on the periodicity of the signal.




Lab 3 Basic Signal Operations
Objective:
To perform basic signal operations on the elementary signals.
Theory:
Basic operations on signals can be classified into two classes:
1. Operations performed on dependent variables:
a. Amplitude scaling
b. Addition
c. Multiplication

2. Operations performed on independent variables:
a. Time scaling
b. Reflection
c. Time shifting
Amplitude Scaling
For a continuous-time signal ( ) x t , the signal ( ) y t resulting from amplitude scaling applied to ( ) x t is
given by:
( ) ( ) y t cx t =
where c is the scaling factor.
Likewise, for a discrete-time signal [ ] x n , the signal [ ] y n resulting from amplitude scaling applied to
[ ] x n is given by:
[ ] [ ] y n cx n =
Signal Addition
Let
1
( ) x t and
2
( ) x t denote a pair of discrete-time signals, then the signal ( ) y t obtained by the
addition of
1
( ) x t and
2
( ) x t is defined by:
1 2
( ) ( ) ( ) y t x t x t = +

Likewise, addition of two discrete-time signals is given by:
1 2
[ ] [ ] [ ] y n x n x n = +

Signal Multiplication
Let
1
( ) x t and
2
( ) x t denote a pair of discrete-time signals, then the signal ( ) y t obtained by the
multiplication of
1
( ) x t and
2
( ) x t

is defined by:
1 2
( ) ( ) ( ) y t x t x t =
Similarly, in discrete-time, multiplication of two signals can be expressed as:
1 2
[ ] [ ] [ ] y n x n x n =

This means that for each n the value of [ ] y n is given by the product of the corresponding values of
1
[ ] x n and
2
[ ] x n .
Reflection
Given a signal ( ) x t , its time reversed or reflected version is given by ( ) ( ) y t x t = . The signal ( ) y t
represents a reflected version of ( ) x t about 0 t = .
Time Scaling
Let ( ) x t denotes a continuous-time signal. Then the signal ( ) y t obtained by scaling the independent
variable, time t, by a factor a is defined as
( ) ( ) y t x at =
If a > 1, ( ) y t is a compressed version of ( ) x t . If 0 < a < 1, then ( ) y t is an extended version of ( ) x t .
Time Shifting
Given a signal ( ) x t , its time-shifted version is defined by:
0
( ) ( ) y t x t t =
where
0
t is the time-shift. For
0
0 t > , the waveform of ( ) y t is obtained by shifting ( ) x t towards the
right, relative to the time-axis. If
0
0 t < , ( ) x t is shifted to the left.
Basic Signal Operations using Matlab
Run the following Matlab programs. The output of Lab 3.1 can be seen in Figure 3.1, while the
output of Lab 3.2 is shown in Figure 3.2.


%% Lab 3.1 - Amplitude scaling

%% Initializing
close all
clear all
clc

%% Generating Exponential Signal
N=10;% Number of sample on both sides of n = 0
n=[-N:1:N];
alpha = 0.1;
A=1;

DiscreteExp=A*exp(alpha*n);

%% Amplitude Scaling
ScaledExp1=2*DiscreteExp;
ScaledExp2=0.3*DiscreteExp;

%% Plotting the signals
figure
subplot(131);
stem(n,DiscreteExp,'LineWidth',2);
axis([n(1) n(length(n)) min(DiscreteExp)-0.1 max(DiscreteExp)+0.1]);
xlabel('Sample Number');
ylabel('Magnitude');
grid on

subplot(132);
stem(n,ScaledExp1,'LineWidth',2);
axis([n(1) n(length(n)) min(ScaledExp1)-0.1 max(ScaledExp1)+0.1]);
xlabel('Sample Number');
ylabel('Magnitude');
grid on;

subplot(133);
stem(n,ScaledExp2,'LineWidth',2);
axis([n(1) n(length(n)) min(ScaledExp2)-0.1 max(ScaledExp2)+0.1]);
xlabel('Sample Number');
ylabel('Magnitude');
grid on;



%% Lab 3.2 - Signal Addition and Multiplication

%% Initializing
close all
clear all
clc
N=10;% Number of sample on both sides of n = 0
n=[-N:1:N];

%% Generating Exponential Signal
alpha = 0.1;
A=1;
DiscreteExp=A*exp(alpha*n);

%% Generating Sinusoidal Signal
A=2;
omega=pi/4;
phi=0;
SineWave=A*sin(omega*n+phi);

%% Adding two signals
y=DiscreteExp+SineWave;

%% Multiplying two signals

%multiplication

%% Plotting the signals
figure
subplot(221);
stem(n,DiscreteExp,'LineWidth',2);
axis([n(1) n(length(n)) min(DiscreteExp)-0.1 max(DiscreteExp)+0.1]);
xlabel('Sample Number');
ylabel('Magnitude');
grid on
title('Signal 1');

subplot(222);
stem(n,SineWave,'LineWidth',2);
axis([n(1) n(length(n)) min(SineWave)-0.1 max(SineWave)+0.1]);
xlabel('Sample Number');
ylabel('Magnitude');
grid on;
title('Signal 2');

subplot(223);
stem(n,y,'LineWidth',2);
axis([n(1) n(length(n)) min(y)-0.1 max(y)+0.1]);
xlabel('Sample Number');
ylabel('Magnitude');
grid on;
title('Signal1 + Signal2')

-10 0 10
0.5
1
1.5
2
2.5
Sample Number
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-10 0 10
1
1.5
2
2.5
3
3.5
4
4.5
5
5.5
Sample Number
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-10 0 10
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
Sample Number
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Figure 3. 1. Output for Lab 3.1





subplot(224);
stem(n,z,'LineWidth',2);
axis([n(1) n(length(n)) min(z)-0.1 max(z)+0.1]);
xlabel('Sample Number');
ylabel('Magnitude');
grid on;
title('Signal1 x Signal2');


Lab 4 Convolution
Objective:
To perform convolution of continuous-time signals.
Theory:

Is a mathematical way of combining two signals to achieve a third, modified signal. The signal we record
seems to respond well to being treated as a series of signals superimposed upon each other that is seismic
signals seem to respond convolutionally. The process of DECONVOLUTION is the reversal of the
convolution process.

Mathematically, a convolution is defined as the integral over all space of one function at x times
another function at u-x. The integration is taken over the variable x (which may be a 1D or 3D
variable), typically from minus infinity to infinity over all the dimensions. So the convolution is
a function of a new variable u, as shown in the following equations. The cross in a circle is used
to indicate the convolution operation.

Note that it doesn't matter which function you take first, i.e. the convolution operation is
commutative.





Basic Example
Let us look at a basic continuous-time convolution example to help express some of the important ideas. We will
convolve together two square pulses, x(t) and h(t), as shown in Figure 1





(a)

(b)


Figure 1: Two basic signals that we will convolve together.
Reflect and Shift
Now we will take one of the functions and reflect it around the y-axis. Then we must shift the function, such that the
origin, the point of the function that was originally on the origin, is labeled as point t. This step is shown in Figure
2, h(t).
(a) Reflected square pulse. (b) Reflected and shifted square pulse.



Figure 2: h() and h(t).
Note that in Figure 2 is the 1st axis variable while t is a constant (in this figure). Since convolution is commutative
it will never matter which function is reflected and shifted; however, as the functions become more complicated
reflecting and shifting the "right one" will often make the problem much easier.
Regions of Integration
We start out with the convolution integral, y(t)=x()h(t)d. The value of the function y at time t is
given by the amount of overlap(to be precise the integral of the overlapping region) between h(t) and x().
Next, we want to look at the functions and divide the span of the functions into different limits of integration. These
different regions can be understood by thinking about how we slide h(t) over x(), see Figure 3.
(a) No overlap.

(b) h(t) on its way "into" x()

(c) h(t) on its way "out of" x()

(d) No overlap.


Figure 3: Figures to help understand the regions of intergration
In this case we will have the following four regions. Compare these limits of integration to the four illustrations
of h(t) and x() in Figure 3.
Four Limits of Integration
1. t<0
2. 0t<1
3. 1t<2
4. t2
Using the Convolution Integral
Finally we are ready for a little math. Using the convolution integral, let us integrate the product of x()h(t). For
our first and fourth region this will be trivial as it will always be 0. The second region, 0t<1, will require the
following math:
y(t)==t01d=t
(1)
The third region, 1t<2, is solved in much the same manner. Take note of the changes in our integration though.
As we move h(t) across our other function, the left-hand edge of the function, t1, becomes our lowlimit for
the integral. This is shown through our convolution integral as
y(t)===1t11d=1(t1)=2t
(2)
The above formulas show the method for calculating convolution; however, do not let the simplicity of this example
confuse you when you work on other problems. The method will be the same, you will just have to deal with more
math in more complicated integrals.
Note that the value of y(t) at all time is given by the integral of the overlapping functions. In this example y for a
given t equals the gray area in the plots in Figure 3.
Convolution Results
(3)
Now that we have found the resulting function for each of the four regions, we can combine them together and
graph the convolution of x(t)h(t).

Figure 4: Shows the system's output in response to
the input, x(t).

Matlab code


% Description: This m-file convolves two signals and plots the result.
%

x = [0.5 0.5 0.5]; % define input signal x[n]
h = [3.0 2.0 1.0]; % define unit-pulse response h[n]

y = conv(x,h); % compute output y[n] via convolution

n = 0:(length(y)-1); % create n-vector for plotting y[n]

stem(n,y); % plot y[n]
grid;
xlabel('n');
ylabel('y[n]');
title('Output of System via Convolution');









0 0.5 1 1.5 2 2.5 3 3.5 4
0
0.5
1
1.5
2
2.5
3
n
y
[
n
]
Output of System via Convolution



clear all
x=[2 3 4 1 2 3 7 8]; %input signal to the system
t=1:10;
h=exp(-t); %System response of the system defined from 1 to 10
sec
y=conv(x,h);
plot(y)
grid on
xlabel('time')
ylabel('output')

Exercise;

Q1) Convolve the A
t
and the unit step function when has the decreasing curve.
Q2) Prove the output of the convolution integral example in the matlab.


0 2 4 6 8 10 12 14 16 18
0
0.5
1
1.5
2
2.5
3
3.5
4
4.5
time
o
u
t
p
u
t
Convolution
Lab 5 Constructing periodic signals from harmonically
related sinusoidal signals

Objectives
To generate periodic signals by combining harmonically related sinusoidal signals
Theory
A signal is periodic if, for some positive value of T, ( ) ( ) x t T x t + = for all t. The fundamental period
of ( ) x t is the minimum positive nonzero value of t for which the above equation is satisfied.

The sinusoidal signal
0
( ) cos( ) x t A t e | = + and the complex exponential signal
0
( ) j t
Ae
e | +
are both
periodic with fundamental frequency
0
e and fundamental period
0
2 / T t e = . Associated with the
complex exponential signal is the set of harmonically related complex exponentials given as:
0
(2 / )
( ) , 0, 1 2,...
jk t jk T t
k
t e e k
e t
u = = =

A linear combination of harmonically related complex exponentials of the form
0 0
(2 / )
( )
jk t jk T t
k k
k k
x t a e a e
e t

= =
= =


is also periodic with period T. In the above equation, the term for k = 0 is a constant, while the
terms for both k = 1 and k = -1have fundamental frequency equal to
0
e and are referred to as
fundamental components or the first harmonic components. In general, the components for k = N
and k = -N are called Nth harmonic components.
Let
3
2
3
( )
jk t
k
k
x t a e
t
=
=


where
0
1 a = ,
1 1
1/ 4 a a

= = ,
2 2
1/ 2 a a

= = ,
3 3
1/ 3 a a

= = .

The above equation can be re-written by collecting the harmonic components with same
fundamental frequencies as follows:
( ) ( ) ( )
2 2 4 4 6 6
1 1 1
( ) 1
4 2 3
j t j t j t j t j t j t
x t e e e e e e
t t t t t t
= + + + + + +
Using Eulers theorem:
1 2
( ) 1 cos2 cos4 cos6
2 3
x t t t t t t t = + + +
In the following code, we will add these harmonically related components one by one to see the
effect of adding each component on the shape of the resulting signal.
MATLAB CODE for constructing periodic signals from harmonically
related signals

%% Lab 5.1

%% Initializing
close all
clear all
clc

t=[-5:0.01:5]; % time indexing vector
h0=1
h1=0.5*cos(2*pi*t); % returns the cosine value for all values of t
h2=cos(4*pi*t);
h3=2/3*cos(6*pi*t);
figure;
subplot(2,2,1);
plot(t,h0);
xlabel('Time');
ylabel('h0');
subplot(2,2,2);
plot(t,h0+h1);
xlabel('Time');
ylabel('h0+h1');
subplot(2,2,3);
plot(t,h0+h1+h2);
xlabel('Time');
ylabel('h0+h1+h2');
subplot(2,2,4);
plot(t,h0+h1+h2+h3);
xlabel('Time');
ylabel('h0+h1+h2+h3');



Figure 1 Output for Lab 5.1

Type the following code and execute the program:


%% Lab 5.2

%% Initializing

close all
clear all
clc

t=[-5:0.01:5]; % time indexing vector
x=zeros(1,length(t)); % plotting zeros from 1 till length of t

NumOfHarmonics=10;

%% L # 5.1 executing a statement a predetermined number of times
for n=1:2:NumOfHarmonics % indicating the starting and ending of loop
x=x+4/pi*1/n*sin(n*pi*t/2);
end
plot(t,x);


Figure 2 Output for lab 5.2




Try different number of harmonics to see the effect of including more harmonics to the signal.




Exercises

Write MATLAB code for generating a signal with the information given below. Vary the number of
harmonics being added and comment on the results.
1.
1
1 1 1
( ) sin
2
n
n t
f x
n L
t
t

=
| |
=
|
\ .

Assume L = 1.
t=[-5:0.01:5]; % time indexing vector
x=zeros(1,length(t)); % plotting zeros from 1 till length of t

NumOfHarmonics=10;

%% L # 5.1 executing a statement a predetermined number of times
for n=1:1:NumOfHarmonics % indicating the starting and ending of loop
x=1/2-(x+1/pi*1/n*sin(n*pi*t));
end
figure(1);
hold on;
grid on;
plot(t,x);

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2.
( 1) / 2
2 2
1,3,5,...
8 ( 1)
( ) sin
n
n
n t
f x
n L
t
t

=
| |
=
|
\ .

.

t=[-5:0.01:5]; % time indexing vector
x=zeros(1,length(t)); % plotting zeros from 1 till length of t

NumOfHarmonics=10;

%% L # 5.1 executing a statement a predetermined number of times
for n=1:2:NumOfHarmonics % indicating the starting and ending of loop
y= (n-1)/2;
y= (-1)^y;
x=(x+8/pi^2*(y/n^2)*sin(n*pi*t));
end
figure;
hold on;
grid on;
plot(t,x);
3.
1,3,5
2 2
( ) sin( )cos( )
n
A A n t n t
x t
T n T T
t tt tt
t

=
= +



Use 1 A= , 1 t = , and 2 T = . Change the value of t to 0.5 and 1.5 and comment on the result.





t=[-5:0.01:5]; % time indexing vector
x=zeros(1,length(t)); % plotting zeros from 1 till length of t
A=1;
T=2;
L=1.5;

NumOfHarmonics=10;

%% L # 5.1 executing a statement a predetermined number of times
for n=1:2:NumOfHarmonics % indicating the starting and ending of loop
x=.5+(x+2/(n*pi)*sin(n*pi*L*t/2).*cos(n*pi*L*t));
end
figure(1);
hold on;
grid on;
plot(t,x);


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Lab 6 - Properties of continuous-time Fourier series-I

Objective
To study and verify various properties of Fourier series for continuous time signals
Theory
In Lab, we studied Fourier series and learnt how periodic signals can be constructed by adding
harmonically related complex exponentials. In this and the next lab, we will study various properties
of Fourier series for continuous-time periodic signals.

Let ( ) x t and ( ) y t denote two periodic signals with period T having Fourier series coefficients
denoted by
k
a and
k
b respectively, i.e.
( )
FS
k
x t a and ( )
FS
k
y t b
Then:
Linearity
( ) ( ) ( )
FS
k k k
z t Ax t By t c Aa Bb = + = +
Time Shifting
0 0
0
( )
FS
jk t
k
x t t a e
e
=
Time Reversal
( )
FS
k
x t a

Time Scaling
0
( ) ( )
j t
k
x t x t e
oe
o

=
=



Verifying the properties using MATLAB
Type the following code and execute:


%% Lab# 6.1 Title: Properties of Continuous Time Fourier Series, Linearity
% Generation of 1st signal xt
close all
clear all
clc
t=-1.5:0.005:1.5; % time indexing vector
xcos=cos(2*pi*t); % returns cosine for all values of t
xt=xcos>0;
subplot(2,1,1);
plot(t,xt);
xlabel('t');
ylabel('x(t)')
title('Peridoic Square Wave (T=1, T1=0.250)')
set(gca,'ylim',[-0.1 1.1]);% sets the named properties to specified values
on the object identified by gca
grid on

% Generation of 2nd signal yt
T=1;
T1=0.125;
lenT=T/0.005;
ytemp=zeros(1,lenT); % plotting zeros from 1 till value of lenT
lenT1=T1/0.005;
ytemp(round(lenT/2)-lenT1:round(lenT/2)+lenT1-1)=ones(1,2*lenT1);
yt=[ytemp ytemp ytemp 0];
% The last 0 added to make the size of yt equal to length(t)
subplot(2,1,2);plot(t,yt);xlabel('t');ylabel('y(t)')
title('Periodic Square Wave (T=1, T1=0.125)')
set(gca,'ylim',[-0.1 1.1]); % sets the named properties to specified
values on the object identified by gca
grid
z1t=xt+yt;
figure;
plot(t,z1t);
set(gca,'ylim',[-0.1 2.1]); % sets the named properties to specified
values on the object identified by gca
grid

% FS coefficients of periodic square waves
k=-50:50;
T1=0.25;
ak=sin(k*2*pi*(T1/T))./(k*pi); % returns sine value of angle
% Manual correction for a0 -> ak(51)
ak(k==0)=2*T1/T;
T1=0.125;
bk=sin(k*2*pi*(T1/T))./(k*pi);

% Manual correction for b0 -> bk(51)
bk(k==0)=2*T1/T;
% Application of linearity property of FS
ck=ak+bk;
% Reconstruction with M=50
w0=2*pi/T;
zt=zeros(1,length(t));
for k=-50:50
zt=zt + ck(k+51)*exp(j*k*w0*t);
end
figure;
plot(t,real(zt));grid;xlabel('t');ylabel('z(t)=x(t)+y(t)')
title('Reconstruction from ak+bk''s with 101 terms')








%% L # 6.2 Title: Properties of Continuous Time Fourier Series

%%Time Shifting

%Generation of periodic square wave
close all
clear all
clc

t=-1.5:0.005:1.5;
xcos=cos(2*pi*t);
xt=xcos>0;
% FS coefficients of periodic square wave
k=-50:50;
T=1;T1=0.25;
ak=sin(k*2*pi*(T1/T))./(k*pi);
% Manual correction for a0 -> ak(51)
ak(k==0)=2*T1/T;
%Amount of time shift
t0=0.25;
% FS coefficients of the time shifted signal
w0=2*pi/T;
bk=ak.*exp(-j*k*w0*t0);
% Reconstruction from bk's with 101 terms (M=50)
yt=zeros(1,length(t));
for k=-50:50
yt=yt + bk(k+51)*exp(j*k*w0*t);
end
figure
subplot(2,1,1);
plot(t,xt);
xlabel('t');
ylabel('x(t)')
title('Periodic Square Wave (T=1, T1=0.25)')
set(gca,'ylim',[-0.2 1.2]);
grid
subplot(2,1,2);
plot(t,real(yt))
xlabel('t');ylabel('y(t)=x(t-0.5)')
title('Reconstruction from bk''s with 101 terms')
set(gca,'ylim',[-0.2 1.2]); % sets the named properties to specified
values on the object identified by gca
grid

Exercise
Write MATLAB codes to prove the time reversal and time scaling properties of MATLAB.
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__________________________________________________________________
__________________________________________________________________
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Lab 7 - Properties of continuous-time Fourier series-II
Objective
To study and verify various properties of Fourier series for continuous time signals
Theory
This lab is a continuation of Lab 6, where we studied some properties of Fourier series for
continuous time signals. In this lab we are going to study to two more properties namely,
multiplication and conjugation.

Let ( ) x t and ( ) y t denote two periodic signals with period T having Fourier series coefficients
denoted by
k
a and
k
b , respectively. Then
Multiplication
( ) ( )
FS
k l k l
l
x t y t h a b

=
= =


It can be observed that the right side of the above equation is actually the convolution sum of the
Fourier coefficients
k
a and
k
b .
Conjugation and conjugate symmetry
* *
( )
FS
k
x t a


MATLAB CODE for verifying the properties
Type the following programs and execute:





%% L # 7.1 Title: Properties of Continuous Time Fourier Series
% Multiplication
% Generation of 1Hz cosine
close all
clear all
clc

t=-1.5:0.005:1.5; % time indexing vector
xt=cos(2*pi*t);
% Generation of periodic square wave
yt=xt>0;
% FS coefficients of periodic square wave
k=-50:50;
T=1;
T1=0.25;
ak=sin(k*2*pi*(T1/T))./(k*pi);
% Manual correction for a0 -> ak(51)
ak(51)=2*T1/T;
% FS coefficients of 1Hz cosine (over k=-1..1)
bk=zeros(1,3);
bk(1)=0.5;bk(3)=0.5;
ck=conv(ak,bk);
ck(103)=[];ck(1)=[];

% Reconstruction from ck's with 101 terms (M=50)
w0=2*pi/T;
zt=zeros(1,length(t));
for k=-50:50
zt=zt + ck(k+51)*exp(j*k*w0*t);
end
figure(1);
set(gcf,'defaultaxesfontsize',8)
subplot(3,1,1);
plot(t,xt);
ylabel('x(t)')
title('1 Hz Cosine')
set(gca,'ylim',[-1.2 1.2]);
grid
subplot(3,1,2);
plot(t,yt);
ylabel('y(t)')
title('Periodic Square Wave (T=1, T1=0.25)')
set(gca,'ylim',[-1.2 1.2]);
grid
subplot(3,1,3);
plot(t,real(zt))
xlabel('t');
ylabel('z(t)=x(t)*y(t)')
title('Reconstruction from ck''s')
set(gca,'ylim',[-1.2 1.2]);
grid



%% Lab# 7.2 Title: Properties of Continuous Time Fourier Series
% Conjugation and Conjugate Symmetry
% Generation of 1Hz cosine (Real part of our signal)
close all
clear all
clc

t=-1.5:0.005:1.5;
xt=cos(2*pi*t);
% Generation of periodic square wave (Imaginary part of our signal)
yt=xt>0;
% Our perodic complex valued signal
zt=xt+j*yt;
% FS coefficients of 1Hz cosine, xt (over k=-50:50)
ak=zeros(1,101);
ak(50)=0.5;ak(52)=0.5;
% FS coefficients of periodic square wave, yt
k=-50:50;
T=1;T1=0.25;
bk=sin(k*2*pi*(T1/T))./(k*pi);
% Manual correction for a0 -> ak(51)
bk(51)=2*T1/T;
% FS coefficients of zt (using linearty property of FS)
ck=ak+j*bk;
% Flip ck's and conjugate
dk=conj(fliplr(ck));
% Reconstruction from dk's with 101 terms (M=50)
w0=2*pi/T;zrt=zeros(1,length(t));
for k=-50:50
zrt=zrt + dk(k+51)*exp(j*k*w0*t);
end
figure(1);
set(gcf,'defaultaxesfontsize',8)
subplot(2,2,1);
plot(t,xt);
ylabel('x(t)')
title('Real part of z(t), 1 Hz Cosine')
set(gca,'ylim',[-1.2 1.2]);
grid
subplot(2,2,2);
plot(t,yt);
ylabel('y(t)')
title('Imag. part of z(t), Periodic Square Wave')
set(gca,'ylim',[-1.2 1.2]);grid
subplot(2,2,3);plot(t,real(zrt))
xlabel('t');
ylabel('Re[zr(t)]')
title('Real part of Reconstruction from dk''s')
set(gca,'ylim',[-1.2 1.2]);grid
subplot(2,2,4);
plot(t,imag(zrt))
xlabel('t');
ylabel('Im[zr(t)]')
title('Imag. part of Reconstruction from dk''s')
set(gca,'ylim',[-1.2 1.2]);grid



Exercise
Write MATLAB programs to prove that:

If ( ) x t is real and even, then
k k
a a

=
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____________________________________________________________________

If
( ) x t
is real and odd, then
k k
a a

=

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______________________________________________________________________________
____________________________________


Lab 8 introduction to Symbolic Toolbox

Objective:
To study the use of Symbolic Toolbox for calculus computation and Fourier series.
Theory:
Symbolic computation or algebraic computation or computer algebra relates to algorithms and
software for manipulating mathematical expressions and equations in symbolic form, as opposed to
manipulating the approximations of specific numerical quantities represented by those symbols.
Software applications that perform symbolic calculations are called computer algebra systems.
These systems might be used for symbolic integration or differentiation, substitution of one
expression into another, simplification of an expression, etc., for most operations of calculus and,
more generally, for every computation with mathematical objects for which algorithms are known.
Computer algebra softwares are widely used in many scientific and engineering domains.
Symbolic computation is also sometimes referred to as symbolic manipulation, symbolic
processing, symbolic mathematics, or symbolic algebra, but these terms also refer to non-
computational manipulation.
Computer Algebra Calculation of Fourier Coefficients
A computer algebra system can greatly ease the burden of calculation of the Fourier coefficients of a
given function f(t). In the case of a function defined "piecewise," we must take care to "split" the
integral according to the different intervals of definition of the function. In the paragraphs that
follow we illustrate the use of MATLAB (Symbolic toolbox) in deriving the Fourier series

of the period 2t square wave function defined on (t, t) by



In this case the function is defined by different formulas on two different intervals, so each
Fourier coefficient integral from t to t must be calculated as the sum of two integrals:


To practice the symbolic derivation of Fourier series in this manner, you can begin by verifying
the Fourier series calculated manually. The period 2t triangular wave and trapezoidal wave
functions illustrated in the figures below have especially interesting Fourier series that we invite
you to discover for yourself.


MATLAB code


Figure 3 (output of Lab 8.1)
%% Lab 8.1
%% Initializing
close all
clear all
clc

syms n t pi
%define the cosine coefficients
an = (1/pi)*(int(-cos(n*t),-pi,0)+int(cos(n*t),0,pi))
%define the sine coefficients
bn = (1/pi)*(int(-sin(n*t),-pi,0)+int(sin(n*t),0,pi));
pretty(bn)
%MATLAB does not yet know that n is an integer
bn = subs(bn,'(-1)^n','cos(pi*n)');
pretty(bn)
%set up a typical Fourier sum
FourierSum = (4/pi)*sin(t);
for k = 3:2:50
FourierSum = FourierSum+subs((4/pi)*sin(n*t)/n,k,n);
end
FourierSum
%plot its graph
ezplot(FourierSum, 3.1416*[-2 4])
grid
Lab 9 Fourier Transform
Objectives
- To evaluate the Fourier Transform of signal and plot its real and imaginary part by using
MATLAB.
- Calculate the inverse Fourier Transform.
- Calculate the Power Spectral Density of the define signal.
Theory
A signal is periodic if, for some positive value of T, ( ) ( ) x t T x t + = for all t. The fundamental period
of ( ) x t is the minimum positive nonzero value of t for which the above equation is satisfied.

The sinusoidal signal
0
( ) cos( ) x t A t e | = + and the complex exponential signal
0
( ) j t
Ae
e | +
are both
periodic with fundamental frequency
0
e and fundamental period
0
2 / T t e = . Associated with the
complex exponential signal is the set of harmonically related complex exponentials given as:
0
(2 / )
( ) , 0, 1 2,...
jk t jk T t
k
t e e k
e t
u = = =

A linear combination of harmonically related complex exponentials of the form
0 0
(2 / )
( )
jk t jk T t
k k
k k
x t a e a e
e t

= =
= =


is also periodic with period T. In the above equation, the term for k = 0 is a constant, while the
terms for both k = 1 and k = -1have fundamental frequency equal to
0
e and are referred to as
fundamental components or the first harmonic components. In general, the components for k = N
and k = -N are called Nth harmonic components.
Let
3
2
3
( )
jk t
k
k
x t a e
t
=
=


Where
0
1 a = ,
1 1
1/ 4 a a

= = ,
2 2
1/ 2 a a

= = ,
3 3
1/ 3 a a

= = .

The above equation can be re-written by collecting the harmonic components with same
fundamental frequencies as follows:
( ) ( ) ( )
2 2 4 4 6 6
1 1 1
( ) 1
4 2 3
j t j t j t j t j t j t
x t e e e e e e
t t t t t t
= + + + + + +
Using Eulers theorem:
1 2
( ) 1 cos2 cos4 cos6
2 3
x t t t t t t t = + + +

Fourier analysis
Fourier analysis follows from Fouriers theorem, which states that every function can be completely
expressed as a sum of sines and cosines of various amplitudes and frequencies. This is a pretty
impressive assertion no matter what the shape of a function, and how little it looks like a sine
wave, it can be rewritten as a sum of sines and cosines. The Fourier series tells you the amplitude
and frequency of the sines and cosines that you should add up to recreate your original function.

Power spectral density
The above definitions of energy spectral density require that the Fourier transforms of the signals
exist, that is, that the signals are integrable/summable or square-integrable/square-summable. (Note:
The integral definition of the Fourier transform is only well-defined when the function is integrable.
It is not sufficient for a function to be simply square-integrable. In this case one would need to use
the Plancherel theorem.) An often more useful alternative is the power spectral density (PSD),
which describes how the power of a signal or time series is distributed with frequency. Here power
can be the actual physical power, or more often, for convenience with abstract signals, can be
defined as the squared value of the signal, that is, as the actual power dissipated in a load if the signal
were a voltage applied across it. This instantaneous power (the mean or expected value of which is
the average power) is then given by
P(t) = s(t)
2

for a signal s(t).
Since a signal with nonzero average power is not square integrable, the Fourier transforms do not
exist in this case. Fortunately, the WienerKhinchin theorem provides a simple alternative. The PSD
is the Fourier transform of the autocorrelation function, R(), of the signal if the signal is treated as a
wide-sense stationary random process
These results are expressed in the mathematical formula,



Matlab Examples


%% Lab 8.1

%% Initializing
close all
clear all
clc

N = 8; %% number of sample
t = [0:N-1]/N; %% define time
f = sin(2*pi*t); %%define function

q=fft(f)
p = abs(q)/(N/2); %% absolute value of the fft
z=ifft(q)
subplot(3,2,1)
plot(t,f)
xlabel('Time on x axis')
ylabel('Amplitude')
title('Orginal signal')
grid on
subplot(3,2,2)
plot(t,q)
xlabel('Time on x axis')
ylabel('Fourier Transform ')
title('Transformed signal')
grid on
subplot(3,2,3)
plot(t,real(q))
xlabel('Time on x axis')
ylabel('Real Part')
title('Real Part of the Fourier Transform')
grid on
subplot(3,2,4)
plot(t,p)
xlabel('Time on x axis')
ylabel('Imiginary Part')
title('Imiginary Part of the Fourier Transform')
grid on
subplot(3,2,5)
plot(t,z)
xlabel('Time on x axis')
ylabel('Ampltiude')
title('Original Signal')
grid on


Figure 4 Output for Lab 8.1


0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9
-1
-0.5
0
0.5
1
Time on x axis
A
m
p
l
i
t
u
d
e
Orginal signal
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9
-5
0
5
10
x 10
-16
Time on x axis
F
o
u
r
i
e
r

T
r
a
n
s
f
o
r
m

Transformed signal
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9
-5
0
5
10
x 10
-16
Time on x axis
R
e
a
l

P
a
r
t
Real Part of the Fourier Transform
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9
0
0.5
1
Time on x axis
I
m
i
g
i
n
a
r
y

P
a
r
t
Imiginary Part of the Fourier Transform
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9
-1
-0.5
0
0.5
1
Time on x axis
A
m
p
l
t
i
u
d
e
Original Signal


Figure 5 Power spectrum of a 10 Hz sine wave sampled at 10,000 points over 3.4 seconds.



0 2 4 6 8 10 12 14 16 18 20
10
-35
10
-30
10
-25
10
-20
10
-15
10
-10
10
-5
10
0

%% Lab 8.2

%% Initializing
close all
clear all
clc

N = 10000; %% number of points
T = 3.4; %% define time of interval, 3.4 seconds
t = [0:N-1]/N; %% define time
t = t*T; %% define time in seconds
f = sin(2*pi*10*t); %%define function, 10 Hz sine wave
p = abs(fft(f))/(N/2); %% absolute value of the fft
p = p(1:N/2).^2 %% take the power of positve freq. half
freq = [0:N/2-1]/T; %% find the corresponding frequency in Hz
semilogy(freq,p); %% plot on semilog scale
axis([0 20 0 1]); %% zoom in
grid on





Exercise

Q1 Create a MATLAB function that takes a data set and returns the power spectrum. The input
arguments to the function should be the data and the actual time spanned by the data. The function
should return two vectors, the frequency and the power.




Lab 10 Applications of Signal & Systems
Objective:
To design and understand the working of the different parameters of the filter.
Theory:

Designing the Filter
In electronics, computer science and mathematics, a digital filter is a system that performs
mathematical operations on a sampled, discrete-time signal to reduce or enhance certain aspects of
that signal. This is in contrast to the other major type of electronic filter, the analog filter, which is
an electronic circuit operating on continuous- time analog signals. An analog signal may be
processed by a digital filter by first being digitized and represented as a sequence of numbers, then
manipulated mathematically, and then reconstructed as a new analog signal (see digital signal
processing). In an analog filter, the input signal is "directly" manipulated by the circuit.
A digital filter system usually consists of an analog-to-digital converter (to sample the input signal), a
microprocessor (often a specialized digital signal processor), and a digital-to-analog converter.
Software running on the microprocessor can implement the digital filter by performing the necessary
mathematical operations on the numbers received from the ADC. In some high performance
applications, an FPGA or ASIC is used instead of a general purpose microprocessor.
Practice Example
%% Lab 10.1 - design the nth order of the filter
%% Initializing
close all
clear all
clc

f = [0 .2 .22 .5 .52 1]; %Specify f vector
m = [1 1 .4 .4 0 0]; %Specify Amplitude respose
n=10; %Specify Order
[b a]=yulewalk(n,f,m); %Compute Transfer Function
[h w]=freqz(b,a,512); %Determine Filter Response
subplot(2,1,1)
plot(f/2,m,w/(2*pi),abs(h)) %plot amplitude response
grid on
xlabel('Normalized frequency r')
ylabel('Amplitude')
phase=angle(h) %Calculate Phase
phase=unwrap(phase) %Unwrap Phase
phasedeg=phase*180/pi; %Express Phase into Degree
subplot(2,1,2)
plot(w/(2*pi),phasedeg)







Sampling and aliasing
This is the conversion of a continuous signal into a discrete time signal obtained by taking samples
of the continuous time signal at discrete time instant.
Thus, if x
a
(t) is the input to the sampler, the output is, x
a
(nT)=x(n)where t is called sampling
interval. Nyquist sampling theorem defines the minimum sampling rate to avoid aliasing. That is
minimum sampling rate must be twice that of the highest frequency component of the signal.
Aliasing the phenomenon that results in a loss of information when a signal is reconstructed from its
sampled signal. In principle, the analog signal can be reconstructed from the samples, provided that
the sampling rate is sufficiently high to avoid the problem called aliasing.

Practice Examples




% Program # 10.2--the sampling of a cosine wave with different sampling
% period.
% Illustration of the Sampling Process
% in the Time-Domain

%% Initializing
close all
clear all
clc
clf;

t = 0:0.0005:1;
f = 13;
xa = cos(2*pi*f*t);
subplot(2,1,1)
plot(t,xa);grid
xlabel('Time, msec');ylabel('Amplitude');
title('Continuous-time signal x_{a}(t)');
axis([0 1 -1.2 1.2])
subplot(2,1,2);
T = 0.1; % Sampling period
n = 0:T:1;
xs = cos(2*pi*f*n);
k = 0:length(n)-1;
stem(k,xs);grid;
xlabel('Time index n');ylabel('Amplitude');
title('Discrete-time signal x[n]');
axis([0 (length(n)-1) -1.2 1.2])

Power Law transformation
A power law is a special kind of mathematical relationship between two quantities. When the
frequency of an event varies as a power of some attribute of that event (e.g. its size), the frequency is
said to follow a power law. For instance, the number of cities having a certain population size is
found to vary as a power of the size of the population, and hence follows a power law. There is
evidence that the distributions of a wide variety of physical, biological, and man-made phenomena
follow a power law, including the sizes of earthquakes craters on the moon and of solar flares, the
foraging pattern of various species the sizes of activity patterns of neuronal population the
frequencies of words in most languages, frequencies of family names, the sizes of power outages and
wars, and many other quantities.
The power law transformations equation is;


% Program P # 10.3--the sampling of a cosine wave with different sampling
% period and then reconstruct it highlighting the loss of transmittion.
% Illustration of Aliasing Effect in the Time-Domain

%% Initializing
close all
clear all
clc
clf;

t = 0:0.0005:1;
f = 13;
xa = cos(2*pi*f*t);
subplot(2,1,1)
plot(t,xa);grid
xlabel('Time, msec');ylabel('Amplitude');
title('Continuous-time signal x_{a}(t)');
axis([0 1 -1.2 1.2])
subplot(2,1,2);
T = 0.1;f = 13;
n = (0:T:1)';
xs = cos(2*pi*f*n);
t = linspace(-0.5,1.5,500)';
ya = sinc((1/T)*t(:,ones(size(n))) - (1/T)*n(:,ones(size(t)))')*xs;
plot(n,xs,'o',t,ya);grid;
xlabel('Time, msec');ylabel('Amplitude');
title('Reconstructed continuous-time signal y_{a}(t)');
axis([0 1 -1.2 1.2]);

Practice Example

clc;
clear all;
I = imread('4107.ppm');
R =I(:,:,1); % red plane
G =I(:,:,2); % green plane
B =I(:,:,3); % blue plane
%power law transformation on red plane
image_double=im2double(R);
[r c]=size(image_double);
for i=1:r
for j=1:c
R1(i,j)=(1*power(image_double(i,j),0.4));
end
end
%power law transformation on green plane
image_double=im2double(G);
[r c]=size(image_double);
for i=1:r
for j=1:c
G1(i,j)=(1*power(image_double(i,j),0.4));
end
end
%power law transformation on blue plane
image_double=im2double(B);
[r c]=size(image_double);
for i=1:r
for j=1:c
B1(i,j)=(1*power(image_double(i,j),0.4));
end
end
% catenating three planes after transformation
rgbb=cat(3,R1,G1,B1);
% displaying images
figure
imshow(I)
title('Given image')
figure
imshow(rgbb)
title('Image after transformation')
figure
subplot(2,3,1)
imshow(R) %plot red intensity
title('RED intensity image')
subplot(2,3,2)
imshow(G); %plot Green intensity
title('GREEN intensity image')
subplot(2,3,3)
imshow(B); %plot Blue intensity
title('BLUE intensity image')
subplot(2,3,4)
imhist(R) %plot red intensity histogram
title('RED intensity histogram')
subplot(2,3,5)
imhist(G); %plot Green intensity

Exercise

Q1) Design the digital filter of order 10, 12 and 15
Q2) Change the sampling period to a higher level T= 0.2and a lower level T=0.05 from program
#10.2
Q3) Change the sampling rate and find the aliasing effect in program # 10.3.
Q4) Sample any continuous time signal x(t) using impulse train method and reconstruct it back
from an ideal low pass filter from its samples.



histogram
title('GREEN intensity histogram after transformation')
subplot(2,3,6)
imhist(B1); %plot Blue intensity
histogram
title('BLUE intensity histogram after transformation')