Jaipur Engineering college and research centre, Shri Ram ki Nangal, via

Sitapura RIICO Jaipur- 302 022.

DSP LAB MANUAL

Lab Name : Digital Signal Processing

Lab Code : 5EC4-22

Branch : Electronics & Communications Engineering

Department of Electronics & Communication

Jaipur Engineering College and Research Center, Jaipur

Name – Anish Jain

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Index

S. No Contents Page No.

1
Motto of JECRC
2
Vision and Mission of the Institute
3
Vision and Mission of the Department
4
Program Educational Objectives (PEOs)
5
Program Outcomes (POs)
6
Program Specific Outcomes PSO of the Department
7
RTU Syllabus
8
Course Outcomes
9
CO/PO-PSO mapping
10
Content Beyond Syllabus

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Motto of JECRC

TEACH
TRAIN
&
TRANSFORM

FOR

 Contribution towards National Development

 Global Competencies among Students
 Incorporating a Value System
Promotion to use of Technology

Vision of the Institute

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To become a renowned centre of outcome based learning, and work towards academic,
professional, cultural and social enrichment of the lives of individuals and communities.

Mission of the Institute

 Focus on evaluation of learning outcomes and motivate students to inculcate research

aptitude by project based learning.
 Identify, based on informed perception of Indian, regional and global needs, areas of
focus and provide platform to gain knowledge and solutions.
 Offer opportunities for interaction between academia and industry.
 Develop human potential to its fullest extent so that intellectually capable and
imaginatively gifted leaders can emerge in a range of professions.

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Vision of the Department

To contribute to the society through excellence in scientific and technical education, teaching and
research aptitude in Electronics and Communication Engineering to meet the needs of Global
Industry.

Mission of the Department

M1: To equip the students with strong foundation of basic sciences and domain knowledge
of ECE, so that they are able to creatively their knowledge to the solution of problems
arising in their career path.

M2: To induce the habit of lifelong learning to continuously enhance overall performance.

M3: Students are able to communicate their ideas clearly and concisely so that they can
work in team as well as an individual.

M4: To make the students responsive towards the ethical, social, environmental and in
economic context for the society.

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Program Educational Objectives

PEO1 To provide students with the fundamentals of engineering sciences with more
emphasis in Electronics & Communication Engineering by way of analysing
and exploiting electronics & communication challenges.

PEO2 To train students with good scientific and Electronics & Communication
Engineering knowledge so as to comprehend, analyse, design and create
electronics & communication based novel products and solutions for the real
life problems.

PEO3 To inculcate professional and ethical attitude, effective communication skills,

teamwork skills, multidisciplinary approach, entrepreneurial thinking and an
ability to relate Electronics & Communication Engineering with social issues.

PEO4 To provide students with an academic environment aware of excellence,

leadership, written ethical codes and guidelines, and the self-motivated life-
long learning needed for a successful Electronics & Communication
Engineering professional career.

PEO5 To prepare students to excel in electronics & communication based industry

and higher education by educating students in Electronics & Communication
Engineering field along with high moral values and knowledge.

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Program Outcomes

PO1 Engineering Knowledge: Apply the knowledge of mathematics, science,

engineering fundamentals, and electronics & communication engineering
specialization to the solution of complex electronics and communication
engineering problems.

PO2 Problem Analysis: Identify, formulate, research literature, and analyse

complex electronics and communication engineering problems reaching
substantiated conclusions using first principles of mathematics, natural
sciences, and engineering sciences.

PO3 Design/Development of Solutions: Design solutions for complex electronics

and communication engineering problems and design system components or
processes that meet the specified needs with appropriate consideration for the
public health and safety, and the cultural, societal, and environmental
considerations.

PO4 Conduct Investigations of Complex Problems: Use research-based

knowledge and research methods including design of electronics and
communication engineering experiments, analysis and interpretation of data,
and synthesis of the information to provide valid conclusions.

PO5 Modern Tool Usage: Create, select, and apply appropriate techniques,
resources, and modern electronic engineering and IT tools including prediction
and modelling to complex electronics and communication engineering activities
with an understanding of the limitations.

PO6 The Engineer and Society: Apply reasoning informed by the contextual
knowledge to assess societal, health, safety, legal and cultural issues and the

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consequent responsibilities relevant to the professional electronics and

communication engineering practice.

PO7 Environment and Sustainability: Understand the impact of the professional

electronics and communication engineering solutions in societal and
environmental contexts, and demonstrate the knowledge of, and need for
sustainable development.

PO8 Ethics: Apply ethical principles and commit to professional ethics and
responsibilities and norms of the electronics and communication engineering
practice.

PO9 Individual and Team Work: Function effectively as an individual, and as a

member or leader in diverse teams, and in multidisciplinary settings.

PO10 Communication: Communicate effectively on complex electronics and

communication engineering activities with the engineering community and with
society at large, such as, being able to comprehend and write effective reports
and design documentation, make effective presentations, and give and receive
clear instructions.

PO11 Project Management and Finance: Demonstrate knowledge and understanding

of the electronics and communication engineering and management principles
and apply these to one’s own work, as a member and leader in a team, to
manage projects and in multidisciplinary environments.

PO12 Life-Long Learning: Recognize the need for, and have the preparation and
ability to engage in independent and life-long learning in the broadest context
of electronics and communication engineering changes.

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Program Specific Outcomes

PSO1 Ability to develop knowledge for Robotics and its applications.

PSO2 Ability to apply the concepts of IoT for challenges of Real-World.

RTU Syllabus

PROGRAM 1

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AIM: Generation of continuous and discrete elementary signals (Periodic and non periodic)
using mathematical expression.

Software requirement: - SCI LAB

Theory: A wave is a disturbance that transfers energy from one place to another without
requiring any net flow of mass. Waves can be broadly separated into pulses and periodic waves.
A pulse is a single disturbance while a periodic wave is a continually oscillating motion. There is
a close connection between simple harmonic motion and periodic waves; in most periodic waves,
the particles in the medium experience simple harmonic motion.

Waves can also be separated into transverse and longitudinal waves. In a transverse wave, the
motion of the particles of the medium is at right angles (i.e., transverse) to the direction the wave
moves. In a longitudinal wave, such as a sound wave, the particles oscillate along the direction of
motion of the wave.

Surface waves, such as water waves, are generally a combination of a transverse and a
longitudinal wave. The particles on the surface of the water travel in circular paths as a wave
moves across the surface.

Periodic waves

A periodic wave generally follows a sine wave pattern, as shown in the diagram.

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PROGRAM CODE

//continous time cosine signal //discrete time cosine signal

t=-5:0.01:5; n = -20:.01:20;
y1=cos(t); y4=cos(n);
subplot(3,2,1); subplot(3,2,2);
plot2d3(t,y); plot2d3(n,y4);
xlabel('Time'); xlabel('Time');
ylabel('Magnitude'); ylabel('Magnitude');
title('continous time cosine signal'); title('discrete time cosine signal');

//continous time sine signal //discrete time sine signal

y2=sin(t); y5=sin(n);
subplot(3,2,3); subplot(3,2,4);
plot2d3(t,y2); plot2d3(n,y5);
xlabel('Time'); xlabel('Time');
ylabel('Magnitude'); ylabel('Magnitude');
title('continous time sine signal'); title('discrete time sine signal');

//continous time aperiodic signal //discrete time aperiodic signal

y3=sin(2*F*t*pi).*t; y6=cos(2*f*n*pi).*n;
subplot(3,2,5); subplot(3,2,6);
plot2d3(t,y3); plot2d3(n,y6);
xlabel('Time'); xlabel('Time');
ylabel('Magnitude'); ylabel('Magnitude');
title('continous time aperiodic signal'); title('discrete time aperiodic signal');

Figure (Expected output):

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Figure 2.2 Continuous and Discrete Time Elementary Signals

VIVA VOICE QUESTION:

Q1.What is Signal?

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Q2.What are periodic signal.

Q3.What is time period.

Q4.Unit step is periodic signal or not.

Q5. u(t)+u(-t) is periodic or not.

Q6. Exponential signal is periodic or not.

Q7. Sum of two periodic signal is always periodic signal. True or false?

PROGRAM 2

AIM: Generation of Continuous and Discrete Unit Step Signal.

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Software requirement: - SCI LAB

Unit step: A signal with magnitude one for time greater than zero. We can assume it as a dc
signal which got switched on at time equal to zero.

Unit step function is denoted by u (t). It is defined as u (t) =

Figure 2.3Continuous Time Unit Step Signal

 It is used as best test signal.

 Area under unit step function is unity

PROGRAM CODE

t=-10:1:10; n=-10:1:10;
y1=[zeros(1,10),ones(1,11)]; y7=[zeros(1,10),ones(1,11)];
subplot(4,3,1); subplot(4,3,7);
plot2d3(t,y1); plot2d3(n,y7);
xlabel('Time'); xlabel('Time');
ylabel('Magnitude'); ylabel('Magnitude');
title('Continuous Unit Step Function'); title('discrete Unit Step u(n) Function');

y2=[zeros(1,16),ones(1,5)]; y8=[zeros(1,16),ones(1,5)];
subplot(4,3,2); subplot(4,3,8);
plot2d3(t,y2); plot2d3(n,y8);
xlabel('Time'); xlabel('Time');
ylabel('Magnitude'); ylabel('Magnitude');

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title('Continuous Unit Step u(t-6) Function'); title('discrete Unit Step u(n-6) Function');

y3=[zeros(1,4),ones(1,17)]; y9=[zeros(1,4),ones(1,17)];
subplot(4,3,3); subplot(4,3,9);
plot2d3(t,y3); plot2d3(n,y9);
xlabel('Time'); xlabel('Time');
ylabel('Magnitude'); ylabel('Magnitude');
title('Continuous Unit Step u(t+6) Function'); title('discrete Unit Step u(n+6) Function');

y4=[zeros(1,6),ones(1,15)]; y10=[zeros(1,6),ones(1,15)];
subplot(4,3,4); subplot(4,3,10);
plot2d3(t,y4); plot2d3(n,y10);
xlabel('Time'); xlabel('Time');
ylabel('Magnitude'); ylabel('Magnitude');
title('Continuous Unit Step u(-t+4) Function'); title('discrete Unit Step u(-n+4) Function');

y5 = y3 - y2; y11 = y9 - y8;

subplot(4,3,5); subplot(4,3,11);
plot2d3(t,y5); plot2d3(n,y11);
xlabel('Time'); xlabel('Time');
ylabel('Magnitude'); ylabel('Magnitude');
title('Continuous Unit Step u(t+6)-u(t-6) Function'); title('discrete Unit Step u(n+6)-u(n-6) Function');

y6 = y3 + y2; y12 = y9 + y8;

subplot(4,3,6); subplot(4,3,12);
plot2d3(t,y6); plot2d3(n,y12);
xlabel('Time'); xlabel('Time');
ylabel('Magnitude'); ylabel('Magnitude');
title('Continuous Unit Step u(t+6)+u(t-6) title('discrete Unit Step u(n+6)+u(n-6) Function');
Function');

Figure (Expected output):

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Figure 2.4 Continuous and Discrete Unit Step Signals

VIVA VOICE QUESTION:

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Q1.What is Signal?

Q2.What is unit step signal?

Q3.Differention of step signal is?

Q4. Integration of step signal is?

Q5. What is continues time signals?

Q6. What are discrete time signals?

Q7. What is Analog signal?

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PROGRAM 3

AIM: Adding and subtracting two given signals (continuous as well as discrete signals).

Software requirement: - SCI LAB

PROGRAM CODE

clc; //discrete signal

//continuous signal n=0:1:30;
t = 0:0.001:30; a=n;
y1 = t; b=2*n;
y2 = 2*t; //First Signal
//First Signal subplot(4,2,5);
subplot(4,2,1); plot2d3(n,a);
plot2d3(t,y1); xlabel('time');
xlabel('time'); ylabel('magnitude');
ylabel('magnitude'); title('First Signal');
title('First Signal'); //Second Signal
//Second Signal subplot(4,2,6);
subplot(4,2,2); plot2d3(n,b);
plot2d3(t,y2); xlabel('time');
xlabel('time'); ylabel('magnitude');
ylabel('magnitude'); title('Second Signal');
title('Second Signal'); c=a+b;
y3 = y1+y2; d=a-b;
y4 = y1-y2; //Addition
//Addition subplot(4,2,7);
subplot(4,2,3); plot2d3(n,c);
plot2d3(t,y3); xlabel('time');
xlabel('time'); ylabel('magnitude');
ylabel('magnitude'); title('Addition');
title('Addition'); //subtraction
//subtraction subplot(4,2,8);
subplot(4,2,4); plot2d3(n,d);
plot2d3(t,y4); xlabel('time');
xlabel('time'); ylabel('magnitude');
ylabel('magnitude'); title('Subtraction');
title('Subtraction');

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Figure (Expected output):

Figure 3.1 Continuous and Discrete Signals with addition and subtraction

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VIVA VOICE QUESTION:

Q1.Addition of two periodic signals is periodic or not?

Q2.Addition of u (t) and u (-t) is …………….signals.

Q3.Subtraction of u (t) and u (-t) is …………….signals.

Q4. Addition of two even signals is even or odd?

Q5. Addition of two odd signals is even or odd?

Q6.Addition of even and odd signals is ?

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PROGRAM 4

AIM: Continuous and Discrete Time Convolution (Using Basic Definition).

Software requirement: - SCI LAB

Theory: A convolution is an integral that expresses the amount of overlap of one function as it is
shifted over another function. It therefore "blends" one function with another. For example, in
synthesis imaging, the measured dirty map is a convolution of the "true" CLEAN map with the
dirty beam (the FOURIER TRANSFORM of the sampling distribution). The convolution is
sometimes also known by its German name, faltung ("folding").

Abstractly, a convolution is defined as a product of functions and that are objects in the
algebra of SCHWARTZ FUNCTIONS in. Convolution of two functions and over a finite
range is given by

Where the symbol denotes convolution of and .

Convolution is more often taken over an infinite range,

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PROGRAM CODE

(4A) (4B) (4C)

x=input('enter first sequence'); x = input('enter the first seq');
b1=input('enter the lower limit'); N1 = length(x);
u1=input('enter the upper limit'); t1=-5:1:0 n1 = 0:1:N1-1;
x1=b1:1:u1; t2=0:1:2; subplot(2,2,1);
h=input('enter second sequence'); t3=2:1:5; plot2d3(n1,x);
b2=input('enter the lower limit'); xlabel('time');
u2=input('enter the upper limit'); h1=zeros(size(t1)); ylabel('mag');
h1=b2:1:u2; h2=ones(size(t2)); title('seq of x');
b=b1+b2; h3=zeros(size(t3));
u=u1+u2; h = input('enter the second
a=b:1:u; t=[t1 t2 t3]; seq');
m=length(x); h=[h1 h2 h3]; N2 = length(h);
n=length(h); subplot(3,1,1); n2 = 0:1:N2-1;
X=[x,zeros(1,n)]; plot2d3(t,h); subplot(2,2,2);
subplot(2,2,1); xlabel('time'); plot2d3(n2,h);
disp('x(n) is:'); ylabel('magnitude'); xlabel('time');
disp(x); ylabel('mag');
plot2d3(x1,x); a1=-5:1:0; title('seq of h');
xlabel('n'); a2=0:1:4;
ylabel('x(n)'); a3=4:1:6; y=conv(x,h);
title('first squence'); n = 0:1:N1+N2-2;
grid on; x1=zeros(size(a1)); subplot(2,2,[3 4]);
H=[h,zeros(1,m)]; x2=ones(size(a2)); plot2d3(n,y);
subplot(2,2,2); x3=zeros(size(a3)); xlabel('time');
disp('h(n) is;'); ylabel('mag');
disp(h); a=[a1 a2 a3]; title('convolution');
plot2d3(h1,h); x=[x1 x2 x3];
xlabel('n'); subplot(3,1,2);
ylabel('h(n)'); plot2d3(a,x);
title('second sequence'); xlabel('time');
grid on; ylabel('magnitude');
for i=1:n+m-1
Y(i)=0;
for j=1:m; c=conv(x2,h2);
if((i-j+1)>0) subplot(3,1,3);
Y(i)=Y(i)+(X(j)*H(i-j+1)); plot2d3(c);
else xlabel('time');
end ylabel('magnitude');
end
end
subplot(2,2,[3 4]);
disp('y(n) is:');
disp(Y);
plot2d3(a,Y);
xlabel('n');
ylabel('Y(n)');
title('output sequence');
grid on;

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Figure (Expected output):

Figure 4(a):

Figure 4.1 Discrete Time Sequence along with Convolution

Input & Output
enter first sequence:=[-2 -4 5 6]
enter the lower limit:=-1
enter the upper limit:=2
enter second sequence:=[3 4 5]
enter the lower limit:=0
enter the upper limit:=2
x(n) is:
-2 -4 5 6

h(n) is;
3 4 5

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y(n) is:
-6 -20 -11 18 49 30 38 24

Figure 4(b):

Figure 4.2Continuous Signals along with Convolution

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Figure 4(c):

Input:
enter the first seq:=[0 1 2 3]
enter the second seq:=[2 3 4]

Output
y(n)= [0 2 7 16 17 12]

Figure 4.3 Discrete Signals along with Convolution

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VIVA VOICE QUESTION:

Q1.What is the importance of linear and circular convolution in signals and systems?

Q2. Write the properties of linear convolution.

Q3.What is formula for continuous time convolution signal?

Q4. What is formula for discrete time convolution signal?

Q5. What is meaning of time shifting.

Q6.What is impulse response? Explain its significance.

Q7. Which command we use in SCI lab for performing convolution.

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PROGRAM 5

AIM: Checking Linearity and Time variance property of a system using convolution, shifting.

Software requirement: - SCI LAB

Theory:
Convolution is a mathematical operation used to express the relation between input and output of
an LTI system. It relates input, output and impulse response of an LTI system as

y(n)=x(n)∗h(n)

Where y (n) = output of LTI

x (n) = input of LTI

h (n) = impulse response of LTI

Discrete Convolution y(n)=x(n)∗h(n) = By using convolution we can find zero state response of
the system.

PROGRAM CODE

clc; n2=0:M-1;
//Linearity and Time variance subplot(312);
x1=input('Enter the first sequence x1(n) = '); stem(n2,x2);
x2=input('Enter the second sequence x2(n) = '); grid on;
L=length(x1); xlabel('n2--->');
M=length(x2); ylabel('amplitude--->');
N=L+M-1; yn=conv(x1,x2);
disp(‘The values of yn are= ‘);
disp(yn);
title('Second sequence');
n3=0:N-1; subplot(313);
n1=0:L-1; stem(n3,yn);
subplot(311); grid on; xlabel('n3--->'); ylabel('amplitude---
stem(n1,x1); >');
grid on; title('Convolved output');
xlabel('n1--->');
ylabel('amplitude--->');

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title('First sequence');

n2=0:M-1;
Enter the first sequence x1(n) = [1 2 3 4 5]
Enter the second sequence x2(n) = [5 8 3 5 4 6]

The values of yn are= 5 18 34 55 80 81 59 59 44 30

Figure (Expected output):

Figure 5.1 Continuous and Discrete Signals with addition and subtraction

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VIVA VOICE QUESTION:

Q1. Explain the significance of convolution.
Q2. Define linear convolution.
Q3. Why linear convolution is called as a periodic convolution?
Q4. Why zero padding is used in linear convolution?
Q5. What are the four steps to find linear convolution?
Q6. What is the length of the resultant sequence in linear convolution?
Q7. How linear convolution will be used in calculation of LTI system response?
Q8. List few applications of linear convolution in LTI system design.
Q9. Give the properties of linear convolution.
Q10. How the linear convolution will be used to calculate the DFT of a signal?

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PROGRAM 6

AIM: To generate and verify random sequences with arbitrary distributions, means and
variances for: (a) Rayleigh
distribution (b) Normal
distributions: N(0,1) (c) Poisson
distributions: N(m, x)

Software requirement: - SCI LAB

Theory:

(a)The Rayleigh distribution- it is a special case of the WEIBULL DISTRIBUTION. If A and

B are the parameters of the Weibull distribution, then the Rayleigh distribution with parameter b
is equivalent to the Weibull distribution with parameters A=21/2 b and B = 2.

If the component velocities of a particle in the x and y directions are two independent normal
random variables with zero means and equal variances, then the distance the particle travels per
unit time is distributed Rayleigh.

In communications theory NAKAGAMI DISTRIBUTION, RICIAN DISTRIBUTION, and

Rayleigh distributions are used to model scattered signals that reach a receiver by multiple paths.
Depending on the density of the scatter, the signal will display different fading characteristics.
Rayleigh and Nakagami distributions are used to model dense scatters, while Rician distributions
model fading with a stronger line-of-sight. Nakagami distributions can be reduced to Rayleigh
distributions, but give more control over the extent of the fading.

(b) The Normal distributions-

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Figure 6.1 Normal Probability Distribution Graph

A normal distribution in a VARIATE with MEAN and VARIANCE is a statistic
distribution with probability density function

(1)

on the domain . While statisticians and mathematicians uniformly use the term
"normal distribution" for this distribution, physicists sometimes call it a Gaussian distribution
and, because of its curved flaring shape, social scientists refer to it as the "bell curve."

The so-called "standard normal distribution" is given by taking and in a general

normal distribution. An arbitrary normal distribution can be converted to a standard normal
distribution by changing variables to , so , yielding

(c) The Poisson distributions-

A Poisson random variable is the number of successes that result from a Poisson experiment. The
probability distribution of a Poisson random variable is called a Poisson distribution.

Given the mean number of successes (μ) that occur in a specified region, we can compute the Poisson
probability based on the following formula:

Poisson Formula. Suppose we conduct a Poisson experiment, in which the average number of successes
within a given region is μ. Then, the Poisson probability is:

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P(x; μ) = (e-μ) (μx) / x!

Where x is the actual number of successes that result from the experiment and e is approximately equal
to 2.71828.

The Poisson distribution has the following properties:

 The mean of the distribution is equal to μ .

 The variance is also equal to σ.

PROGRAM CODE

//normal distribution //rayleigh distribution //poission distribution

x = -5:0.01:5; x = -5:1:15; x = -5:1:15;
y1 = (normpdf(x,0,1)); y1 = (poisspdf(x,4)); y1 = (raylpdf(x,4));
y2 = (normpdf(x,0.1,2)); y2 = (poisspdf(x,2)); y2 = (raylpdf(x,2));
y3 = (normpdf(x,0,0.5)); y3 = (poisspdf(x,1)); y3 = (raylpdf(x,1));
subplot(3,1,1); subplot(3,1,2); subplot(3,1,3);
plot2d3(x,y1,'.',x,y2,'-',x,y3,'*') plot2d3(x,y1,'.',x,y2,'-',x,y3,'*') plot2d3(x,y1,'.',x,y2,'-',x,y3,'*')
; ; ;
xlabel('value of x'); xlabel('value of x'); xlabel('value of x');
ylabel('value of y'); ylabel('value of y'); ylabel('value of y');
title('normal distribution'); title('rayleigh distribution'); title('poission distribution');

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Figure (Expected output):

Figure 6.2 Graphs of Various Distribution Function in Continuous Time Domain

VIVA VOICE QUESTION:

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Q1.What is pdf?

Q2.What is cdf.

Q3.What are the Mean and variance of Poisson distribution?

Q4. What are the Mean and variance of Normal distribution?

Q5.Which command we use in scilab to plot pdf of normal distribution.

Q6.Normal distribution is continuous or discrete?

Q7.If value of probability is very small and number is large than which distribution is used?

PROGRAM 7

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AIM: To find DFT / IDFT of given DT signal.

Software requirement: - SCI LAB

Theory: Basic equation to find the DFT of a sequence is given below.

Basic equation to find the IDFT of a sequence is given below.

PROGRAM CODE

//pdf
xn=input('Enter the sequence x(n)'); % Find the phases of individual DFT points %
ln=length(xn); plot the magnitude sequence
xk=zeros(1,ln); t=0:ln-1;
xk=zeros(1,ln); subplot(223);
stem(t,phase);
%DFT of sequence ylabel ('Phase');
xlabel ('K');
for k=0:ln-1 title ('Phase Response');
for n=0:ln-1
xk(k+1)=xk(k+1)+ %IDFT of the sequence
(xn(n+1)*exp((i)*2*pi*k*n/ln)); for n=0:ln-1
end for k=0:ln-1 ixk(n+1)=ixk(n+1)+
end (xk(k+1)*exp(i*2*pi*k*n/ln));
%Plotting input sequence end
end
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t=0:ln-1; ixk=ixk./ln;
subplot(221); %code block to plot the input sequence t=0:ln-
stem(t,xn); 1;
ylabel ('Amplitude'); subplot(224);
xlabel ('Time Index'); stem(t,ixk);
title('Input Sequence'); ylabel ('Amplitude');
magnitude=abs(xk); xlabel ('Time Index');
title ('IDFT sequence');
% plot the magnitude response
t=0:ln-1; subplot(222);
stem(t,magnitude);
ylabel ('Amplitude');
xlabel ('K');
title('Magnitude Response');subplot(2,1,2);
phase=angle(xk);

Output:

Xn=[1 2 3 4 5]

Xk = 15,-2.50+3.44i,-2.50+0.81i,-2.49-0.81i,-2.49-3.44i

Figure (Expected output):

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Figure 7.1DFT and IDFT Sequnce

VIVA VOICE QUESTION:

Q1. Define signal, Give Examples for 1-D, 2-D, 3-D signals.

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Q2. Define transform. What is the need for transform?

Q3. Differentiate Fourier transform and discrete Fourier transform.

Q4. Differentiate DFT and DTFT

Q5. Explain mathematical formula for calculation of DFT.

Q6. Explain mathematical formula for calculation of IDFT.

Q7. How to calculate FT for 1-D signal?

Q8. What is meant by magnitude plot, phase plot, power spectrum?

Q9. Explain the applications of DFT.

Q10. What are separable transforms?

PROGRAM 8

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AIM: N-point FFT algorithm.

Software requirement: - SCI LAB

Theory: DFT of a sequence

Where N= Length of sequence. K= Frequency Coefficient. n = Samples in time domain.

FFT: -Fast Fourier transform. There are two methods.

1. Decimation in time (DIT ) FFT.

2. Decimation in Frequency (DIF) FFT.

Why we need FFT?

The no of multiplications in DFT = N2.

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

For FFT

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

The no of additions = N log2 N.

PROGRAM CODE

close all; xlabel('n---->');

x=input('Enter the sequence : ') ylabel('magnitude');
N=length(x) title('magnitude response');
xK=fft(x,N) subplot (2,2,3);
xn=ifft(xK) stem(n,angle(xK));
n=0:N-1; xlabel('n---->');
subplot (2,2,1); ylabel('phase');

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stem(n,x); title('Phase responce');

xlabel('n---->'); subplot (2,2,4);
ylabel('amplitude'); stem(n,xn);
title('input sequence'); xlabel('n---->');
subplot (2,2,2); ylabel('amplitude');
stem(n,abs(xK));
xlabel('n---->');
ylabel('magnitude');
title('magnitude response');
subplot (2,2,3);
stem(n,angle(xK));
xlabel('n---->');
ylabel('phase');

OUTPUT

Enter the sequence: [1 2 3 4 5]

x=12345N=5

xK = 15.0000, -2.5000 + 3.4410i, -2.5000 + 0.8123i , -2.5000 - 0.8123i, -2.5000 - 3.4410i

xn = 1 2 3 4 5

Figure (Expected output):

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Figure 8.1 DFT Sequence

VIVA VOICE QUESTION:

Q1.Define transform. What is the need for transform?
Q 2. Differentiate Fourier transform and discrete Fourier transform.
Q3. Differentiate DFT and DTFT.
Q4. What are the advantages of FFT over DFT?
Q 5. Differentiate DITFFT and DIFFFT algorithms.
Q6. What is meant by radix?
Q7. What is meant by twiddle factor and give its properties?
Q8. How FFT is useful to represent a signal?
Q 9. Compare FFT and DFT with respect to number of calculation required?
Q 10. How the original signal is reconstructed from the FFT of a signal?
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PROGRAM 9

AIM: To implement Circular convolution.

Software requirement: - SCI LAB

PROGRAM CODE

Clc;
xl = input ('Enter the First Sequence to be xlabel('Time');
convoluted:'); ylabel('Amplitude');
11 = length(x1); title('First Sequence');
x2 = input ('Enter the Second Sequence to be subplot(312);
convoluted:'); stem(x2);
12 = length(x2); xlabel('Time');
13 = 11 + 12 -1; ylabel('Amplitude');
xl =[x1 ,zeros( 1,13-11)]; title('First Sequence');
x2= [x2,zeros(1,13-l2)]; subplot(313);
disp('The Input Squence:'); stem(f);
x1,x2 f = cconv(x1,x2,13); xlabel('Time');
disp('The Circular Convoluted Sequence is:'); ylabel('Amplitude');
disp(f); title('Circular Convoluted sequence');
subplot(311);
stem(x1);

OUTPUT

xl = 1 2 0 0
x2 = 1 2 4 0

The Circular Convoluted Sequence is: 1 488

Figure (Expected output):

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Figure 9.1 Circular Convolution

VIVA VOICE QUESTION:

Q1.Define transform. What is the need for transform?

Q 2. Differentiate Fourier transform and discrete Fourier transform.

Q3. Differentiate DFT and DTFT.

Q4. What are the advantages of FFT over DFT?

Q 5. Differentiate DITFFT and DIFFFT algorithms.

Q6. What is meant by radix?

Q7. What is meant by twiddle factor and give its properties?

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Q8. How FFT is useful to represent a signal?

Q 9. Compare FFT and DFT with respect to number of calculation required?

Q 10. How the original signal is reconstructed from the FFT of a signal?

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PROGRAM 10

AIM: MATLAB code for implementing z-transform.

Software requirement: - SCI LAB

PROGRAM CODE

Clc;
%Compute the Z-transform of sin(n)
syms n
f = sin(n);
ztrans(f)

ans = (z*sin(1))/(z^2 - 2*cos(1)*z + 1)

% Compute the Z-transform of exp(m+n)

syms m n
f = exp(m+n);
ztrans(f)

ans = (z*exp(m))/(z - exp(1))

%Compute the Z-transform of symbolic

functions
syms f1(x) f2(x) a b
f1(x) = exp(x);
f2(x) = x;
ztrans([f1 f2],x,[a b])

ans = [ a/(a - exp(1)), b/(b - 1)^2]

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PROGRAM 11

AIM: MATLAB program to find frequency response of analog LP/HP filters.

Software requirement: - SCI LAB

Theory:

1. Butterworth filter

Syntax [n,Wn] = buttord(Wp,Ws,Rp,Rs)

Description
calculates the minimum order of a digital or analog Butterworth filter required to meet a set of
filter design

Parameters Description
Wp Passband corner frequency
Ws Stopband corner frequency
Rp Passband ripple, in decibels
Rs Stopband attenuation, in decibels
n the lowest order the proposed Butterworth filter
Wn corresponding cut-off frequency

2. Butter

Syntax

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[b,a] = butter(n,Wn)

[b,a] = butter(n,Wn,'ftype’)

[b,a] = butter(n,Wn,'s’)

Description : Designs lowpass, bandpass, highpass, and bandstop digital and analog
Butterworth filters. Butterworth filters are characterized by a magnitude response that is
maximally flat in the passband and monotonic overall.

PROGRAM CODE

clc;
clear all; clc; clear all; close all;
close all; format long;
format long; % Program fopr Butterworth HPF
% Program for Butterwoth LPF wp=input('Please enter the first pass band
wp=input('Please enter the first pass band frequency (Hz): ');
frequency (Hz): '); ws=input('Please enter the first stop band
ws=input('Please enter the first stop band frequency (Hz): ');
frequency (Hz): '); rp=input('Please enter the pass band ripple: ');
rp=input('Please enter the pass band ripple: '); rs=input('Please enter the stop band ripple: ');
rs=input('Please enter the stop band ripple: '); fs=input('Please enter the sampling frequency:
fs=input('Please enter the sampling frequency: (Hz)');
(Hz)'); w1=2*wp/fs;
w1=2*wp/fs; w2=2*ws/fs;
w2=2*ws/fs; [n,wn]=buttord(w1,w2,rp,rs);
[n,wn]=buttord(w1,w2,rp,rs); [b,a]=butter(n,wn,'high');
[b,a]=butter(n,wn) w=0:0.01:pi;
w=0:0.01:pi; [h,om]=freqz(b,a,w);
[h,om]=freqz(b,a,w); m=20*log10(abs(h));
m=20*log10(abs(h)); an=angle(h);
an=angle(h); subplot(2,1,1); plot(om/pi,m);
subplot(2,1,1); plot(om/pi,m); ylim([-300 50]);
ylim([-300 50]); ylabel('gain in dB');
ylabel('gain in dB'); xlabel('Normalised frequency');

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xlabel('Normalised frequency'); subplot(2,1,2);plot(om/pi,an); ylabel('Phase in

subplot(2,1,2);plot(om/pi,an); Radian’); xlabel('Normalised frequency');
ylabel('Phase in Radian');
xlabel('Normalised frequency');

Figure (Expected output for Butterworth LPF):

Figure 11.1 Butterworth Low Pass Filter

Figure (Expected output for Butterworth HPF):

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Figure 11.2 Butterworth High Pass Filter

VIVA VOICE QUESTION:

Q1List some advantages of digital filters over analog filters.

Q2. Write some differences between FIR and IIR filters.

Q3. What are the different methods to design IIR filters?

Q4. Why IIR filters are not reliable?

Q5. What are different applications of IIR filters?

Q 6. What are advantages of IIR filters?

FIR FILTER DESIGN An FIR filter is designed by finding the coefficients and filter order that
meet certain specifications, which can be in the time-domain (e.g. a matched filter) and/or the
frequency domain (most common). Matched filters perform a cross-correlation between the input
signal and a known pulse-shape. The FIR convolution is a cross-correlation between the input
signal and a time-reversed copy of the impulse-response. Therefore, the matched-filter's impulse
response is "designed" by sampling the known pulse-shape and using those samples in reverse

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order as the coefficients of the filter. When a particular frequency response is desired, several
different design methods are common: Q7. What are disadvantages of IIR filters?

Q 8. Differentiate Butterworth and Chebyshev approximations.

Q9. What is meant by impulse response?

Q 10. Difference between IIR low pass and High pass filters.

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PROGRAM 12

AIM: MATLAB program to find frequency response of analog LP/HP filters.

Software requirement: - SCI LAB

Theory:

FIR filters are digital filters with finite impulse response. They are also known as non-recursive
digital filters as they do not have the feedback. An FIR filter has two important advantages over
an IIR design:

Firstly, there is no feedback loop in the structure of an FIR filter. Due to not having a feedback
loop, an FIR filter is inherently stable. Meanwhile, for an IIR filter, we need to check the
stability. Secondly, an FIR filter can provide a linear-phase response. As a matter of fact, a
linear-phase response is the main advantage of an FIR filter over an IIR design otherwise, for the
same filtering specifications; an IIR filter will lead to a lower order.

1. Window design method

2. Frequency Sampling method
3. Weighted least squares design

PROGRAM CODE

clc;
clear all; %Band Pass filter
close all; wn=[wp ws];
% Program code for FIR Butterworth filter b=fir1(n,wn, ‘band', y);
using Rectangular window [h,o]=freqz(b,1,256);
fp=input('Please enter the pass band frequency m=20*log10(abs(h));

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(Hz): '); subplot(2,2,3); plot(o/pi,m);

fs=input('Please enter the stop band frequency ylim([-120 20]);
(Hz): '); ylabel('Gain in dB');
rp=input('Please enter the pass band ripple: '); xlabel('Normalised frequency');
rs=input('Please enter the stop band ripple: '); title('Band pass filter');
sf=input('Please enter the sampling frequency:
(Hz)'); % Band stop filter
wp=2*fp/sf; b=fir1(n,wn,'stop',y);
ws=2*fs/sf; [h,o]=freqz(b,1,256);
num=-20*log10(sqrt(rp*rs))-13; m=20*log10(abs(h));
dem=14.6*(fs-fp)/sf; subplot(2,2,4); plot(o/pi,m);
n=ceil(num/dem); ylim([-15 5]);
n1=n+1; ylabel('Gain in dB');
if(rem(n,2)~=0) xlabel('Normalised frequency');
n1=n; title('Stop pass filter');
n=n-1;
end
y=rectwin(n1)

%Low pass filter

b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1); plot(o/pi,m);
ylim([-120 20]);
ylabel('Gain in dB');
xlabel('Normalised frequency');
title('Low pass filter');
%High pass filter
b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,2); plot(o/pi,m);
ylim([-100 20]);
ylabel('Gain in dB');
xlabel('Normalised frequency');
title('High pass filter');

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Figure (Expected output):

Figure 12.1 Low Pass and High Pass Filter

PROGRAM CODE

clc;
clear all; %Band Pass filter
close all; wn=[wp ws];
fp=input('Please enter thepass band frequency b=fir1(n,wn, ‘band', y);
(Hz): '); [h,o]=freqz(b,1,256);
fs=input('Please enter the stop band frequency m=20*log10(abs(h));
(Hz): '); subplot(2,2,3); plot(o/pi,m);

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rp=input('Please enter the pass band ripple: '); ylim([-120 20]);

rs=input('Please enter the stop band ripple: '); ylabel('Gain in dB');
sf=input('Please enter the sampling frequency: xlabel('Normalised frequency');
(Hz)'); title('Band pass filter');
beta=input('Please enter the value of beta'); % Band stop filter
wp=2*fp/sf; b=fir1(n,wn,'stop',y);
ws=2*fs/sf; [h,o]=freqz(b,1,256);
num=-20*log10(sqrt(rp*rs))-13; m=20*log10(abs(h));
dem=14.6*(fs-fp)/sf; subplot(2,2,4); plot(o/pi,m);
n=ceil(num/dem); ylim([-15 5]);
n1=n+1; ylabel('Gain in dB');
if(rem(n,2)~=0) xlabel('Normalised frequency');
n1=n; title('Stop pass filter');
n=n-1;
end
y=kaiser(n1,beta);

%Low pass filter

b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1); plot(o/pi,m);
ylim([-120 20]);
ylabel('Gain in dB');
xlabel('Normalised frequency');
title('Low pass filter');
%High pass filter
b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,2); plot(o/pi,m);
ylim([-100 20]);
ylabel('Gain in dB');
xlabel('Normalised frequency');
title('High pass filter');

Figure (Expected output):

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Figure 12.2 High Pass Filter

VIVA VOICE QUESTION:
Q1. Define filter.
Q 2. What are the different types of filters?
Q3. Why are FIR filters generally preferred over IIR filters in multirate (decimating and
interpolating) systems/
Q4. Difference between IIR and FIR filters?
Q5. Differentiate ideal filter and practical filter responses.
Q6. What is the filter specifications required to design the analog filters?
Q7. What is meant by frequency response of filter?
Q8. What is meant by magnitude response?
Q9. What is meant by phase response?
10. Difference between FIR low pass filter and high pass filter.

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