LAB MANUAL

Lab Name Lab Code Branch Year

: : : :

FPGA LAB 8CS6 Computer Engineering 4th Year

Swami Keshvanand Institute of Technology, Management & Gramothan, Jaipur Department of Computer Science & Engineering (Rajasthan Technical University, KOTA)
1

INSTRUCTIONS OF LAB

DOS

1. Student should get the record of previous experiment checked before starting the new experiment. 2. Read the manual carefully before starting the experiment. 3. Before starting the experiment, get circuit diagram checked by the teacher.

4. Before switching on the power supply, get the circuit connections checked. 5. Get your readings checked by the teacher. 6. Apparatus must be handled carefully. 7. Maintain strict discipline. 8. Keep your mobile phone switched off or in vibration mode. 9. Students should get the experiment allotted for next turn, before leaving

the lab.

DONTS

1. Do not touch or attempt to touch the mains power supply wire with bare hands. 2. Do not overcrowd the tables. 3. Do not tamper with equipments. 4. Do not leave the lab without permission from the teacher.

Jitendra Gautam

INSTRUCTIONS TO THE STUDENTS GENERAL INSTRUCTIONS Maintain separate observation copy for each laboratory. Observations or readings should be taken only in the observation copy. Get the readings counter signed by the faculty after the completion of the experiment. Maintain Index column in the observation copy and get the signature of the faculty before leaving the lab. BEFORE ENTERING THE LAB The previous experiment should have been written in the practical file, without which the students will not be allowed to enter the lab. The students should have written the experiment in the observation copy that they are supposed to perform in the lab. The experiment written in the observation copy should have aim, apparatus required, circuit diagram/algorithm, blank observation table (if any), formula (if any), programme (if any), model graph (if any) and space for result. WHEN WORKING IN THE LAB Necessary equipments/apparatus should be taken only from the lab assistant by making an issuing slip. Never switch on the power supply before the permission from the faculty.

BEFORE LEAVING THE LAB

3

The equipments/components should be returned back to the lab assistant in good condition after the completion of the experiment. The students should get the signature from the faculty in the observation copy.

They should also check whether their file is checked and counter signed in the index.

INDEX
S.NO 1 2. 3. 4. Exp:- 1 SYLLABUS PROGRAM EDUCATIONAL OBJECTIVES INTRODUCTION THEORY OF LAB LIST OF EXPERIMENTS (RTU SYLLABUS) Objectives :- Fundamental Theory Introduction to DSP architectures and programming. Sampling Theory, Analog-to-Digital Converter (ADC), Digital-to- Analog Converter (DAC), and Quantization; Decimation, Interpolation, Convolution, Simple Moving Average; Periodic Signals and harmonics; Fourier Transform (DFT/FFT), Spectral Analysis, and time/spectrum representations; FIR and IIR Filters; Exp:- 1.a OBJECTIVE: Plotting of various elementary signals like impulse function, unit step function, ramp function, quadratic function, sine wave,and a general sinusoidal function. Exp:- 1.b Exp:- 1.c Exp:- 1.d OBJECTIVE: Verification of sampling theorem. OBJECTIVE: Derive the Impulse response of a given system.. OBJECTIVE: Perform linear convolution of two given sequences. 36 40 43 30 CONTENTS PAGE NO. 8 9 22

Exp:- 1.e Exp:- 1.f

OBJECTIVE: Perform Circular convolution of two given sequences. OBJECTIVE: Computation of N-point DFT of a given sequence and to plot magnitude and phase spectrum.

46 48

Exp:- 1.g Exp:- 1.h

OBJECTIVE: Computation of Linear Convolution using DFT & IDFT. OBJECTIVE: Down sampling or decimation of the sum of two sinusoids by a factor M.

50 53

Exp:- 1.i

OBJECTIVE: . To study the design and implementation of FIR filter to meet given specifications

56

Exp:- 1.j

OBJECTIVE: To study the design and implementation of IIR(LPF/HPF) filter to meet given specifications.

65

Sample Viva Question Exp:2 Objectives :- Design (Simulation) using MATLAB/ Simulink Simulate the lab exercises using MATLAB/Simulink Exp:- 2.a OBJECTIVE: Realising a given block diagram having multiplier, adder/subtractor and system with given Impulse response. Calculating output for given input. Exp:- 2.b Exp:- 2.c Exp:- 2.d OBJECTIVE: To simulate the transmitter and receiver for BPSK. OBJECTIVE: Evaluate a second-order differential equation. OBJECTIVE : To study Sigma-Delta A/D Conversion

69

71

78 81 83 86

Sample Viva Question Exp:- 3 Implementation using pure DSP, pure FPGA and Hybrid DSP/FPGA platforms. Digital Communications: On-Off- Keying (OOK), BPSK modulation, and a simple transceiver design. Wireless

Communications: Channel coding/decoding, Equalization, Simple Detection Algorithm, OFDM. Speech Classification and Synthesis Exp:- 3.a OBJECTIVE : To study the basic introduction with DSP. OBJECTIVE: To study the architecture of TMS320C6XXX DSP kit. OBJECTIVE: To perform linear convolution using DSP processor. OBJECTIVE: To implement circular convolution of two sequences using DSP processor. OBJECTIVE : To compute N-Point DFT of a given sequence using DSP Processor. OBJECTIVE: To find Impulse response of a first order of 88

architecture of TMS320C6XXX DSP kits using Bloom Exp:- 3.b Exp:- 3.c Exp:- 3.d Exp:- 3.e Exp:- 3.f 93 112 125 140 153 168

and second order LTI system using DSP processor. Sample Viva Question

5. Exp:4.a

LIST OF EXPERIMENTS (BEYOND SYLLABUS) OBJECTIVE: Implementation of Transfer function of Log Gabor Filters. Sample Viva Question 170

174 175

Exp:4.b

OBJECTIVE: Removal of noise from image signals using various spatial filtering operations. Sample Viva Question

179 180

Exp:4.c

OBJECTIVE: Write a program in MATLAB to Remove noise from an image signal using various spatial filtering operations. Sample Viva Question

183 184

Exp:4.d

OBJECTIVE: Wavelets, wavelet transform and wavelet decomposition. Sample Viva Question

189 190

Exp:-

OBJECTIVE: Write a program in MATLAB to decompose an image

4.e

signal using Discrete Wavelet Transform. Sample Viva Question 192

Swami Keshvanand Institute of Technology, Management & Gramothan

SYLLABUS

8CS6 FPGA LAB. (Comp. Engg.):

List of Experiments:

1.

Fundamental Theory Introduction to DSP architectures and programming Sampling Theory, Analog-to-Digital Converter (ADC), Digital-to- Analog Converter (DAC), and Quantization; Decimation, Interpolation, Convolution, Simple Moving Average; Periodic Signals and harmonics; Fourier Transform (DFT/FFT), Spectral Analysis, and time/spectrum representations; FIR and IIR Filters;

2. Design (Simulation) using MATLAB/ Simulink. Simulate the lab exercises using MATLAB/Simulink.

3. Implementation using pure DSP, pure FPGA and Hybrid DSP/FPGA platforms. Digital Communications: On-Off- Keying (OOK), BPSK modulation, and a simple transceiver design. Adaptive Filtering: Echo/Noise Cancellation, Least Mean Square (LMS) algorithm . Wireless Communications: Channel coding/decoding, Equalization, Simple Detection Algorithm, OFDM. Speech Processing: Prediction Algorithms, Speech Classification and Synthesis.

Swami Keshvanand Institute of Technology, Management & Gramothan

PROGRAM EDUCTION OBJECTIVES AND OUTCOMES

Branch: Computer Engineering Course Name: FPGA Lab. External Marks: 40 Internal Marks: 60 Total Marks: 100 1. PROGRAM OBJECTIVES:-

Semester: 8th Code: 8CS6 Practical hrs: 3 hrs/week

i. Preparation:- To prepare to pursue advanced graduate studies in computing or related disciplines and provide students broad-based education in core areas of Computer Science, including theoretical foundations, algorithms and data structures, and computer hardware, with an appropriate blend of theory and practice and to specialize in a variety of areas of Computer Science through a selection of elective courses. ii. Core Competence:-To provide students with a solid foundation in computer engineering field required to solve computing problems using various programming languages and softwares, and students can solve problems through logical and analytical thinking. iii. Breathe:-To train students with good computer and engineering breadth so as to comprehend, analyze, design, and create novel products and solutions for the real life. iv. Professionalism:-To inculcate in students professional and ethical attitude, effective communication skills, teamwork skills, multidisciplinary approach, and an ability to relate computer engineering issues to broader social context. v. Learning Environment:-To provide students with an academic environment aware of excellence leadership and lifelong learning needed for successful professional career through independent studies, thesis, internships etc. 2. PROGRAM OUTCOMES:9

a)

Graduates will demonstrate knowledge of mathematics, science & Computer Engineering and will develop ability to apply fundamental principles of computing, mathematics and sciences as appropriate to the discipline of computer science.

b)

Graduates will demonstrate an ability to identify formulate and solve Computer Engineering problems and analyze a problem and model it as a computing system using appropriate methodologies.

c) d)

Graduates will demonstrate an ability to design and conduct experiments, analyze and interpret data. Graduates will develop ability to design a system, component or process implement and test a computer application and to evaluate and compare the efficiencies of alternative solutions as per the needs and specifications.

e) f) g) h) i) j) k)

Graduates will develop an ability to visualize and work on laboratory and multidisciplinary class. Graduates will demonstrate skills to use modern engineering tools softwares and equipments to analyze problems. Graduates will demonstrate knowledge of professional and ethical responsibilities. Graduates will able to communicate effectively in both verbal and written form. Graduate will show the understanding of impact of Computer engineering solutions on the society and also be aware of contemporary issues. Graduate will develop confidence for self education and ability for lifelong learning. Graduate can participate and succeed in competitive examinations like GATE etc.

3. MAPPING:-

PROGRAM OBJECTIVES

PROGRAM OUTCOMES A b C D E f G h i j
10

I II III IV V

X X

x x x X X X X X X X x x x x x

x X

4) Course Educational Objective:After studying FPGA LAB practical in their graduation plan student will be able to: i. Apply the knowledge to manage and to handle various signal based applications like extraction/modification of information in a signal, to solve the real world problems. ii. Gain a repudiated designation as good signal processing manager and administrator ethically by applying signal processing software technologies. iii. Be successfully accepted in futures development scenario as an engineering graduate pursuit of lifelong learning. iv. v. Exhibit team spirit management & effective signal processing dealings. Can give better emerging computer based techniques and ideas to analyze design implement industry based on FPGA module development. vi. The FPGA experiments attempts to provide practical knowledge of applications in digital signal processing using FPGA and DSP platforms used in real world for extracting information from signals.

11

vii.

It provides the convenient environment for the development of chips/modules used to process signals in various electronic systems according to our requirements.

5) Course Outcomes (student learning outcomes):

Each student learning outcome (SLO) maps to one of the program education objective (PEO) as indicates in parentheses following the outcomes. Graduate of the program will: 1. Implementation and understanding of DSP and FPGA circuits/modules. 2. They are able to understand the difference among various DSP applications. 3. Implement proficiency in using and developing various DSP application softwares according to market demand. 4. Can show their ability to apply conceptual skills in signal processing. 5. Graduates will show interest towards the development of different modern and efficient tool (MATLAB and CCStudio 3.) 6. Graduates will able to know about the programming as well as block diagram implementation of a DSP system using MATLAB and SIMULINK of a DSP system. 7. Implement different types of DSP applications and use of them.

8. By understanding it, graduate will be able to design and implement existing, as well as innovative system designs in DSP. 9. They will introduce the concepts of using DSP processor kit and interfacing it with computer software. .

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6. COURSE

OBJECTIVES

CONTRIBUTION

TO

PROGRAM

OUTCOMES:Students who have successfully completed this course will have full understanding of following concepts:COURSE OBJECTIVE PROGRAM OUTCOME i. Apply the knowledge to manage b) Graduates will demonstrate and to handle various signal based applications like extraction/ modification of information in a signal, to solve the real world problems. solve Computer

an

ability to identify formulate and Engineering problems and analyze a problem and model it as a computing system using appropriate methodologies. c) Graduates will demonstrate and an ability data. d) Graduates will develop ability to design a system, component or process computer evaluate implement and and test and a to the application to design conduct

ii.

Gain a repudiated designation as good signal processing manager and administrator ethically by applying signal processing software technologies.

experiments, analyze and interpret

iii.

Be successfully accepted in futures development scenario as an engineering graduate pursuit of lifelong learning.

compare

efficiencies of alternative solutions as per the needs and specifications. i) Graduate will of show impact the of

iv.

Exhibit team spirit management & effective dealings. signal processing

understanding

Computer engineering solutions on the society and also be aware of contemporary issues.

v.

Can give better emerging computer

13

based techniques and ideas to analyze design implement industry based vi. on FPGA module j) Graduate will develop confidence for self education and ability for lifelong learning. 1. Graduates will demonstrate skills to use modern engineering tools software and equipments to analyze problems. development The FPGA experiments attempts to provide practical knowledge of applications in digital signal processing using FPGA and DSP platforms used in real world for extracting information from signals.

MAPPING OF COURSE OBJECTIVE WITH PROGRAM OUTCOMES

SLOs/PEOs A 1 2 3 4 5 6 7

b x

D X

x x

x X x X

x X x

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7. TOPICS COVERED BEYOND THE CURRICULUM:- With the topics in the syllabus the following topics will be covered in this subject which are important for FPGA lab:

Uni t

Beyond Syllabus topic To study the function of Filters At Bandwidth limitations of Gabor filters The Log-Gabor Filter Applications of logGabor filters

Source

Transfer . Log Gabor www.csse.uwa.edu.au/~pk/research/matlabfns/.../convexp

l.html

Removal of noise from image Gonzales R.C., Woods R.E. Digital Image signals using various spatial Processing, (2ndedition). Prentice Hall Publication. filtering operations. II Concept of image. Noise in images Filtering methods to remove noise

Remove noise from an image MATLAB Image Processing Toolbox. III signal using various spatial filtering MATLAB operations using

15

Wavelets, wavelet transform Gonzales R.C., Woods R.E. Digital Image IV and wavelet decomposition Processing, (2ndedition). Prentice Hall Publication.

Decompose an image signal MATLAB Wavelet Toolbox V using Discrete Wavelet Transform using MATLAB. 8. BOOKS:Text books:

1. Getting started with MATLAB 7 -By Rudra Pratap -Chapman 2. Digital signal processing By S Salivahanan
. Reference Books :

Digital signal processing - By Farookh Hussain Communication Systems - By Sanjay Sharma Digital signal processing -By B.P.Lathi Introduction to Simulink with Engineering Applications - Steven T. Karris

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9. INSTRUCTIONAL METHODS:9.1. Direct Instructions:

I. II. III.

Black board presentation PowerPoint presentation Multimedia like video lectures

9.2. Interactive Instruction:

I. II.

Think, pair, share Quiz 9.3 Indirect Instructions:

I. Problem solving

9.4 Independent Instructions: Assignments 10) LEARNING MATERIALS:1. Text/lecturer notes/ lecturer PPT 2 .Multimedia material (videos, text with animations) 3. Web Resources:

www.mathworks.com www.matlab.com www.dsplaboratory.com www.matrixlab.com

12) ASSESSMENT OF OUTCOMES:1. Sessional tests (two in each semester and assessment is done on the basis of average of marks. 2. End term exam (Conducted by RTU, KOTA) 3. Surprise Quiz/ Tests.

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4. Presentation by students. 5. Daily class room interaction. 6. Assignments. 7. Online Exam(Two exams in each semester) 13). OUTCOMES WILL BE ACHIEVED THROUGH FOLLOWING:1. Class room teaching (through chalk and board /PPT). 2. Discussion on case- studies. 3. Video lectures through NPTEL, MATHWORKS.

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LIST OF PROGRAMS

S. NO. 1.a

TITLE Write a program for plotting of various elementary signals like impulse function, unit step function, ramp function, quadratic function, sine wave,and a general sinusoidal function. Write a program for verification of sampling theorem. Write a program to derive the Impulse response of a given system. Write a program to perform linear convolution of two given sequences. Write a program to perform Circular convolution of two given sequences. Write a program for computation of N-point DFT of a given sequence and to plot magnitude and phase spectrum.

PAGE NO 30

1.b 1.c 1.d 1.e 1.f

36 40 43 46 48

1.g 1.h

Write a program for computation of Linear Convolution using DFT & IDFT. Write a program for Down sampling or decimation of the sum of two sinusoids by a factor M.

50 53

1.i 1.j

Write a program to study the design and implementation of FIR filter to meet given specifications Write a program to study the design and implementation of IIR(LPF/HPF) filter to meet given specifications.

56 65

19

2.a

Realising a given block diagram having multiplier, adder/subtractor and system with given Impulse response. Calculating output for given input using SIMULINK

71

2.b 2.c 2.d 3.a 3.b 3.c 3.d 3.e 3.f

To simulate the transmitter and receiver for BPSK using SIMULINK. Evaluate a second-order differential equation using SIMULINK.. To study Sigma-Delta A/D Conversion using SIMULINK. To study the basic introduction of architecture of TMS320C6XXX DSP kits using Bloom with DSP. To study the architecture of TMS320C6XXX DSP kit. Write a program to perform linear convolution using DSP processor. Write a program to implement circular convolution of two sequences using DSP processor. Write a program To compute N-Point DFT of a given sequence using DSP Processor. Write a program to find Impulse response of a first order and second order LTI system using DSP processor.

78 81 83 88 93 112 125 140 153

PROGRAMS BEYOND THE SYLLABUS

4 5

Implementation of Transfer function of Log Gabor Filters. Removal of noise from image signals using various spatial filtering operations.

170 175

20

6 7 8

Write a program in MATLAB to Remove noise from an image signal using various spatial filtering operations. Wavelets, wavelet transform and wavelet decomposition. Write a program in MATLAB to decompose an image signal using Discrete Wavelet Transform.

180 184 190

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INTRODUCTION THEORY OF LAB General instructions for the students regarding MATLAB:1. Always prefer M-files to write the programs in MAT-LAB. 2. Save the M-files with .m extension. 3. Do not save the file with the name which resembles to any predefined function in the MATLAB. 4. Always refer to the MATLAB help while using any new function or in case of confusion. 5. Clear the command window while running any new program by giving the clear command, because it generates confusion. 6. To see the diagrammatic result refer to the figure window of MATLAB. 7. In case of slow processing, dont press the run button or any other button again and again. Give it some time to recover. 8. Save the programs on each turn and simultaneously prepare the file, it will save your time. 9. Never use numbers while saving the programs.

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INTRODUCION TO MATLAB MATLAB MATLAB stands for MATRIX-LABORATORY. MATLAB is a high-level language and interactive environment that enables you to perform computationally intensive tasks faster than with traditional programming languages such as C, C++, and FORTRAN. MATLAB allows easy matrix manipulation, plotting of functions and data, implementation of algorithms, creation of user interfaces and interfacing with programs in other languages. MAT-LAB contains matrix which does not require any dimensions so it is very easy to implement any program or any algorithmmin a few dozen of lines with a great accuracy and with a great efficiency. HISTORY OF MATLAB MAT-LAB was invented in late 1970s by Cleve Moler.he was the chair person of computer science department at the university of New Mexico. He invented it to give the direct access to linpak and eispak without learning fortranto his students. Very soon it becomes very popular in other universities and in other communities. Jack Little an engineer exposed to it when Cleve Moler made a visit to the Stanford University. Later on realizing his potential he joined with Cleve Moler and Steve Bangertto rewroye the MAT-LAB in Cwhich is known as math works in 1984. These rewritten libraries are known as jack-pack.it was firstly used by control design engineers. This is a high-level matrix/array language with control flow statements, functions, data structures, input/output, and object-oriented programming features. It allows both "programming in the small" to rapidly create quick and dirty throw-away programs, and "programming in the large" to create large and complex programs. MATLAB has extensive facilities for displaying vectors and matrices as graphs, as well as annotating and printing these graphs. It includes high-level functions for two-dimensional and three-dimensional data visualization, image processing, animation, and presentation graphics. It also includes low-level functions that allow you to fully customize the appearance of graphics as well

23

as to build complete graphical user interfaces on your MATLAB applications. MATLAB has evolved over a period of years with input from many users. In university environments, it is the standard instructional tool for introductory and advanced courses in mathematics, engineering, and science. In industry, MATLAB is the tool of choice for high-productivity research, development, and analysis In MATLAB, a matrix is a rectangular array of numbers. Special meaning is sometimes attached to 1-by-1 matrices, which are scalars, and to matrices with only one row or column, which are vectors. MATLAB has other ways of storing both numeric and nonnumeric data, but in the beginning, it is usually best to think of everything as a matrix. The operations in MATLAB are designed to be as natural as possible. Where other programming languages work with numbers one at a time, MATLAB allows you to work with entire matrices quickly and easily. A good example matrix, used throughout this book, appears in the Renaissance engraving Melencolia I by the German artist and amateur mathematician Albrecht Drer.

How to get started

When the computer has started go through the following steps in the different menus Look for the Network Applications folder and double click on that Within this you will see a little icon for Matlab double click on that Within about 30 seconds Matlab will open. Starting MATLAB After logging into your account, you can enter MATLAB by double-clicking on the MATLAB shortcut icon on your Windows desktop. When you start MATLAB, a special window called the MATLAB desktop appears. The desktop is a window that contains other windows. The major tools within or accessible from the desktop are: The Command Window

24

 The Command History The Workspace The Current Directory The Help Browser The Start button

Figure-1

25

MAT-LAB PRODUCT FAMILY:MAT-LAB has six members in its product family. 1. Technical computing it includes the mathematical computation, data analysis, algorithm development and vizualisation. 2. Control designing -it is a very good fascility provided in MAT-LAB.by using this fascility we can design open loop and closed loop control systems, we can analyse these systems by connecting different input signals to the input port of the system. 3. Signal processing -it is also related with model designing. It includes the analysis of different signals, which are applied to the input ports of different systems. 4. Image processing -it includes the processing of images and videos, analysis, algorithm development and system design.the tools for image processing in MAT-LAB gives a great help to scientists, researchers, astronauts in the different disciplines such as medical imaging, aerospace, material science. 5. Test and measurements - Test & Measurement solution provides a complete set of tools for test, data analysis and modeling, and presentation-quality reports, all in a single environment. 6. Modeling and simulation -this is a very good facsility provided in MAT-LAB.before implementing any system in real time world, we can simulate it and can see its response to the different inputs. According to its output we can decide its merits and demerits. so before implementing the system in real world , we can improve it with the help of simulation.

SIMULINK

26

Simulink is software for modeling, simulating, and analyzing Dynamic systems. It supports linear and nonlinear systems, modeled in continuous time, sampled time, or a hybrid of the two. Systems can also be multirate, i.e., have different parts that are sampled or updated at different rates. Simulink is a software package that enables you to model, simulate and analyze systems whose output changes over time. Such systems are often referred to as dynamic systems. Simulink helps you to explore the behavior of a wide range of real world dynamic systems including electrical circuits, shock absorbers braking systems and many other electrical, mechanical and thermodynamic systems. Simulating a dynamic system is a two step process. First the user creates a block diagram using a simulink model editor that graphically depicts time dependent mathematical relationship among the systems input states and outputs. The user then commands simulink to simulate the system represented by the model from a specified start time to a specified stop time. Simulink enables you to pose a question about a system, model it, and see what happens. With Simulink, you can easily build models from scratch, or take an existing model and add to it. Thousands of engineers around the world use Simulink to model and solve real problems in a variety of industries. Simulink turns your computer into a lab for modeling and analyzing systems that simply wouldn't be possible or practical otherwise, whether the behavior of an automotive clutch system, the flutter of an airplane wing, the dynamics of a predator-prey model, or the effect of the monetary supply on the economy. Simulink provides numerous demos that model a wide variety of such real-world phenomena. Block-sets Block-sets are specialized collection of simulink blocks built for solving particular problems. It gives the block presentation of the various control systems. Signal processing block-sets is a tool for digital signal processing algorithm simulation and code generation. All the blocks support double and single floating point data types. Most blocks also support fixed point and integer data types when you have simulink fixed point. You can interconnect signal processing block-sets block to create sophisticated models capable of simulating operations such as speech and audio processing, wireless digital communications, radar/sonar and medical electronics.

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Simulink libraries Simulink libraries are used to design the models. It contains the elements required to construct a model of a system. It includes sources, sinks, mathematical operations, transfer function blocks, feedback loops, various types of output devices and input signals.

Figure 2-Simulink Library

Figure 3 -Simulink Library Browser Here in the above figure you can see the sources which are generally connected to the input port of the system. The second element is sink which are connected to the output port of the system to plot the response of the system. The various transfer functions can be written by using the continuous and discrete facility. The math operation provides the mathematical expressions like product and addition. To make a model in MAT-LAB
28

first of all we have to open the simulink library browser. It gives the list of all the essential elements required to construct a model. When we further open the listed elements they will give us the detailed members of the respected families. For example:Simulink library browser sources sine function Unit step function Sinks scope Bode plotter

A SIMPLE BLOCK DIAGRAM:-

Figure-4- Simple block In the above figure the input section contains the sine wave generator which is given to the quantizer. The quantizer converts the continuous wave into the discrete levels. The output can be seen in the scope. TOOL-BOX Control System Toolbox builds on the foundations of MATLAB to provide functions designed for control engineering. Control System Toolbox is a collection of algorithms, written mostly as M-files, that implements common control system design, analysis, and

29

modeling techniques. These are specialized collection of M-files built specifically for solving particular classes of problems. For example:1. Filter design 2. Signal processing EXPERIMENT NO-1.a

OBJECTIVE: Plotting of various elementary signals like impulse function, unit step function, ramp function, quadratic function, sine wave,and a general sinusoidal function. Theory: Here we attempt to plot some elementary functions using MATLAB commonly used in various applications in DSP . these functions are discussed belowImpulse function: The Dirac delta function, or Impulse function, is function that is zero everywhere except at zero such that, The Dirac delta can be loosely thought of as a function on the real line which is zero everywhere except at the origin, where it is infinite-

Figure-1.a.1- impulse function Unit Step Function: Unit Step function u(t) is one for all positive values, and zero for negative values of input. u(t) useful for representing the opening or closing of switches
30

Figure-1.a.2 -step function

Ramp function: In Ramp function, input is equal to output for positive values, and zero elsewhere.

Figure-1.a.3- Ramp function

Quadratic function: This function is integral of ramp function, its non linear in nature.

31

Figure-1.a.4 Quadratic function

y = t2 Sine wave: Sine wave is a sinusoidal function with a single frequency. y = sin(t )

Figure-1.a.5 Sine function

Sinusoidal function: It is a sinusoidal function having two or more frequencies. y = sin(t1) + sin(t 2)

32

Figure-1.a.6 Sinusoidal function

PROGRAM: clear all; close all; clc; t = (0:pi:100); y = sin(2*pi*100*t); y1 = sin(2*pi*10*t) + 2*sin(2*pi*50*t); imp= [1; zeros(99,1)]; unit_step = ones(100,1); ramp_sig= t; quad_sig=t.^2; subplot(3,2,1),stem(imp); title('Impulse Function'); xlabel('time'); ylabel('Amplitude');

33

axis([-10 10 0 1]) subplot(3,2,2),stem(unit_step); title('Unit step Function'); xlabel('time'); ylabel('Amplitude'); axis([-10 10 0 1]); subplot(3,2,3),plot(t,ramp_sig); title('Ramp Function'); xlabel('time'); ylabel('Amplitude'); subplot(3,2,4),plot(t,quad_sig); title('Quadratic Function'); xlabel('time'); ylabel('Amplitude'); subplot(3,2,5),plot(t,y); title('Sinusoidal function'); xlabel('time'); ylabel('Amplitude'); subplot(3,2,6),plot(t,y1); title('y1'); xlabel('time');

34

ylabel('Amplitude');

Output:

35

EXPERIMENT NO-1.b . OBJECTIVE: Verification of sampling theorem. TheorySampling: Is the process of converting a continuous time signal into a discrete time signal. It is the first step in conversion from analog signal to digital signal. Sampling Theorem (Nyquist Theorem ): Sampling theorem states that Exact reconstruction of a continuous time base-band signal from its samples is possible, if the signal is band-limited and the sampling frequency is greater than twice the signal bandwidth. i.e. fs > 2fm, where fm is the highest frequency present in that signal. Nyquist Rate Sampling: The Nyquist rate is the minimum sampling rate required to avoid aliasing, equal to the highest modulating frequency(fm) contained within the signal. In other words, Nyquist rate is equal to two sided bandwidth of the signal (Upper and lower sidebands). To avoid aliasing, the sampling rate must exceed the Nyquist rate. i.e. fs > fm. Aliasing: Aliasing is a phenomenon where the high frequency components of the sampled signal interfere with each other, because of inadequate sampling fs < 2fm. Aliasing leads to distortion in recovered signal. This process is called Undersampling. This is the reason why sampling frequency should be at least twice the bandwidth of the signal. In practice signal are oversampled, where fs is significantly higher than Nyquist rate to avoid aliasing. . This process is called Oversampling. Frequency spectrum of a signal which has undergone aliasing is shown below.

36

Figure-1.b.1 -Distortion due to undersampling(aliasing)

Programclear all; close all; clc; tf=0.05; t=0:0.00005:tf; f=input('Enter the analog frequency,f = '); xt=cos(2*pi*f*t); fs1=1.3*f; n1=0:1/fs1:tf; xn=cos(2*pi*f*n1); subplot(3,1,1),plot(t,xt,'b',n1,xn,'r*-'); title('Undersampling plot');

37

xlabel('time'); ylabel('Amplitude'); fs2=2*f; n2=0:1/fs2:tf; xn=cos(2*pi*f*n2); subplot(3,1,2),plot(t,xt,'b',n2,xn,'r*-'); title('Nyquist plot'); xlabel('time'); ylabel('Amplitude'); fs3=20*f; n3=0:1/fs3:tf; xn=cos(2*pi*f*n3); subplot(313),plot(t,xt,'b',n3,xn,'r*-'); title('Oversampling plot'); xlabel('time'); ylabel('Amplitude');

38

Output: Enter the sampling frequency: 100

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EXPERIMENT NO-1.c OBJECTIVE: Derive the Impulse response of a given system.. TheoryA 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,

Figure-1.c.1 Transform Operation

If the input to the system is unit impulse i.e. x(n) = (n) then the output of the system is known as impulse response denoted by h(n) where,

h(n) = T[(n)]

Any arbitrary sequence x(n) can be represented as a weighted sum of discrete impulses. Now the system response for a linear system is given byy(n) = x(k) T[(n-k)] for k = - to . The response to the shifted impulse sequence can be denoted byh(n- k) = T[(n-k)] that implies that,

40

y(n) = x(k) h(n-k) for k= - to . That implies that for a linear time-invariant system if the input sequence is x(n) and impulse response h(n) is given, we can fine output y(n) by using above equation, which is known as convolution sum(discussed later) and can be represented by y(n) = x(n) * h(n). Program-

clear all; close all; clc; disp('Difference Equation of a digital system'); N=input('Desired Impulse response length = '); b=input('Coefficients of x[n] terms = '); a=input('Coefficients of y[n] terms = '); h=impz(b,a,N); disp('Impulse response of the system is h = '); disp(h); n=0:1:N-1; figure(1); stem(n,h); xlabel('time index'); ylabel('h[n]'); title('Impulse response');

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OUTPUT: [Given y(n)-y(n-1)+0.9y(n-2)= x(n)] Difference Equation of a digital system Desired Impulse response length = 100 Coefficients of x[n] terms = 1 Coefficients of y[n] terms = [1 -1 0.9]

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EXPERIMENT NO-1.d OBJECTIVE: Perform linear convolution of two given sequences. Theory: Convolution is an integral concatenation of two signals. It has many applications in numerous areas of signal processing. The most popular application is the determination of the output signal of a linear time-invariant system by convolving the input signal with the impulse response of the system. Note that convolving two signals is equivalent to multiplying the Fourier Transform of the two signals. Mathematic Formula: The linear convolution of two continuous time signals x(t) and h(t) is defined by-

For discrete time signals x(n) and h(n), is defined by

Where x(n) is the input signal and h(n) is the impulse response of the system. In linear convolution length of output sequence is, length(y(n)) = length(x(n)) + length(h(n)) 1; Graphical Interpretation: Reflection of h(k) resulting in h(-k) Shifting of h(-k) resulting in h(n-k) Element wise multiplication of the sequences x(k) and h(n-k) Summation of the product sequence x(k) h(n-k) resulting in the convolution

43

value for y(n).

Program: clc; x1 = input('Enter the 1st seq:'); x2 = input('Enter the 2nd seq:'); y = conv(x1, x2); disp('The linear convolution of two sequences:'); disp(y); n = 0:length(y)-1; stem(n, y); xlabel('Time'); ylabel('Magnitude'); title('Linear convolution'); Output: Enter the 1st seq:[1 2 3 1] Enter the 2nd seq:[1 1 1] The linear convolution of two sequences: 136641

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EXPERIMENT NO-1.e OBJECTIVE: Perform Circular convolution of two given sequences. Theory: Let x1(n) and x2(n) are finite duration sequences both of length N with DFTs X1(k) and X2(k). Convolution of two given sequences x1(n) and x2(n) is given by the equation, x3(n) = IDFT[X3(k)] X3(k) = X1(k). X2(k) Then,

where N stands for modulo operation.

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Program: clc; x1 = input('Enter 1st seq:'); x2 = input('Enter 2nd seq:'); n = max(length(x1),length(x2)); x1 = fft(x1,n); x2 = fft(x2,n); y = x1.*x2; yc = ifft(y,n); disp('circular convolution:'); disp(yc); N=0:1:n-1; subplot(1,1,1); stem(N,yc); xlabel('Time'); ylabel('Magnitude'); title('Circular convolution'); axis([-1 5 0 15]); OUTPUT: Enter 1st seq:[1 1 2 1] Enter 2nd seq:[1 2 3 4] Circular convolution: 13 14 11 12

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EXPERIMENT NO-1.f OBJECTIVE: Computation of N-point DFT of a given sequence and to plot magnitude and phase spectrum. Theory: Discrete Fourier Transform: The Discrete Fourier Transform is a powerful computation tool which allows us to evaluate the Fourier Transform X (e j ) on a digital computer or specially designed digital hardware. Since X (e j ) is continuous and periodic, the DFT is obtained by sampling one period of the Fourier Transform at a finite number of frequency points. Apart from determining the frequency content of a signal, DFT is used to perform linear filtering operations in the frequency domain. The sequence of N complex numbers x0 ,......xN 1 is transformed into the sequence of N complex numbers X 0 ,...... X N 1 by the DFT according to the formula:

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Program: clc; x1 = input('Enter the sequence:'); n = input('Enter the length:'); m = fft(x1,n); disp('N-point DFT of a given sequence:'); disp(m); N = 0:1:n-1; subplot(2,1,1); stem(N,m); axis([-1 5 -10 10]); xlabel('Length'); ylabel('Magnitude of X(k)'); title('Magnitude spectrum:'); an = angle(m)*180/pi; subplot(2,1,2); stem(N, an); axis([-1 5 -180 180]); xlabel('Length'); ylabel('Phase of X(k)'); title('Phase spectrum:'); Output: Enter the sequence:[1 2 3 4] Enter the length:4 N-point DFT of a given sequence:

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10.0000

-2.0000 + 2.0000i

-2.0000

-2.0000 - 2.0000i

EXPERIMENT NO-1.g

OBJECTIVE: Computation of Linear Convolution using DFT & IDFT. Theory: Convolution is an integral concatenation of two signals. It has many applications in numerous areas of signal processing. The most popular application is the determination of the output signal of a linear time-invariant system by convolving the input signal with the impulse response of the system. Note that convolving two signals is equivalent to multiplying the Fourier Transform of the two signals. Mathematic Formula:
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The linear convolution of two continuous time signals x(t) and h(t) is defined by:

For discrete time signals x(n) and h(n), is defined by

Where x(n) is the input signal and h(n) is the impulse response of the system. For example, let us take two sequences, x1 and x2. For linear convolution, length of output sequence x3 will be, Length N(x3) = Length(x1) + Length(x2) - 1 Also, convolution of two sequences x1(n) and x2(n) is, x3(n) = IDFT[X3(k)] x3(n) = IDFT[X1(k) .X2(k)] Where, X1(k) = DFT [x1(n)]

X2(k) = DFT [x2(n)]

Where,

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And,

Program: clc; x1 = input('Enter the 1st seq:'); x2 = input('Enter the 2nd seq:'); n = length(x1) + length(x2)-1; x1 = fft(x1,n); x2 = fft(x2,n); y = x1.*x2; yc = ifft(y,n); disp('Linear convolution using DFT and IDFT:'); disp(yc); N = 0:1:n-1; stem(N,yc); Axis([-1 4 -5 5]); xlabel('Time'); ylabel('Magnitude'); title('Linear convolution using DFT and IDFT:');

Output: Enter the 1st seq:[1 1 2] Enter the 2nd seq:[1 2] Linear convolution using DFT and IDFT: 1 3 4 4

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Plot:

EXPERIMENT NO-1.h

OBJECTIVE: . Down sampling or decimation of the sum of two sinusoids by a factor M. THEORY: Multirate Digital Signal Processing to change the sampling rate of a signal. Different sampling rates can be obtained using upsampling and downsampling. Decimation is for reducing the sampling rate and interpolation is for increasing sampling rate of a signal. Decimation requires that aliasing should be prevented. Hence, prior to down-samplingwith the factor of M, the original signal has to be bandlimited to /M. This means that the factor of M decimation has to be implemented in two steps: (1) Bandlimiting of the original signal to /M

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(2) Down-sampling by the factor-of-M Figure 1.1 shows the block diagram of a decimator implemented as a cascade of the decimation filter H(z), also called the anti-aliasing filter, and the factor-of- M downsampler.

Figure 1.h.1- Factor-of- M down-sampler Interpolation: The signal x(n) and its three fold upsampled signal y(n). In practice, the zero valued samples inserted by upsampler are replaced with approximate nonzero values using some type of filtering process. The process is called interpolation. This means that the factor-of-L interpolation has to be implemented in two steps: 1. Up-sampling of the original signal by inserting L-1 zero-valued samples between two consecutive samples 2.Replacing zero values from the non-zero values using interpolation. To replace sampling rate of a signal by a fractional rate L/M , block diagram is shown as-

Figure 1.h.2- Factor-of- L/M sampling rate converter.

Program: clc;
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clear all; close all; N = input('Length of ip signal= '); M = input('Dwnsmpling factor= '); f1 = input('Freq. of first sinusoid= '); f2 = input('Freq. of second sinusoid= '); n = 0:N-1; x = 2*sin(2*pi*f1*n)+ 3*sin(2*pi*f2*n); y = decimate(x,M,'fir'); subplot(2,1,1); stem(n,x(1:N)); title('Input sequence'); xlabel('Time'); ylabel('amplitude'); subplot(2,1,2); m = 0:N/M - 1; stem(m,y(1:N/M)); title(['output seq. dwn sampling factor ',num2str(M)]); xlabel('Time'); ylabel('amplitude');

Output: Length of ip signal= 100 Dwnsmpling factor= 4 Freq. of first sinusoid= 40 Freq. of second sinusoid= 60

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EXPERIMENT NO-1.i

OBJECTIVE: . To study the design and implementation of FIR filter to meet given specifications. Theory: Filters are the frequency selective circuit that passes a specified band of frequencies and

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blocks or attenuates signals of frequencies outside this band. Filters are broadly classified in a number of ways:1. Analog filters or Digital filters 2. Passive filters or Active filters 3. Audio frequency filters (A.F) or Radio frequency filters (RF) Analog filter are designed to process analog signals, while digital filters process analog signals using digital techniques. Depending on the type of element used in their construction, filters may be classified as passive or active. Elements used in passive filters are resistors, capacitors and inductors. Active filters, on the other hand, employ transistors or op-amps in addition to the resistors and capacitors. Depending on the frequency used the filters are classified as AF and RF. The most commonly used filters are:1. low pass filter 2. high pass filter 3. band pass filter 4. band reject filter 5. all pass filter A Finite Impulse Response (FIR) filter is a discrete linear time-invariant system Whose output is based on the weighted summation of a finite number of past inputs. An FIR transversal filter structure can be obtained directly from the equation for Discrete-time convolution.

In this equation, x (k) and y (n) represent the input to and output from the filter at time n. h(n-k) is the transversal filter coefficients at time n. These coefficients are Generated by using FDS (Filter Design Software or Digital filter design package).
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FIR filter is a finite impulse response filter. Order of the filter should be specified. Infinite response is truncated to get finite impulse response. Placing a window of finite length does this. Types of windows available are Rectangular, Barlett, Hamming, Hanning, Blackmann window etc. This FIR filter is an all zero filter.

Finite Impulse Response (FIR) Filter: The FIR filters are of non-recursive type, whereby the present output sample is depending on the present input sample and previous input samples. The transfer function of a FIR causal filter is given by-

Where h(n) is the impulse response of the filter. The Fourier transform of h(n) isH (e j ) = h(n)e j n
n =0 N 1

In the design of FIR filters most commonly used approach is using windows. The
j desired frequency response H d (e ) of a filter is periodic in frequency and can be

expanded in Fourier series. The resultant series is given by,

And known as Fourier coefficients having infinite length. One possible way of obtaining [( N 1) / 2] FIR filter is to truncate the infinite Fourier series at n = + Where N is the length of the desired sequence. The Fourier coefficients of the filter are modified by multiplying the infinite impulse response with a finite weighing sequence w(n) called a window.

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Where w(n) = w(-n) =0

[( N 1) / 2] for n + [( N 1) / 2] for n > +

After multiplying w(n) with hd(n), we get a finite duration sequence h(n) that satisfies the desired magnitude response, [( N 1) / 2] for n + [( N 1) / 2] for n > +

h(n) = hd(n).w(n) =0

j The frequency response H (e j ) of the filter can be obtained by convolution of H d (e )

and W (e j ) is given by, H (e j ) = (1/ 2 ) H d (e j )W ((e j ( ) )).d

H (e j ) = H d (e j )*W (e j )

Program: clear all; close all; clc; rp=input('enter passband ripple'); rs=input('enter the stopband ripple'); fp=input('enter passband freq'); fs=input('enter stopband freq'); f=input('enter sampling freq '); wp=2*fp/f; ws=2*fs/f; num=-20*log10(sqrt(rp*rs))-13; dem=14.6*(fs-fp)/f;
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n=ceil(num/dem); n1=n+1; if(rem(n,2)~=0) n1=n; n=n-1; end c=input('enter your choice of window function 1. rectangular 2. triangular 3.kaiser: \n '); if(c==1) y=rectwin(n1); disp('Rectangular window filter response'); end if (c==2) y=triang(n1); disp('Triangular window filter response'); end if(c==3) y=kaiser(n1); disp('kaiser window filter response'); end %LPF b=fir1(n,wp,y); [h,o]=freqz(b,1,256); m=20*log10(abs(h)); subplot(2,2,1);plot(o/pi,m); title('LPF'); ylabel('Gain in dB-->'); xlabel('(a) Normalized frequency-->');

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%HPF 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); title('HPF'); ylabel('Gain in dB-->'); xlabel('(b) Normalized frequency-->'); %BPF wn=[wp ws]; b=fir1(n,wn,y); [h,o]=freqz(b,1,256); m=20*log10(abs(h)); subplot(2,2,3);plot(o/pi,m); title('BPF'); ylabel('Gain in dB-->'); xlabel('(c) Normalized frequency-->'); %BSF b=fir1(n,wn,'stop',y); [h,o]=freqz(b,1,256); m=20*log10(abs(h)); subplot(2,2,4);plot(o/pi,m); title('BSF'); ylabel ('Gain in dB-->'); xlabel ('(d) Normalized frequency-->');

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Output: enter passband ripple0.02 enter the stopband ripple0.01 enter passband freq1000 enter stopband freq1500 enter sampling freq 10000 enter your choice of window function 1. rectangular 2. triangular 3.kaiser: 1. Rectangular window filter response

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2. Enter passband ripple0.02 enter the stopband ripple0.01 enter passband freq1000 enter stopband freq1500 enter sampling freq 10000 enter your choice of window function 1. rectangular 2. triangular 3.kaiser:

3. Triangular window filter response

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3. Enter passband ripple0.02 enter the stopband ripple0.01 enter passband freq1000 enter stopband freq1500 enter sampling freq 10000 enter your choice of window function 1. rectangular 2. triangular 3.kaiser: 4. kaiser window filter response-

EXPERIMENT NO -1.j

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OBJECTIVE : To study the design and implementation of IIR(LPF/HPF) filter to meet given specifications. Theory: The IIR filter can realize both the poles and zeroes of a system because it has a rational transfer function, described by polynomials in z in both the numerator and the denominator:

The difference equation for such a system is described by the following:

M and N are order of the two polynomials bk and ak are the filter coefficients. These filter coefficients are generated using FDS(Filter Design software or Digital Filter design package). IIR filters can be expanded as infinite impulse response filters. In designing IIR filters, cutoff frequencies of the filters should be mentioned. The order of the filter can be estimated using butter worth polynomial. Thats why the filters are named as butter worth filters. Filter coefficients can be found and the response can be plotted.

Program: clc;
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clear all; close all; disp('enter the IIR filter design specifications'); rp=input('enter the passband ripple'); rs=input('enter the stopband ripple'); wp=input('enter the passband freq'); ws=input('enter the stopband freq'); fs=input('enter the sampling freq'); w1=2*wp/fs;w2=2*ws/fs; [n,wn]=buttord(w1,w2,rp,rs,'s'); c=input('enter choice of filter 1. LPF 2. HPF \n '); if(c==1) disp('Frequency response of IIR LPF is:'); [b,a]=butter(n,wn,'low','s'); end if(c==2) disp('Frequency response of IIR HPF is:'); [b,a]=butter(n,wn,'high','s'); end w=0:.01:pi; [h,om]=freqs(b,a,w); m=20*log10(abs(h)); an=angle(h); figure,subplot(2,1,1);plot(om/pi,m); title('magnitude response of IIR filter is:'); xlabel('(a) Normalized freq. -->'); ylabel('Gain in dB-->'); subplot(2,1,2);plot(om/pi,an); title('phase response of IIR filter is:'); xlabel('(b) Normalized freq. -->'); ylabel('Phase in radians-->');

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Output: Low Pass Partnerenter the IIR filter design specifications enter the passband ripple0.15 enter the stopband ripple60 enter the passband freq1500 enter the stopband freq3000 enter the sampling freq7000 enter choice of filter 1. LPF 2. HPF 1 Frequency response of IIR LPF is:

enter the IIR filter design specifications enter the passband ripple0.15 enter the stopband ripple60

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enter the passband freq1500 enter the stopband freq3000 enter the sampling freq7000 enter choice of filter 1. LPF 2. HPF 2 Frequency response of IIR HPF is:

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Unit -1 Sample Viva Questions

Q.1 Define various elementary signals used in DSP applications, along with their mathematical interpretion. Q.2 Define sampling. State sampling theorm for perfect reconstruction of signal. Q.Define aliasing? How aliasing is eliminated? Q.3 What is nyquist sampling, oversampling and undersampling. Q.4 What is a LTI system ? Define impulse response of an LTI system. Q.5 What are FIR and IIR LTI systems? Q. 6 Differentiate between recursive and non-recursive LTI systems? Q.7What is linear convolution? Q.8 Define circular convolution. Q.9 What is difference between circular and linear convolution? Q.10 What is Fourier Transform? Define discrete Fourier Transform. Q.11 How linear convolution can be performed using DFT? Q.12 Define Direct form I and Direct Form II network structures for IIR systems. Q.13 Differentiate between Analog filters and Digital filters. Q.14 Differentiate between Passive filters or Active filters. Q.15 What is an IIR digital filter ? Q.16 What is an FIR digital filter ?

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Q.17 Compare IIR and FIR digital filters. Q.18 Define window function. Q.19 Discuss various types of window function used in the design of FIR digital filters? Q.20 Define digital Butterworth filter. Q.21 What is Multirate Digital Signal Proccessing? Q.22 Differentiate between up-sampling and down-sampling. Q.23 Define decimation and its significance. Q. 24 Define interpolation and its significance. Q. 25 Describe the sampling rate conversion by a rational factor L/M.

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Implementation using Simulink

EXPERIMENT 2.a
OBJECTIVE Realising a given block diagram having multiplier, adder/subtractor and system with given Impulse response. Calculating output for given input.

THEORY-

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71

Example-Given difference equation-y[n]= .75 y[n-1]- 0.125y[n-2]+ x[n]+2x[n1]+x[n-2]

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Direct form 1 block diagram

Direct form 2 block diagram

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Procedure for make block diagram1. Start matlab. 2. Open model option in new option. 3. A new window is open and in this new window open simulink library browser. 4. Drag different blocks from library browser according to requirement. 5. Draw the block diagram. 6. Start the simulation. 7. Output can see in scope block that is same in case of direct form1 and direct form2.

Simulink model-

We implement here, two Direct form 2, systems, with given transfer functionsFirst one has a Transfer function-

Its block diagram is shown below-

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OUTPUT: If we give a square wave of amplitude 1, and frequency of 0.005 Hz, Input and output waveforms for the model will be-

Second one has a Transfer function-

Its block diagram is-

A demo model using fixedpoint Simulink blocks can be displayed by typingfxpdemo_direct_form2

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in MATLABs Command Window. This demo is an implementation of the thirdorder transfer function. Input and Output waveforms will be like-

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EXPERIMENT 2.b
OBJECTIVE : To simulate the transmitter and receiver for BPSK. SOFTWARE USED- MATLAB 7.5 THEORYBPSK-binary phase shift keying. In BPSK modulation, the phase of the RF carrier is shifted 180 degrees in accordance with a digital bit stream. The digital coding scheme used is called NRZ-M. A "one" causes a phase transition, and a "zero" does not produce a transition. That way you don't have to worry about the polarity of the signal. The receiver performs a differentially coherent detection process, in which the phase of each bit is compared to the phase of the preceding bit. Better performance can be obtained with fully coherent PSK, but that requires an absolute phase reference at each end, and no phase variations in the propagation path. Pretty hard to achieve, even on LF. Because the output of the receiver's coherent detector swings from +V to -V when there's a phase transition, rather than from +V to zero in the case of on-off keying, BPSK offers a 6-dB advantage in signal-to-noise ratio over on-off keying for a given carrier level. BPSK TRANSMITTER-

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Simulink model-

BPSK RECEIVER-

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Output of bpsk transmitter-

Outputs of bpsk receiver-

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EXPERIMENT 2.c
OBJECTIVE : Evaluate a second-order differential equation. THEORYHere we will evaluate a second-order, non-homogeneous differential equation with constant coefficients, using simulink. We have a differential equation obtained by an electrical circuit. Differential equation is-

Block diagram for the above given eq. is-

Figure 2.c.1- Block diagram for the given differential equation

Here, u0(t) stands for unit step function. From above figure it is clear that we need 2 integraters, 1 adder, and 3 gain blocks. The resultant simulink model is-

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Simulink model-

Output: The output waveform for vc is-

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EXPERIMENT 2.d
OBJECTIVE - To study Sigma-Delta A/D Conversion THEORYSigma-delta modulation has recently become the method of choice for high-resolution A/D conversion. The benefits of oversampled noise-shaping converters include inherent linearity, reduced anti-aliasing filter complexity, high tolerance to circuit imperfection. The process of converting an analog signal into a finite range number system (quantization) introduces an error signal that depends on how the signal is being approximated. This quantization error is on the order of one least-significant-bit (LSB) in amplitude, and it is quite small compared to full-amplitude signals. However, as the input signal gets smaller, the quantization error becomes a larger portion of the total signal. The arrangement shown in Figure below is called a Sigma-Delta (S-D) Modulator.

Figure 2.d.1- Sigma-Delta A/D Conversion The name Sigma-Delta modulator comes from putting the integrator (sigma) in front of the delta modulator. Delta modulation is based on quantizing the change in the signal from sample to sample rather than the absolute value of the signal at each sample. Delta modulation is based on quantizing the change in the signal from sample to sample rather than the absolute value of the signal at each sample. The 1-bit digital output from the

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modulator is supplied to a digital decimation filter which yields a more accurate representation of the input signal. The output of the sigma-delta modulator is a 1-bit data stream at the sampling rate, which can be in the megahertz range. The purpose of the digital-and-decimation filter is to extract information from this data stream and reduce the data rate to a more useful value. In a sigma-delta ADC, the digital filter averages the 1-bit data stream, improves the ADC resolution, and removes quantization noise that is outside the band of interest. It determines the signal bandwidth, settling time, and stopband rejection. Our Sigma-Delta A/D Converter approximation loop runs at 512 kHz. Decimation by 64 yields final 8 kHz A/D rate. the analog input is prefiltered by an antialiasing prefilter whose structure is simplified because of oversampling. The input signal is oversampled by a factor of 64. The Integrator, 1-Bit Quantizer, and Zero-Order Hold blocks comprise a two-level analog to digital converter (ADC). The output of the Zero-Order Hold is then subtracted from the analog input. The feedback, or approximation, loop causes the quantization noise generated by the ADC to be highpass filtered, pushing its energy towards the higher frequencies (64*fs/2) and away from the relevant signal band. The decimation stage reduces the sampling rate back to 8 KHz. During this process, it removes the high frequency quantization noise that was introduced by the feedback loop and removes any undesired frequency components beyond fs/2 (4 KHz) that were not removed by the simple analog prefilter. Each decimator stage reduces the sampling rate by a factor of four. In figure, above is analog input, and below is the discrete digitalized output.

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Simulink model-

Output Waveforms-

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Unit -2 Sample Viva Questions

Q.1 Describe MATLAB and its applications. Q.2 What are desktop tools available in MATLAB. Q.3 What is the function of command window, and workspace. Q.4 Why are the function of commands clc, clear all, close all. Q.5 What is difference between plot and stem command. Q.6 What is the function of subplot command. Q.7 What is a MATLAB Toolbox. Q.8 Define SIMULINK. What are SIMULINK Libraries. Q.9 Define SIMULINK Blockset. Q.10 What is the function of a scope in SIMULINK. Q.11How we can a sub-system in SIMULINK. Q.12 Define BPSK modulation. Q.13Describe briefly BPSK transmitter and receiver. Q.14 What is a direct form I and direct form II representation of an LTI system. Q.15 Define second order, non-homogenous differential equation. Q.16 What is a delta modulation? Q.17What are the basic steps in A/D conversion? Q.18 Define a sigma-delta A/D converter?

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References:1. Getting started with MATLAB 7 -By Rudra Pratap 2. Digital signal processing - By S Salivahanan A Vallavaraj C Gyanapriya 3. Digital signal processing -By B.P.Lathi 4. Analog & digital communication -By Sanjay Sharma Web references: www.mathworks.com www.matlab.com www.dsplaboratory.com www.matrixlab.com

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DSP Lab using TMS320C6XXX DSP Kits

Experiment 3.a
OBJECTIVE : To study the basic introduction of architecture of TMS320C6XXX DSP kits using Bloom with DSP. INTRODUCTION TO DSP PROCESSORS A signal can be defined as a function that conveys information, generally about the state or behavior of a physical system. There are two basic types of signals viz Analog (continuous time signals which are defined along a continuum of times) and Digital (discrete-time). Remarkably, under reasonable constraints, a continuous time signal can be adequately represented by samples, obtaining discrete time signals. Thus digital signal processing is an ideal choice for anyone who needs the performance advantage of digital manipulation along with todays analog reality. digital processing must offer some clear advantages that include: Programmability: A single piece of digital DSP hardware can perform many functions. For example, a multimedia PC can play music and also function as a word processor if it is loaded with suitable programs. This ability to use the same hardware for many functions provides important flexibility. Stability: The stability of analog circuits depends upon several factors. Analog circuits are affected by temperature and aging, and tolerence. Also, two analog systems using the same design and components may differ in performance. Repeatability: A properly designed digital circuit will produce the same result every time, in addition to being identicalfrom unit to unit. If the same multiplication is performed on 500 computers, all 500 computers shouldproduce the same result.

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Component tolerances, aging, and temperature drifts also do not affect digital circuits nearly as much. Temperature: Digital circuits do not gradually change their characteristics over time, temperature, or humidity. Theyeither work or they dont work. In other words, digital circuits are repeatable as long as they are designed with enough tolerance to operate properly over the range of expected conditions. Aging: The effects of component aging can be detrimental to analog circuits as characteristics and performancechange. These effects can sometimes be anticipated, or their effect may not be critical. Analog designers must be aware of these effects. Some practical applications of DSP systems are: Toys, Videophones, Modems, Phone Systems, 3D Graphics, Image Processing. Components of a Typical DSP System Typical DSP systems consist of a DSP chip, memory, possibly an analog-to-digital converter (ADC), digital-to-analog converter (DAC), and communication channels. DSP Chip A DSP chip can contain many hardware elements; some of the more common ones are listed below. Central Arithmetic Unit This part of the DSP performs major arithmetic functions such as multiplication and addition. It is the part that makes the DSP so fast in comparison with traditional processors. Auxiliary Arithmetic Unit DSPs frequently have an auxiliary arithmetic unit that performs pointer arithmetic, mathematical calculations, or logical operations in parallel with the main arithmetic unit. Serial Ports DSPs normally have internal serial ports for high-speed communication with other DSPs and data converters. These serial ports are directly connected to the internal buses to improve performance, to reduce external address decoding problems, and to reduce cost.

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Memory Memory holds information, data, and instructions for DSPs and is an essential part of any DSP system. Although DSPs are intelligent machines, they still need to be told what to do. Memory devices hold a series of instructions that tell the DSP which operations to perform on the data (i.e., information). A/D and D/A Converters Analog-to-digital converters (ADCs) accept analog input and turn it into digital data that consist of only 0s and 1s. Digital-to-analog converters (DACs) perform the reverse process; they accept digital data and convert it to a continuous analog signal. Ports Communication ports are necessary for a DSP system. Raw information is received and processed; then that information is transmitted to the outside world through these ports. For example, a DSP system could output information to a printer through a port. The most common ports are serial and parallel ports. A serial port accepts a serial (single) stream of data and converts it to the processor format. When the processor wishes to output serial data, the port accepts processor data and converts it to a serial stream (e.g., modem connectionson PCs). A parallel port does the same job, except the output and input are in parallel (simultaneous) format. The most common example of a parallel port is a printer port on a PC. Hence a processor which is designed to perform the special operations (digital manipulations) on the digital signal within very less time can be called as a Digital signal processor. The difference between a DSP processor, conventional microprocessor and a microcontroller are listed below. Microprocessor or General Purpose Processor such as Intel xx86 or Motorola 680xx family

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Contains - only CPU -No RAM -No ROM -No I/O ports -No Timer Microcontroller such as 8051 family Contains - CPU - RAM - ROM -I/O ports - Timer & - Interrupt circuitry Some Micro Controllers also contain A/D, D/A and Flash Memory DSP Processors such as Texas instruments and Analog Devices Contains - CPU - RAM -ROM - I/O ports - Timer Optimized for fast arithmetic - Extended precision - Dual operand fetch - Zero overhead loop - Circular buffering The basic features of a DSP Processor are:

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Feature Fast-Multiply accumulate

Use Most DSP algorithms, including filtering, transforms, etc. are multiplication- intensive

Multiple access memory architecture

Many data-intensive DSP operations require reading a program instruction and multiple data items during each instruction cycle for best performance

Specialized addressing modes

Efficient handling of data arrays and first-in, firstout buffers in memory

Specialized program control

Efficient control of loops for many iterative DSP algorithms. Fast interrupt handling for frequent I/O operations.

On-chip peripherals and I/O interfaces

On-chip peripherals like A/D converters allow for small low cost system designs. Similarly I/O interfaces tailored for common peripherals allow clean interfaces to off-chip I/O devices.

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Experiment-3.b
OBJECTIVE : To study the architecture of TMS320C6XXX DSP kit. Theory: Computers need instructions to operate. At every clock cycle, they must be told what to do. If the instructions are stored, the computer just has to fetch and execute them. Such computers are called stored Program machines. Our computer typically fetches an instruction and then data, operates on the data, and returns the resulting data to the store. Stored program machines use two well-known and widely used computer architectures: von Neuman and Harvard.. The following diagram shows the structure of the two architectures. von Neuman Architecture The von Neuman machines store programming and data in the same memory area. In this type of machine, an instruction contains the operation command and the address of the data on which the operation is performed. There are two basic operation units within these machines: the arithmetic logic unit (ALU) and the input/output unit. The ALU performs the core operations: multiply, add, subtract, and many more. It is on these very simple core operations that complex software, such as word processing software, can be built. The input/output unit manages the flow of external data for the machine. Harvard Architecture The primary difference between Harvard architecture and von Neuman architecture is that with Harvard, program and data memories are physically separated transmission paths. This enables the machine to transfer instructions and data simultaneously. Such a structure can greatly enhance performance, because instructions and data can be fetched simultaneously. Harvard machines also have ALUs and input/output units. The drawback to using a true Harvard architecture is that since it uses separate program and data memories, it needs twice as many address and data pins on the chip and twice as much external memory. Unfortunately, as the number of pins or chips increases, so does the price.
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Electronic designers, who have had to tackle problems like these before, have come up with an elegant solution: a single data and address bus is used externally, while two (or more) separate buses for program and data are used internally. Timing (multiplexing) handles the separation of program and data information. In one clock cycle, the program information flows on the pins, and in the second cycle, data follows on the same pins. Program and data information is then routed onto separate internal program and data buses. Such machines are called modified Harvard architecture processors because the internal architecture is Harvard while the external architecture is von Neuman.

Figure 3.b.1- Harward and von Neuman architecture.

ARCHITECTURE OF 6713 DSP PROCESSOR This chapter provides an overview of the architectural structure of the TMS320C67xx DSP, which comprises the central processing unit (CPU), memory, and on-chip peripherals. The C67xE DSPs use an advanced modified Harvard architecture that maximizes processing power with eight buses. Separate program and data spaces allow simultaneous access to program instructions and data, providing a high degree of parallelism. For example, three reads and one write can be performed in a single cycle. Instructions with parallel store and application-specific instructions fully utilize this

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architecture. In addition, data can be transferred between data and program spaces. Such Parallelism supports a powerful set of arithmetic, logic, and bit-manipulation operations that can all be performed in a single machine cycle. Also, the C67xx DSP includes the control mechanisms to manage interrupts, repeated operations, and function calling.

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Figure 3.d.2- BLOCK DIAGRAM OF TMS 320VC 6713

Bus Structure The C67xx DSP architecture is built around eight major 16-bit buses (four program/data buses and four address buses):

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_ the program bus (PB) carries the instruction code and immediate operands from Program memory. _ Three data buses interconnect to various elements, such as the CPU, data address generation logic, program address generation logic, on-chip peripherals, and data memory. _ The CB and DB carry the operands that are read from data memory. _ The EB carries the data to be written to memory. _ Four address buses (PAB, CAB, DAB, and EAB) carry the addresses needed for Instruction execution. The C67xx DSP can generate up to two data-memory addresses per cycle using the two auxiliary register arithmetic units (ARAU0 and ARAU1). The PB can carry data operands stored in program space (for instance, a coefficient table) to the multiplier and adder for multiply/accumulate operations or to a destination in data space for data move instructions (MVPD and READA). This capability, in conjunction with the feature of dual-operand read, supports the execution of single-cycle, 3-operand instructions such as the FIRS instruction. The C67xx DSP also has an on-chip bidirectional bus for accessing on-chip peripherals. This bus is connected to DB and EB through the bus exchanger in the CPU interface. Accesses that use this bus can require two or more cycles for reads and writes, depending on the peripherals structure. Central Processing Unit (CPU) The CPU is common to all C67xE devices. The C67x CPU contains: _ 40-bit arithmetic logic unit (ALU) _ Two 40-bit accumulators _ Barrel shifter _ 17 17-bit multiplier _ 40-bit adder _ Compare, select, and store unit (CSSU)
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_ Data address generation unit _ Program address generation unit Arithmetic Logic Unit (ALU) The C67x DSP performs 2s-complement arithmetic with a 40-bit arithmetic logic unit (ALU) and two 40-bit accumulators (accumulators A and B). The ALU can also perform Boolean operations. The ALU uses these inputs: _ 16-bit immediate value _ 16-bit word from data memory _ 16-bit value in the temporary register, T _ Two 16-bit words from data memory _ 32-bit word from data memory _ 40-bit word from either accumulator The ALU can also function as two 16-bit ALUs and perform two 16-bit operations Simultaneously.

Figure 2.d.3 ALU UNIT

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Accumulators Accumulators A and B store the output from the ALU or the multiplier/adder block. They can also provide a second input to the ALU; accumulator A can be an input to the Multiplier/adder. Each accumulator is divided into three parts: _ Guard bits (bits 3932) _ High-order word (bits 3116) _ Low-order word (bits 150) Instructions are provided for storing the guard bits, for storing the high- and the loworder accumulator words in data memory, and for transferring 32-bit accumulator words in or out of data memory. Also, either of the accumulators can be used as temporary storage for the other.

Barrel Shifter The C67x DSP barrel shifter has a 40-bit input connected to the accumulators or to data memory (using CB or DB), and a 40-bit output connected to the ALU or to data memory (using EB). The barrel shifter can produce a left shift of 0 to 31 bits and a right shift of 0 to 16 bits on the input data. The shift requirements are defined in the shift count field of the instruction, the shift count field (ASM) of status register ST1, or in temporary register T (when it is designated as a shift count register).The barrel shifter and the exponent encoder normalize the values in an accumulator in a single cycle. The LSBs of the output are filled with 0s, and the MSBs can be either zero filled or sign extended, depending on the state of the sign-extension mode bit (SXM) in ST1. Additional shift capabilities enable the processor to perform numerical scaling, bit extraction, extended arithmetic, and overflow prevention operations. Multiplier/Adder Unit
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The multiplier/adder unit performs 17 _ 17-bit 2s-complement multiplications with a 40bit addition in a single instruction cycle. The multiplier/adder block consists of several elements: a multiplier, an adder, signed/unsigned input control logic, fractional control logic, a zero detector, a rounder (2s complement), overflow/saturation logic, and a 16-bit temporary storage register (T). The multiplier has two inputs: one input is selected from T, a data-memory operand, or accumulator A; the other is selected from program memory, data memory, accumulator A, or an immediate value. The fast, on-chip multiplier allows the C54x DSP to perform operations efficiently such as convolution, correlation, and filtering. In addition, the multiplier and ALU together execute multiply/accumulate (MAC) computations and ALU operations in parallel in a single instruction cycle. This function is used in determining the Euclidian distance and in implementing symmetrical and LMS filters, which are required for complex DSP algorithms. See section 4.5, Multiplier/Adder Unit, on page 4-19, for more details about the multiplier/adder unit.

Figure 2.d.4 - MULTIPLIER/ADDER UNIT

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These are the some of the important parts of the processor and you are instructed to go through the detailed architecture once which helps you in developing the optimized code for the required application. PROCEDURE FOR 6713 DSK Step: I 1. First connect the dsk 6713 kit by USB cable and power on. 2. Now some LEDs glowing on 3. Open the 6713 DSK diagnostic (driver) icon. 4. Reset the kit 5. Give start in driver, now it start to checks all . 6. Finally it displays PASS. 7. Then click close the window. Step: II 1. Open the CCS setup and select Family-c67xx, platform-DSK, endianneslittle. 2. Select C67xx DSK board and add then save & quit. 3. Now CCS will open. 4. Go Debugconnect 5. ProjectNew projectgive project name Finish. 6. File New Source file. (now editor window will open). 7. Type your C code in editor window. 8. File Save give name.C Save. 9. Project Add files to project select your C program and open it. 10. Project Save project. 11. Project Build. (now your c code converted into .out format) 12. File Load program Select your .out file Open. (.out file present in the path where you are stored .C file)

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13. Now your program is loading and disassembly will open. 14. If your program wants inputs go View Memory. 15. Give data memory location Ok (now new window will open) 16. Double click on data side new window will open give data increment the address. 17. Open your output memory location also. 18. Debug Run debug Halt. 19. Now you will see the output in your output memory location

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Experiment-3.c
OBJECTIVE : To perform linear convolution using DSP processor. Procedure to create new Project: 1. To create project, Go to Project and Select New.

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Mathematical Formula: The linear convolution of two continuous time signals x(t) and h(t) is defined by:

For discrete time signals x(n) and h(n), is defined by:

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Where x(n) is the input signal and h(n) is the impulse response of the system. In linear convolution length of output sequence is, Length (y(n)) = length(x(n)) + length(h(n)) 1. Program: #include<stdio.h> main() { int m=4; /*Lenght of i/p samples sequence*/ int n=4; /*Lenght of impulse response Co-efficients */ int i=0,j; int x[10]={1,2,3,4,0,0,0,0}; /*Input Signal Samples*/ int h[10]={1,2,3,4,0,0,0,0}; /*Impulse Response Co-efficients*/ /*At the end of input sequences pad 'M' and 'N' no. of zero's*/ int *y; y=(int *)0x0000100; for(i=0;i<m+n-1;i++) { y[i]=0; for(j=0;j<=i;j++) y[i]+=x[j]*h[i-j]; } for(i=0;i<m+n-1;i++) printf("%d\n",y[i]); } Output: 1, 4, 10, 20, 25, 24, 16. Or

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#include<stdio.h> int x[15],h[15],y[15]; main () { int i,j,m,n; printf("\n enter value for m"); scanf("%d",&m); printf("\n enter value for n"); scanf("%d",&n); printf("Enter values for i/p x(n):\n"); for(i=0;i<m;i++) scanf("%d",&x[i]); printf("Enter Values for i/p h(n) \n"); for(i=0;i<n; i++) scanf("%d",&h[i]); // padding of zeros for(i=m;i<=m+n-1;i++) x[i]=0; for(i=n;i<=m+n-1;i++) h[i]=0; /* convolution operation */ for(i=0;i<m+n-1;i++) { y[i]=0; for(j=0;j<=i;j++) { y[i]=y[i]+(x[j]*h[i-j]); }} //displaying the o/p for(i=0;i<m+n-1;i++) printf("\n The Value of output y[%d]=%d",i,y[i]);

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} Result: enter value for m4 enter value for n4 Enter values for i/p 1234 Enter Values for n 1234 The Value of output y[0]=1 The Value of output y[1]=4 The Value of output y[2]=10 The Value of output y[3]=20 The Value of output y[4]=25 The Value of output y[5]=24 The Value of output y[6]=16

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Experiment 3.d
OBJECTIVE : To implement circular convolution of two sequences. Theory: Circular Convolution: Let x1(n) and x2(n) are finite duration sequences both of length N with DFTs X1(k) and X2(k). Convolution of two given sequences x1(n) and x2(n) is given by the equation, x3(n) = IDFT[X3(k)] X3(k) = X1(k) X2(k)

Program: #include<stdio.h> int m,n,x[30],h[30],y[30],i,j,temp[30],k,x2[30],a[30]; void main() { int *y; y=(int *)0x0000100; printf(" enter the length of the first sequence\n"); scanf("%d",&m); printf(" enter the length of the second sequence\n"); scanf("%d",&n); printf(" enter the first sequence\n"); for(i=0;i<m;i++) scanf("%d",&x[i]); printf(" enter the second sequence\n");

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for(j=0;j<n;j++) scanf("%d",&h[j]); if(m-n!=0) /*If length of both sequences are not equal*/ { if(m>n) /* Pad the smaller sequence with zero*/ { for(i=n;i<m;i++) h[i]=0; n=m; } for(i=m;i<n;i++) x[i]=0; m=n; } y[0]=0; a[0]=h[0]; for(j=1;j<n;j++) /*folding h(n) to h(-n)*/ a[j]=h[n-j]; /*Circular convolution*/ for(i=0;i<n;i++) y[0]+=x[i]*a[i]; for(k=1;k<n;k++) { y[k]=0; /*circular shift*/ for(j=1;j<n;j++) x2[j]=a[j-1]; x2[0]=a[n-1]; for(i=0;i<n;i++) {

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a[i]=x2[i]; y[k]+=x[i]*x2[i]; } } /*displaying the result*/ printf(" the circular convolution is\n"); for(i=0;i<n;i++) printf("%d ",y[i]); } Or /* program to implement circular convolution */ #include<stdio.h> int m,n,x[30],h[30],y[30],i,j, k,x2[30],a[30]; void main() { printf(" Enter the length of the first sequence\n"); scanf("%d",&m); printf(" Enter the length of the second sequence\n"); scanf("%d",&n); printf(" Enter the first sequence\n"); for(i=0;i<m;i++) scanf("%d",&x[i]); printf(" Enter the second sequence\n"); for(j=0;j<n;j++) scanf("%d",&h[j]); if(m-n!=0) /*If length of both sequences are not equal*/ { if(m>n) /* Pad the smaller sequence with zero*/ { for(i=n;i<m;i++)

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h[i]=0; n=m; } for(i=m;i<n;i++) x[i]=0; m=n; } y[0]=0; a[0]=h[0]; for(j=1;j<n;j++) /*folding h(n) to h(-n)*/ a[j]=h[n-j]; /*Circular convolution*/ for(i=0;i<n;i++) y[0]+=x[i]*a[i]; for(k=1;k<n;k++) { y[k]=0; /*circular shift*/ for(j=1;j<n;j++) x2[j]=a[j-1]; x2[0]=a[n-1]; for(i=0;i<n;i++) { a[i]=x2[i]; y[k]+=x[i]*x2[i]; } } /*displaying the result*/ printf(" The circular convolution is\n"); for(i=0;i<n;i++) printf("%d \t",y[i]);

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} Output: enter the length of the first sequence 4 enter the length of the second sequence 4 enter the first sequence 4321 enter the second sequence 1111 the circular convolution is 10 10 10 10 Procedure to create new Project: 1. To create project, go to Project and Select New.

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Experiment 3.e
OBJECTIVE : To compute N-Point DFT of a given sequence using DSP Processor. Theory: Discrete Fourier Transform: The Discrete Fourier Transform is a powerful computation tool which allows us to evaluate the Fourier Transform X(ej) on a digital computer or specially designed digital hardware. Since X(ej) is continuous and periodic, the DFT is obtained by sampling one period of the Fourier Transform at a finite number of frequency points. Apart from determining the frequency content of a signal, DFT is used to perform linear filtering operations in the frequency domain. The sequence of N complex numbers x0,..., xN1 is transformed into the sequence of N complex numbers X0, ..., XN1 by the DFT according to the formula:

Program: //DFT of N-point from lookup table. Output from watch window #include <stdio.h> #include <math.h> #define N 4 //number of data values float pi = 3.1416; short x[N] = {1,1,0,0}; //1-cycle cosine float out[2] = {0,0}; //initialize Re and Im results void dft(short *x, short k, float *out) //DFT function { float sumRe = 0; //initialize real component float sumIm = 0; //initialize imaginary component int i = 0;

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float cs = 0; //initialize cosine component float sn = 0; //initialize sine component for (i = 0; i < N; i++) //for N-point DFT { cs = cos(2*pi*(k)*i/N); //real component sn = sin(2*pi*(k)*i/N); //imaginary component sumRe = sumRe + x[i]*cs; //sum of real components sumIm = sumIm - x[i]*sn; //sum of imaginary components } out[0] = sumRe; //sum of real components out[1] = sumIm; //sum of imaginary components 87 printf("%f %f\n",out[0],out[1]); } void main() { int j; for (j = 0; j < N; j++) dft(x, j, out); //call DFT function } Output: 2.000000 0.000000 0.999996 -1.000000 0.000000 0.000007 1.000011 1.000000

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Procedure to create new Project: 1. To create project, Go to Project and Select New.

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Experiment 3.f
OBJECTIVE : To find Impulse response of a first order and second order LTI system. 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,

Figure 3.f.1- Transform operation. If the input to the system is unit impulse i.e. x(n) = (n) then the output of the system is known as impulse response denoted by h(n) where,

h(n) = T[(n)]

Any arbitrary sequence x(n) can be represented as a weighted sum of discrete impulses. Now the system response for a linear system is given byy(n) = x(k) T[(n-k)] for k = - to . The response to the shifted impulse sequence can be denoted byh(n- k) = T[(n-k)] that implies that, y(n) = x(k) h(n-k) for k= - to .

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That implies that for a linear time-invariant system if the input sequence is x(n) and impulse response h(n) is given, we can fine output y(n) by using above equation, which is known as convolution sum(discussed later) and can be represented by y(n) = x(n) * h(n).

The general form of difference equation is,

Impulse response of second order difference equation: Program: #include<stdio.h> #define Order 2 #define Len 5 float h[Len] = {0.0,0.0,0.0,0.0,0.0},sum; void main() { int j, k; float a[Order+1] = {0.1311, 0.2622, 0.1311}; float b[Order+1] = {1, -0.7478, 0.2722}; for(j=0; j<Len; j++) { sum = 0.0; for(k=1; k<=Order; k++) { if ((j-k) >= 0)

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sum = sum+(b[k]*h[j-k]); } if (j<= Order) h[j] = a[j]-sum; else h[j] = -sum; printf (" %f ",h[j]); } } Output: 0.131100 0.360237 0.364799 0.174741 0.031373

For first order difference equation. Program: #include<stdio.h> #define Order 1 #define Len 5 float h[Len] = {0.0,0.0,0.0,0.0,0.0},sum; void main() { int j, k; float a[Order+1] = {0.1311, 0.2622}; float b[Order+1] = {1, -0.7478}; for(j=0; j<Len; j++) { sum = 0.0; for(k=1; k<=Order; k++) {

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if((j-k)>=0) sum = sum+(b[k]*h[j-k]); } if(j<=Order) h[j] = a[j]-sum; else h[j] = -sum; printf("%f ", j, h[j]); } } Output: 0.131100 0.360237 0.269385 0.201446 0.150641

Procedure to create new Project: 1. To create project, Go to Project and Select New.

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4. Enter the source code and save the file with .C extension.

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5. Right click on source, Select add files to project .. and Choose .C file Saved before.

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6. Right Click on libraries and select add files to Project.. and choose C:\CCStudio_v3.1\C6000\cgtools\lib\rts6700.lib and click open.

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a) Go to Project to Compile. b) Go to Project to Build. c) Go to Project to Rebuild All.

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8. Go to file and load program and load .out file into the board.

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9. Go to Debug and click on run to run the program.

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10. Observe the output in output window.

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Unit -3 Sample Viva Questions Q.1 Define digital data, and a DSP processor. Q.2 Specify some practical applications of DSP systems. Q.3 What are the advantages of digital systems over the analog systems? Q.4 What is the difference between microprocessor, microcontroller, and a DSP processor. Q.5 Explain the Components of a Typical DSP System. Q.6 What are the two basic architectures used in DSP processor. Q.7 Explain Von Neuman architecture. Q.8 Explain Harward architecture. Q.9What is the difference between Von Neuman and Harward architecture. Q.10 What is modified Harvard architecture. Q.11 Specify additional features in a DSP processor, making it superior from microprocessor/ microcontroller. Q.12 Explain bus structure in C67xx processor. Q.13 Explain the function of data buses and address bus. Q.14 Specify any four features of CPU in C67xx processor. Q.15 Explain the Arithmetic Logic Unit (ALU) in C67x DSP. Q.16 Define data storage in accumulator. Q.17 What is the functionality of a barrel shifter. Q.18 Explain MAC(Multiply add accumulate). Q.19 What is the advantage of using a MAC.

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Beyond Syllabus
EXPERIMENT NO-4 OBJECTIVE: Implementation of Transfer function of Log Gabor Filters. THEORY: A Gabor filter, named after Dennis Gabor, is a linear filter used for edge detection in images. In the spatial domain, a 2D Gabor filter is a 2D Gaussian function modulated by a sinusoidal plane wave. Means, a Gaussian function is multiplied to a sinusoidal wave. A generalized equation for Gabor function in 2D form is-

Gabor filters are a traditional choice for obtaining localized frequency information. These are filters having a Gaussian transfer function when viewed on the linear frequency scale. Bandwidth limitations of Gabor filters The maximum bandwidth of a Gabor filter is limited to approximately one octave and Gabor filters are not optimal if one is seeking broad spectral information. One cannot construct Gabor functions of arbitrarily wide bandwidth and still maintain a reasonably small DC component in the even-symmetric filter. This difficulty can be seen if we look at the transfer function of an even-symmetric Gabor filter in the frequency domain. The transfer function is the sum of two Gaussians centered at plus and minus the centre frequency. If the standard deviation of these Gaussians becomes more than about one third of the centre frequency the tails of the two Gaussians will start to overlap excessively at the origin, resulting in a nonzero DC component, as shown in figure.1. At

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the limiting situation where the centre frequency is equal to three standard deviations, the bandwidth will be approximately one octave.

Figure-4.a.1- Transfer function of a high bandwidth even-symmetric Gabor

The Log-Gabor Filter An alternative to the Gabor function is the log-Gabor function proposed by Field [1987]. Field suggests that natural images are better coded by filters that have Gaussian transfer functions when viewed on the logarithmic frequency scale. (Gabor functions have Gaussian transfer functions when viewed on the linear frequency scale). On the linear frequency scale the log-Gabor function has a transfer function of the form G(w) = e (-log(w/wo)2) / ( 2 (log(k/wo)2 ) where wo is the filter's centre frequency. To obtain constant shape ratio filters the term k/wo must also be held constant for varying wo. For example, a k/wo value of .74 will result in a filter bandwidth of approximately one octave, .55 will result in two octaves, and .41 will produce three octaves. There are two important characteristics to note. Firstly, log-Gabor functions, by definition, always have no DC component, and secondly, the transfer function of the log Gabor function has an extended tail at the high frequency end.

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Figure-4.a.2- An example of a log-Gabor transfer function viewed on both linear and logarithmic frequency scales.

In spatial domain, their appearance is similar to Gabor functions though their shape becomes much `sharper' as the bandwidth is increased. The shapes of log Gabor and Gabor functions are almost identical for bandwidths less than one octave. Applications of log-Gabor filtersWe have studied that compared with Gabor filters, Log-Gabor filters have two important characteristics. Firstly, Log-Gabor filters, always have no DC component. Secondly, the transfer function of the Log-Gabor filters has an extended tail at the high frequency end. Fields studies indicate that natural images have amplitude spectra that fall off at approximately 1/f. To encode images having such spectral characteristics one should use filters having the similar spectra. Field suggests that Log-Gabor filters, having extended tails, should be able to encode natural images more efficiently than ordinary Gabor filters, which would over-represent the low frequency components and under-represent the high frequency components in any encoding. Log-Gabor filters are widely used in the field of image processing like, texture Analysis /synthesis, contour extraction , segmentation ,object recognition, image analysis and compression , movement estimation or image restoration characteristics: - Directionality: filters can be rotated to discriminate spectral features in multiple
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directions. - Complex modulation (odd/even phases): effective for analyzing different phasedfeatures like abrupt ridges or edges.

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Sample Viva Questions Q.1 Define a filter. Q.2 What is a transfer function of a filter? Q.3 Classify filters according to the range of frequencies it pass. Also specify their transfer functions. Q.4 What do you mean log frequency scale? Q.5 How does the log frequency scale differ from linear frequency scale? Q.6 What is a 2-D Gabor filter? State its transfer function. Q.7 What is the bandwidth limitation of a Gabor filter. Q.8 State transfer function of a log Gabor filter . Q.9 State two important characteristics of transfer function of log Gabor filter. Q.10 How do you differentiate a log Gabor filter from a Gabor filter? Q.11 What are the advantages of Gabor filter over log Gabor filter? Q.12 What is the range of bandwidth for which log Gabor filter and Gabor filter have same characteristics. Q.13 What property of log Gabor filter transfer function make them suitable to encode natural images? Q.14 What do you mean by directionality of log Gabor filter. Q.15 State some applications of log Gabor filter?

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EXPERIMENT NO-5 OBJECTIVE: Removal of noise from image signals using various spatial filtering operations. Theory: Here we will study about the concept of an image, various noise signals contaminating the image, and spatial filtering techniques to remove that noise. Concept of image An image may be defined as a two-dimensional function,f(x, y), where x and y are spatial (plane) coordinates, also called as a pixel, and the amplitude of f at any pair of coordinates (x, y) is called the intensity or gray level of the image at that point(pixel). When x, y and the amplitude values of f are all finite, discrete quantities, we call the image a digital image. Thus, a rectangular black-and-white image is a matrix where each entry represents an intensity of gray at that spot and ranges from 0 to 255, 0 being the darkest spot and 255 being the brightest spot. Noise in images In practice, an image may be degraded by various types and forms of noise. However, the most common type of noise is the additive one. The principal source of noise in digital images arise during image acquisition (digitization) or transmission. The performance of imaging sensors is affected by a variety of factors, such as the environmental conditions during image acquisition, and by the quality of the sensing elements themselves. Various additive noises in image are Gaussian noise, Uniform noise,and Impulse (salt-and-pepper) noise. Each noise component of the model may be considered random variables, characterized by its specific probability density function (PDF). Two most common occurring additive noises, Impulse and Gaussian noise are shown in fig.-3.

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Figure-4.b.1- PDF of (a)Gaussian noise

(b)Impulse(Salt and Pepper noise)

MATLAB command for adding noise Here we discuss important command in MATLAB which are used to add noise to image signals. ImnoiseThis command adds noise to an image. Its syntax for above discussed noises areJ = imnoise (I ,' gaussian' ,m ,v) J = imnoise (I, 'salt & pepper', d) WhereJ = Noisy image; I = Input image; 'gaussian' = Gaussian white noise with constant mean and variance. The default is zero mean noise with 0.01 variance.; m, v = mean (m) and variance (v) to the image I. 'salt & pepper' = On and off pixels.

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d = d is the noise density. The default for d is 0.05. Filtering methods to remove noiseSpatial filtering is the method of choice in situations when only additive noise is present. Some neighborhood operations work with the values of the image pixels in the neighborhood and the corresponding values of a sub-image that has the same dimensions as the neighborhood. The sub-image is called a filter, or window. The mechanics of spatial filtering consists simply of moving the filter window from point to point in an image. At each point (x, y), the response ofthe filter at that point is calculated using a predefined relationship. In this section we see some noise reduction spatial filters. Let us assume noisy image as g . S represent the set of coordinates in a rectangular sub image window of size m x n, centered at point (x, y) . f represent the denoised image. Linear(Mean) filtering A mean filter acts on an image by smoothing it; that is, it reduces the intensity variation between adjacent pixels. It is preferable for impulse noise. The mean filter is nothing but a simple sliding window spatial filter that replaces the center value in the window with the average of all the neighboring pixel values including itself. It is implemented with a convolution mask, which provides a result that is a weighted sum of the values of a pixel and its neighbors. It is also called a mean filter. The mask or kernel is a square. Linear filter is of three types1). Arithmetic mean filter- The arithmetic mean filtering process computes the average value of the corrupted image g(x, y) in the area defined by S .The value of the denoised image f at any point (x, y) is simply the arithmetic mean computed using the pixels in the region defined by S. 2). Geometric mean filter- An image denoised using a geometric mean filter is given by each denoised pixel is the geometric mean of the pixels in the subimage window. 3). Harmonic mean filter- An image denoised using a harmonicic mean filter is given by each denoised pixel is the harmonic mean of the pixels in the subimage window.

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Median filter

The median filter calculates median of the pixel values in the window is computed, and the center pixel of the window is replaced with the computed median. Thus , Median filter, replaces the value of a pixel by the median of the gray levels in the neighborhood of that pixel. Median filtering is done by, first sorting all the pixel values from the surrounding neighborhood into numerical order and then replacing the pixel being considered with the middle pixel value.

Adaptive Filtering The fundamental difference between the mean filter and the adaptive filter lies in the fact that the filter mask varies after each iteration in the adaptive filter. Adaptive filters are capable of denoising images that have abrupt changes in intensity. Mean and variance are the parameters on which to base an adaptive filler because they are quantities closely related to the appearance of an image.

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Sample Viva Questions Q.1 How we represent an image mathematically in image processing? Q.2 What is a pixel? Q.3 Define grey level of a pixel? Q.4 What are the principal source of noise in digital images? Q.5 What is an additive noise? Q.6 State any three additive noise. How are they characterized or differ? Q.7 Which MATLAB command is used to add noise in image? Give its syntax. Q.8 What is mean and variance of noise in an image? Q.9 Define spatial filtering? Q.10 What is the function of window in spatial filtering? Q.11 Define linear filtering. Q.12 What are the various types of linear filtering? Define each. Q.13 Define median filtering. How it differ from adaptive filtering? Q.14 Give an example of adaptive filtering? Q.15 What is multiplicative noise? How it differ from additive noise?

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EXPERIMENT NO-6

OBJECTIVE: Write a program in MATLAB to Remove noise from an image signal using various spatial filtering operations.

OBJECTIVE - Removal of noise from an image signal using various spatial filtering operations.

Program1. clc; 2. clear all; 3. close all; 4. I = imread('cameraman.tif'); 5. figure(1),imshow(I); 6. J = imnoise(I,'salt & pepper',0.04); 7. figure(2),imshow(J); 8. K = filter2(fspecial('average',3),J)/255; 9. figure(3),imshow(K); 10. L = medfilt2(J,[3 3]); 11. figure(4), imshow(L); 12. M = imnoise(I,'gaussian',0,0.01); 13. figure(5),imshow(M); 14. N = wiener2(J,[5 5]); 15. figure(6),imshow(N);

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Output-

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Sample Viva Questions

Q.1What is the function of clc, clear all, close all commands in MATLAB? Q.2 What is the command for reading an image in MATLAB? Q.3 What is the command for displaying an image in MATLAB? Q.4 What is the command for adding noise in image? Q.5Which are the noises added in above program in an image? Also give syntax to add these noises to image. Q.6 What is the function of command fspecial? Q.7 Write syntax for applying mean(average) filter on an image in MATLAB? Q.8 What is the command for applying a median filter on a noisy image? Q.9 What do you mean by mean and variance of Gaussian noise? Q.10 What do you mean by standard deviation of Impulse noise? Q.11 Why do we call an Impulse noise, a salt and pepper noise? Q12 What is the standard size of mean and median filter we are applying on image? Q.13What is the basic difference between mean and median filter? Q.14 Which type of filter, the Weiner filter is? Q.15What basic operation does the Weiner filter apply on image?

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EXPERIMENT NO-7 OBJECTIVE: To study Wavelets, wavelet transform and wavelet decomposition. Theory: Need of Wavelet Transform Fourier Transforms gives the us spectral/frequency content of the signal, but it gives no information regarding where in time those spectral components appear. Therefore, Fourier Transforms are not suitable technique for non-stationary signals. Signals whose frequency content change in time are called non-stationary signals. Means all frequency contents does not exist all the time, or the existing frequency changes linearly in time. Ex chirp signal. Thus, the Fourier Transform gives what frequency components (spectral components) exist in the signal. Nothing more, nothing less. Introduction to wavelet transformThe term wavelet means a small wave . The smallness refers to the condition that this (window) function is of finite length (compactly supported). The wave refers to the condition that this function is oscillatory. A wavelet is a waveform of effectively limited duration that has an average value of zero. These wavelets form the basis functions for wavelet transform. Compare wavelets with sine waves, which are the basis of Fourier analysis. Sinusoids do not have limited duration they extend from minus to plus Infinity. Fourier analysis consists of breaking up a signal into sine waves of various frequencies. Similarly, wavelet analysis is the breaking up of a signal into shifted and scaled versions of the original (or mother) wavelet. Signals with sharp changes in frequencies might be better analyzed with an irregular wavelet than with a smooth sinusoid.

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Continuous Wavelet Transform The continuous wavelet transform is defined as follows-

In the CWT, the analyzing function is a wavelet, . The CWT compares the signal to shifted and compressed or stretched versions of a wavelet. Stretching or compressing a function is collectively referred to as dilation or scaling. By continuously varying the values of the scale parameter, tau, and the position parameter, s , you obtain the cwt coefficients CWT(tau, s). x(t) is the signal to be analyzed. For convenience, the procedure will be started from scale s=1 and will continue for the increasing values of s , i.e., the analysis will start from high frequencies and proceed towards low frequencies. This first value of s will correspond to the most compressed wavelet. As the value of s is increased, the wavelet will dilate. The wavelet is placed at the beginning of the signal at the point which corresponds to time=0. The wavelet function at scale ``1'' is multiplied by the signal and then integrated over all times. The result of the integration is then multiplied by the constant number 1/sqrt{s} . This multiplication is for energy normalization purposes so that the transformed signal will have the same energy at every scale. The wavelet at scale s=1 is then shifted towards the right by tau amount to the location tau , and the above equation is computed to get the transform value at tau, s=1 in the time-frequency plane. This procedure is repeated until the wavelet reaches the end of the signal. The above procedure is repeated for every value of s. When the process is completed for all desired values of s, the CWT of the signal has been calculated.

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The Discrete Wavelet Transform Basic scaling function in Discrete Wavelet Transform are given by -

Basic scaling function in Discrete Wavelet Transform are given by-

Where, j,k are scaling and translation indices. If the function being expanded is a sequence of numbers, like a discrete function f(x), the resulting coefficients are called the discrete wavelet transform (DWT) of f(x). The DWT transform pair is given by-

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Here, f(x) , j 0,k ( x) , j , k ( x) are functions of the discrete variable x = 0, 1,2,..., M -1. The coefficients defined as W ( jo , k ) , W ( j , k ) are usually called approximation and detail coefficients, respectively, This is because the first sum in above equation uses scaling functions to provide an approximation of f(x) at scale j0. For each higher scale > j0 in the second sum, a finer resolution functiona sum of waveletsis added to the approximation to provide increasing detail. Normally, we let j0 = 0 and select M to be a power of 2 (i.e., M = 2 j ) . As, we have seen, Wavelet Transform decomposes a signal into approximation and detail coefficients. The approximations are the high-scale, low-frequency components of the signal. The details are the low-scale, high-frequency components. The filtering process, at its most basic level, looks like Figure 4.

Figure 4.d.1- Wavelet Transform decomposition of a signal

The original signal, S, passes through two complementary filters and emerges as two signals. If we actually perform this operation on a real digital signal, we wind up with twice as much data as we started with. Suppose, for instance, that the original signal S consists of 1000 samples of data. Then the resulting signals will each have 1000 samples, for a total of 2000. So, to reduce its size we down sample the both resulting low passed and high passed filtered signal, to reduce its length to half.

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For many signals, the low-frequency content is the most important part. It is the average of original signal. The high-frequency content, on the other hand, imparts finer details of the signal, like glitches, and high frequency components. The decomposition process can be iterated, with successive approximations Being decomposed in turn, so that one signal is broken down into many lower Resolution components. This is called the wavelet decomposition tree. Looking at a signals wavelet decomposition tree can yield valuable information. Since the analysis process is iterative, in theory it can be continued indefinitely. Figure- 5 showes the wavelet decomposition process, and decomposition tree for a signal.

Figure 4.d.2-Wavelet decomposition process, and decomposition tree for a signal.

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Sample Viva Questions Q.1 Define wavelet. Q.2 State any three wavelets. Q.3 What is a transform? Q.4Why we transform a signal into other domain? Q.5 What are basis function for Fourier and wavelet transforms? Q.6 What is the basic difference between wavelet and fourier transform? Q.7 What is the advantage of using Wavelet transform, over Fourier Transform? Q.8 Explain continuous wavelet transform with the equation. Q.9 What do you mean by scale and transition in wavelet transform? Q.10 What are scaling function and wavelet functions? Q.11 What are the parameters used in scaling and wavelet functions for scaling and n translation? Q.12 What do you mean by wavelet decomposition? Q.13 What are the levels of wavelet decomposition? Q.14 In context to the image, what do approximation and detail coefficients represent ? Q.15 State any two applications of wavelet decomposition in image processing.

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EXPERIMENT NO-4.e OBJECTIVE: Write a program in MATLAB to decompose an image signal using Discrete Wavelet Transform.

1. clear all; 2. clc; 3. f = imread('cameraman.tif'); 4. F = im2double(f); 5. [C,S] = wavedec2(F,1,'haar'); 6. A = appcoef2(C,S,'haar',1); 7. D1 = detcoef2('h', C,S,1); 8. D2 = detcoef2('v', C,S,1); 9. D3 = detcoef2('d', C,S,1); 10. c = cell(2); 11. c{1,1} = A; 12. c{1,2} = D1; 13. c{2,1} = D2; 14. c{2,2} = D3; 15. c = cell2mat(c); 16. c1 = im2uint8(c); 17. imshow(c1);

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Sample Viva Questions Q.1 Write the MATLAB command for wavelet decomposition with syntax. Q.2 What effect does the level of decomposition causes on output? Q.3 What are C, S matrix called? Also describe them in brief? Q.4 Write the command to read an image in MATLAB. Q.5 Specify different data-types in MATLAB. Q.6 What do you understand by data type uint8? Q.7 What do you understand by data type double? Q.8 In which data type the image is saved in MATLAB? Q.9 In which data type we need to change image to process it? Q.10 What is the command for the conversion of image? Give syntax. Q.11What change will the conversion cause in image variable? Q.12 State any two types of wavelets used in image decomposition? Q.13 What are approximate and detail coefficients. Q.14 What are commands to generate approximate and detail coefficients Q.15 What are cells? What is the command to convert cell data to matrix data.

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References:

1. Gonzales R.C., Woods R.E. Digital Image Processing, (2ndedition). Prentice Hall Publication. 2http://www.mathworks.com/access/helpdesk/help/pdf_doc/wavelet/wavelet_ug.pdf . 3.http://www.mathworks.com/access/helpdesk/help/pdf_doc/images/images_tb.pdf 4.www.csse.uwa.edu.au/~pk/research/matlabfns/.../convexpl.html

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