Download Anna University B-Tech ECE 5th Sem Digital Signal Processing DSP Lab Manual Question Paper

Download Anna University B.Tech (Bachelor of Technology) ECE (Electronics And Communications Engineering) 5th Sem Digital Signal Processing DSP Lab Manual Question Paper.




?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
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?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
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?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







FirstRanker.com - FirstRanker's Choice



?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







FirstRanker.com - FirstRanker's Choice



?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
FirstRanker.com - FirstRanker's Choice



?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





FirstRanker.com - FirstRanker's Choice



?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
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?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
7 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

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?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
7 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
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?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
7 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

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?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
7 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
FirstRanker.com - FirstRanker's Choice



?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
7 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
FirstRanker.com - FirstRanker's Choice



?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
7 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
12 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
FirstRanker.com - FirstRanker's Choice



?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
7 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
12 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
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?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
7 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
12 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
FirstRanker.com - FirstRanker's Choice



?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
7 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
12 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

FirstRanker.com - FirstRanker's Choice



?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
7 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
12 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
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DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
7 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
12 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
FirstRanker.com - FirstRanker's Choice



?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
7 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
12 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
18 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
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?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
7 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
12 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
18 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
19 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

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?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
7 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
12 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
18 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
19 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

20 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
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DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
7 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
12 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
18 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
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5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

20 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
21 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.5 DESIGN OF IIR FILTERS
Aim:
To write a program to design and obtain the response of the given IIR filter using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
A digital filter, the derivation of the digital filter transfer function requires three steps.
1. Map the desired digital filter specification into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
The various parameters in the figure are,
?p = Pass band frequency in radians
?s = Stop band frequency in radians
?c = 3-db cutoff frequency in radians
? = Parameter specifying allowable pass band
? = Parameter specifying allowable stop band
Design of IIR filters from analog filters:
There are several methods that can be used to design digital filters having an infinite duration unit
sample response. The techniques described are all based on converting an analog filter into digital filter. If
the conversion technique is to be effective, it should posses the following desirable properties.
The j ?-axes in the s-plane should map into the unit circle in the z-plane. Thus there will be a
direct relationship between the two frequency variables in the two domains.
The left-half plane of the s-plane should map into inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
Algorithm:
Butterworth low pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
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DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
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5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
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6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
18 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
19 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

20 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
21 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.5 DESIGN OF IIR FILTERS
Aim:
To write a program to design and obtain the response of the given IIR filter using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
A digital filter, the derivation of the digital filter transfer function requires three steps.
1. Map the desired digital filter specification into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
The various parameters in the figure are,
?p = Pass band frequency in radians
?s = Stop band frequency in radians
?c = 3-db cutoff frequency in radians
? = Parameter specifying allowable pass band
? = Parameter specifying allowable stop band
Design of IIR filters from analog filters:
There are several methods that can be used to design digital filters having an infinite duration unit
sample response. The techniques described are all based on converting an analog filter into digital filter. If
the conversion technique is to be effective, it should posses the following desirable properties.
The j ?-axes in the s-plane should map into the unit circle in the z-plane. Thus there will be a
direct relationship between the two frequency variables in the two domains.
The left-half plane of the s-plane should map into inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
Algorithm:
Butterworth low pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
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Butterworth high pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band stop filter
1. Get the pass band and stop band ripples
2. Get the pass band and stop band edge frequencies
3. Get the sampling frequencies
4. Calculate the order of the filter
5. Find the filter co-efficient
6. Draw the magnitude and phase response
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the IIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Implement IIR filter for given specifications
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DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
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PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







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PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
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EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





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Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
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5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

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Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
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Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

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Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
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6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
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4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
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5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

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Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
21 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.5 DESIGN OF IIR FILTERS
Aim:
To write a program to design and obtain the response of the given IIR filter using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
A digital filter, the derivation of the digital filter transfer function requires three steps.
1. Map the desired digital filter specification into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
The various parameters in the figure are,
?p = Pass band frequency in radians
?s = Stop band frequency in radians
?c = 3-db cutoff frequency in radians
? = Parameter specifying allowable pass band
? = Parameter specifying allowable stop band
Design of IIR filters from analog filters:
There are several methods that can be used to design digital filters having an infinite duration unit
sample response. The techniques described are all based on converting an analog filter into digital filter. If
the conversion technique is to be effective, it should posses the following desirable properties.
The j ?-axes in the s-plane should map into the unit circle in the z-plane. Thus there will be a
direct relationship between the two frequency variables in the two domains.
The left-half plane of the s-plane should map into inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
Algorithm:
Butterworth low pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
22 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Butterworth high pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band stop filter
1. Get the pass band and stop band ripples
2. Get the pass band and stop band edge frequencies
3. Get the sampling frequencies
4. Calculate the order of the filter
5. Find the filter co-efficient
6. Draw the magnitude and phase response
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the IIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Implement IIR filter for given specifications
23 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

2. Plot the response of the same.
Applications:
1. Implementation of filter circuits
2. Speech processing and video processing
3. Radio transmission
4. Filters in various industrial applications


1. Define ? IIR Filter
2. List the steps to design a digital filter using impulse invariant method.
3. What are the disadvantages of impulse invariant method?
4. Differentiate recursive & non-recursive systems.
5. List the parameters obtained from the Chebyshev filter specifications.
6. What is warping effect? Where we found this effect? How can we eliminate warping effect
7. What is the difference between type I & type II Chebyshev filters?
8. Define ? Cut-off frequency
9. Compare direct form I and direct form II realization of IIR filter.
10. List the digital techniques used to convert analog filter to digital filter.
11. List the properties of Butterworth low pass filter.
12. What is the role of delay element in filter design?
13. List the properties of Chebyshev low pass filter.
14. What are the properties of bilinear transformations?
15. Distinguish between the frequency responses of Chebyshev type I filter for N odd and N even.
16. Write the expression of the order of the Butterworth filter.
17. Define ? Prewarping
18. List the steps to design digital filter from analog filter.
19. How many number of additions, multiplications and memory locations are required to realize a
digital system in IIR filter?
20. Write the expression of the order of the Chebyshev filter.
Viva ?voce
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DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





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Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
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5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

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Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
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Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
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6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
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4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
19 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

20 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
21 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.5 DESIGN OF IIR FILTERS
Aim:
To write a program to design and obtain the response of the given IIR filter using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
A digital filter, the derivation of the digital filter transfer function requires three steps.
1. Map the desired digital filter specification into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
The various parameters in the figure are,
?p = Pass band frequency in radians
?s = Stop band frequency in radians
?c = 3-db cutoff frequency in radians
? = Parameter specifying allowable pass band
? = Parameter specifying allowable stop band
Design of IIR filters from analog filters:
There are several methods that can be used to design digital filters having an infinite duration unit
sample response. The techniques described are all based on converting an analog filter into digital filter. If
the conversion technique is to be effective, it should posses the following desirable properties.
The j ?-axes in the s-plane should map into the unit circle in the z-plane. Thus there will be a
direct relationship between the two frequency variables in the two domains.
The left-half plane of the s-plane should map into inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
Algorithm:
Butterworth low pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
22 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Butterworth high pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band stop filter
1. Get the pass band and stop band ripples
2. Get the pass band and stop band edge frequencies
3. Get the sampling frequencies
4. Calculate the order of the filter
5. Find the filter co-efficient
6. Draw the magnitude and phase response
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the IIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Implement IIR filter for given specifications
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2. Plot the response of the same.
Applications:
1. Implementation of filter circuits
2. Speech processing and video processing
3. Radio transmission
4. Filters in various industrial applications


1. Define ? IIR Filter
2. List the steps to design a digital filter using impulse invariant method.
3. What are the disadvantages of impulse invariant method?
4. Differentiate recursive & non-recursive systems.
5. List the parameters obtained from the Chebyshev filter specifications.
6. What is warping effect? Where we found this effect? How can we eliminate warping effect
7. What is the difference between type I & type II Chebyshev filters?
8. Define ? Cut-off frequency
9. Compare direct form I and direct form II realization of IIR filter.
10. List the digital techniques used to convert analog filter to digital filter.
11. List the properties of Butterworth low pass filter.
12. What is the role of delay element in filter design?
13. List the properties of Chebyshev low pass filter.
14. What are the properties of bilinear transformations?
15. Distinguish between the frequency responses of Chebyshev type I filter for N odd and N even.
16. Write the expression of the order of the Butterworth filter.
17. Define ? Prewarping
18. List the steps to design digital filter from analog filter.
19. How many number of additions, multiplications and memory locations are required to realize a
digital system in IIR filter?
20. Write the expression of the order of the Chebyshev filter.
Viva ?voce
24 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt.No.6 MULTIRATE FILTERS
Aim
To design multirate FIR filters using interpolation and decimation and to plot the impulse responses and
the magnitude responses for all designs.
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the multirate filters was designed using interpolation and decimation for the given input.
Outcome:
After finishing this experiment the students are able to design linear-phase FIR multirate filters and do
interpolation and decimation of the signals
Applications:
1. Sampling rate conversion systems
2. Communication systems
3. Design of phase shifters
4. Antenna arrays





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DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
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5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

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Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
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Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
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6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
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4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
19 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

20 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
21 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.5 DESIGN OF IIR FILTERS
Aim:
To write a program to design and obtain the response of the given IIR filter using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
A digital filter, the derivation of the digital filter transfer function requires three steps.
1. Map the desired digital filter specification into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
The various parameters in the figure are,
?p = Pass band frequency in radians
?s = Stop band frequency in radians
?c = 3-db cutoff frequency in radians
? = Parameter specifying allowable pass band
? = Parameter specifying allowable stop band
Design of IIR filters from analog filters:
There are several methods that can be used to design digital filters having an infinite duration unit
sample response. The techniques described are all based on converting an analog filter into digital filter. If
the conversion technique is to be effective, it should posses the following desirable properties.
The j ?-axes in the s-plane should map into the unit circle in the z-plane. Thus there will be a
direct relationship between the two frequency variables in the two domains.
The left-half plane of the s-plane should map into inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
Algorithm:
Butterworth low pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
22 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Butterworth high pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band stop filter
1. Get the pass band and stop band ripples
2. Get the pass band and stop band edge frequencies
3. Get the sampling frequencies
4. Calculate the order of the filter
5. Find the filter co-efficient
6. Draw the magnitude and phase response
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the IIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Implement IIR filter for given specifications
23 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

2. Plot the response of the same.
Applications:
1. Implementation of filter circuits
2. Speech processing and video processing
3. Radio transmission
4. Filters in various industrial applications


1. Define ? IIR Filter
2. List the steps to design a digital filter using impulse invariant method.
3. What are the disadvantages of impulse invariant method?
4. Differentiate recursive & non-recursive systems.
5. List the parameters obtained from the Chebyshev filter specifications.
6. What is warping effect? Where we found this effect? How can we eliminate warping effect
7. What is the difference between type I & type II Chebyshev filters?
8. Define ? Cut-off frequency
9. Compare direct form I and direct form II realization of IIR filter.
10. List the digital techniques used to convert analog filter to digital filter.
11. List the properties of Butterworth low pass filter.
12. What is the role of delay element in filter design?
13. List the properties of Chebyshev low pass filter.
14. What are the properties of bilinear transformations?
15. Distinguish between the frequency responses of Chebyshev type I filter for N odd and N even.
16. Write the expression of the order of the Butterworth filter.
17. Define ? Prewarping
18. List the steps to design digital filter from analog filter.
19. How many number of additions, multiplications and memory locations are required to realize a
digital system in IIR filter?
20. Write the expression of the order of the Chebyshev filter.
Viva ?voce
24 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt.No.6 MULTIRATE FILTERS
Aim
To design multirate FIR filters using interpolation and decimation and to plot the impulse responses and
the magnitude responses for all designs.
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the multirate filters was designed using interpolation and decimation for the given input.
Outcome:
After finishing this experiment the students are able to design linear-phase FIR multirate filters and do
interpolation and decimation of the signals
Applications:
1. Sampling rate conversion systems
2. Communication systems
3. Design of phase shifters
4. Antenna arrays





25 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00



1. What is the importance of decimation for a given signal/sequence?
2. What is aliasing? What is the condition to avoid aliasing for sampling?
3. Define ? Decimation
4. Define ? Interpolation
5. What is multirate signal processing?
6. Name the process of converting a signal from a given rate to a different rate
7. State the formula for poly phase interpolation using z-transform.
8. List the applications of multirate signal processing.
9. List the methods used in sampling for converting a digital signal
10. List the advantages of multirate processing.
11. What is up sampling?
12. List the applications of multirate DSP.
13. What is the need for antialiasing filter prior to down sampling?
14. List the applications of adaptive filters
15. What are the sections in Quadrature mirror filter bank?
16. What is down sampling?
17. What is the need for anti-imaging filter after up sampling a signal?
18. State the formula for poly phase decimation using z-transform.
19. What is anti-aliasing filter?
20. Define ? Sampling theorem







Viva ?voce
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DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





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Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
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5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

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Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
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Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
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6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
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4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
19 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

20 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
21 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.5 DESIGN OF IIR FILTERS
Aim:
To write a program to design and obtain the response of the given IIR filter using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
A digital filter, the derivation of the digital filter transfer function requires three steps.
1. Map the desired digital filter specification into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
The various parameters in the figure are,
?p = Pass band frequency in radians
?s = Stop band frequency in radians
?c = 3-db cutoff frequency in radians
? = Parameter specifying allowable pass band
? = Parameter specifying allowable stop band
Design of IIR filters from analog filters:
There are several methods that can be used to design digital filters having an infinite duration unit
sample response. The techniques described are all based on converting an analog filter into digital filter. If
the conversion technique is to be effective, it should posses the following desirable properties.
The j ?-axes in the s-plane should map into the unit circle in the z-plane. Thus there will be a
direct relationship between the two frequency variables in the two domains.
The left-half plane of the s-plane should map into inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
Algorithm:
Butterworth low pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
22 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Butterworth high pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band stop filter
1. Get the pass band and stop band ripples
2. Get the pass band and stop band edge frequencies
3. Get the sampling frequencies
4. Calculate the order of the filter
5. Find the filter co-efficient
6. Draw the magnitude and phase response
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the IIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Implement IIR filter for given specifications
23 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

2. Plot the response of the same.
Applications:
1. Implementation of filter circuits
2. Speech processing and video processing
3. Radio transmission
4. Filters in various industrial applications


1. Define ? IIR Filter
2. List the steps to design a digital filter using impulse invariant method.
3. What are the disadvantages of impulse invariant method?
4. Differentiate recursive & non-recursive systems.
5. List the parameters obtained from the Chebyshev filter specifications.
6. What is warping effect? Where we found this effect? How can we eliminate warping effect
7. What is the difference between type I & type II Chebyshev filters?
8. Define ? Cut-off frequency
9. Compare direct form I and direct form II realization of IIR filter.
10. List the digital techniques used to convert analog filter to digital filter.
11. List the properties of Butterworth low pass filter.
12. What is the role of delay element in filter design?
13. List the properties of Chebyshev low pass filter.
14. What are the properties of bilinear transformations?
15. Distinguish between the frequency responses of Chebyshev type I filter for N odd and N even.
16. Write the expression of the order of the Butterworth filter.
17. Define ? Prewarping
18. List the steps to design digital filter from analog filter.
19. How many number of additions, multiplications and memory locations are required to realize a
digital system in IIR filter?
20. Write the expression of the order of the Chebyshev filter.
Viva ?voce
24 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt.No.6 MULTIRATE FILTERS
Aim
To design multirate FIR filters using interpolation and decimation and to plot the impulse responses and
the magnitude responses for all designs.
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the multirate filters was designed using interpolation and decimation for the given input.
Outcome:
After finishing this experiment the students are able to design linear-phase FIR multirate filters and do
interpolation and decimation of the signals
Applications:
1. Sampling rate conversion systems
2. Communication systems
3. Design of phase shifters
4. Antenna arrays





25 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00



1. What is the importance of decimation for a given signal/sequence?
2. What is aliasing? What is the condition to avoid aliasing for sampling?
3. Define ? Decimation
4. Define ? Interpolation
5. What is multirate signal processing?
6. Name the process of converting a signal from a given rate to a different rate
7. State the formula for poly phase interpolation using z-transform.
8. List the applications of multirate signal processing.
9. List the methods used in sampling for converting a digital signal
10. List the advantages of multirate processing.
11. What is up sampling?
12. List the applications of multirate DSP.
13. What is the need for antialiasing filter prior to down sampling?
14. List the applications of adaptive filters
15. What are the sections in Quadrature mirror filter bank?
16. What is down sampling?
17. What is the need for anti-imaging filter after up sampling a signal?
18. State the formula for poly phase decimation using z-transform.
19. What is anti-aliasing filter?
20. Define ? Sampling theorem







Viva ?voce
26 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.7 EQUALIZATION

Aim
To write MATLAB program to design adaptive channel equalization using LMS algorithm
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the adaptive channel equalization was designed using LMS algorithm.
Outcome:
After finishing this experiment the students are able to design adaptive channel equalization using LMS
algorithm.
Applications:
1. Echo cancellation in telecommunication systems
2. Digital radio communication systems
3. Channel equalization for dispersive channels
4. Signal processing and control applications








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DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
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5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

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Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
12 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
18 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
19 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

20 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
21 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.5 DESIGN OF IIR FILTERS
Aim:
To write a program to design and obtain the response of the given IIR filter using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
A digital filter, the derivation of the digital filter transfer function requires three steps.
1. Map the desired digital filter specification into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
The various parameters in the figure are,
?p = Pass band frequency in radians
?s = Stop band frequency in radians
?c = 3-db cutoff frequency in radians
? = Parameter specifying allowable pass band
? = Parameter specifying allowable stop band
Design of IIR filters from analog filters:
There are several methods that can be used to design digital filters having an infinite duration unit
sample response. The techniques described are all based on converting an analog filter into digital filter. If
the conversion technique is to be effective, it should posses the following desirable properties.
The j ?-axes in the s-plane should map into the unit circle in the z-plane. Thus there will be a
direct relationship between the two frequency variables in the two domains.
The left-half plane of the s-plane should map into inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
Algorithm:
Butterworth low pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
22 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Butterworth high pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band stop filter
1. Get the pass band and stop band ripples
2. Get the pass band and stop band edge frequencies
3. Get the sampling frequencies
4. Calculate the order of the filter
5. Find the filter co-efficient
6. Draw the magnitude and phase response
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the IIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Implement IIR filter for given specifications
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2. Plot the response of the same.
Applications:
1. Implementation of filter circuits
2. Speech processing and video processing
3. Radio transmission
4. Filters in various industrial applications


1. Define ? IIR Filter
2. List the steps to design a digital filter using impulse invariant method.
3. What are the disadvantages of impulse invariant method?
4. Differentiate recursive & non-recursive systems.
5. List the parameters obtained from the Chebyshev filter specifications.
6. What is warping effect? Where we found this effect? How can we eliminate warping effect
7. What is the difference between type I & type II Chebyshev filters?
8. Define ? Cut-off frequency
9. Compare direct form I and direct form II realization of IIR filter.
10. List the digital techniques used to convert analog filter to digital filter.
11. List the properties of Butterworth low pass filter.
12. What is the role of delay element in filter design?
13. List the properties of Chebyshev low pass filter.
14. What are the properties of bilinear transformations?
15. Distinguish between the frequency responses of Chebyshev type I filter for N odd and N even.
16. Write the expression of the order of the Butterworth filter.
17. Define ? Prewarping
18. List the steps to design digital filter from analog filter.
19. How many number of additions, multiplications and memory locations are required to realize a
digital system in IIR filter?
20. Write the expression of the order of the Chebyshev filter.
Viva ?voce
24 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt.No.6 MULTIRATE FILTERS
Aim
To design multirate FIR filters using interpolation and decimation and to plot the impulse responses and
the magnitude responses for all designs.
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the multirate filters was designed using interpolation and decimation for the given input.
Outcome:
After finishing this experiment the students are able to design linear-phase FIR multirate filters and do
interpolation and decimation of the signals
Applications:
1. Sampling rate conversion systems
2. Communication systems
3. Design of phase shifters
4. Antenna arrays





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1. What is the importance of decimation for a given signal/sequence?
2. What is aliasing? What is the condition to avoid aliasing for sampling?
3. Define ? Decimation
4. Define ? Interpolation
5. What is multirate signal processing?
6. Name the process of converting a signal from a given rate to a different rate
7. State the formula for poly phase interpolation using z-transform.
8. List the applications of multirate signal processing.
9. List the methods used in sampling for converting a digital signal
10. List the advantages of multirate processing.
11. What is up sampling?
12. List the applications of multirate DSP.
13. What is the need for antialiasing filter prior to down sampling?
14. List the applications of adaptive filters
15. What are the sections in Quadrature mirror filter bank?
16. What is down sampling?
17. What is the need for anti-imaging filter after up sampling a signal?
18. State the formula for poly phase decimation using z-transform.
19. What is anti-aliasing filter?
20. Define ? Sampling theorem







Viva ?voce
26 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.7 EQUALIZATION

Aim
To write MATLAB program to design adaptive channel equalization using LMS algorithm
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the adaptive channel equalization was designed using LMS algorithm.
Outcome:
After finishing this experiment the students are able to design adaptive channel equalization using LMS
algorithm.
Applications:
1. Echo cancellation in telecommunication systems
2. Digital radio communication systems
3. Channel equalization for dispersive channels
4. Signal processing and control applications








27 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


1. Define ? Equalization
2. What is recursive algorithm?
3. List the steps in the process of Adaptive equalization.
4. What is decision directed mode?
5. What are the uses of adaptive filters?
6. Define ? Rate of Convergence
7. What is the relation between analog and digital filter?
8. Name the algorithm used in Adaptive Equalization.
9. What is the impact of LMS algorithm?
10. What is the relation between analog and digital filter?
11. What is the factor for designing an equalizer?
12. What is non-recursive algorithm?
13. What is the purpose of receiver decision?
14. What is difference between recursive and non-recursive algorithm?
15. What is the maximum expected time delay spread in the channel?













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DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
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PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





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Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
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5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

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Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
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Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
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6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
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4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
19 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

20 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
21 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.5 DESIGN OF IIR FILTERS
Aim:
To write a program to design and obtain the response of the given IIR filter using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
A digital filter, the derivation of the digital filter transfer function requires three steps.
1. Map the desired digital filter specification into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
The various parameters in the figure are,
?p = Pass band frequency in radians
?s = Stop band frequency in radians
?c = 3-db cutoff frequency in radians
? = Parameter specifying allowable pass band
? = Parameter specifying allowable stop band
Design of IIR filters from analog filters:
There are several methods that can be used to design digital filters having an infinite duration unit
sample response. The techniques described are all based on converting an analog filter into digital filter. If
the conversion technique is to be effective, it should posses the following desirable properties.
The j ?-axes in the s-plane should map into the unit circle in the z-plane. Thus there will be a
direct relationship between the two frequency variables in the two domains.
The left-half plane of the s-plane should map into inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
Algorithm:
Butterworth low pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
22 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Butterworth high pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band stop filter
1. Get the pass band and stop band ripples
2. Get the pass band and stop band edge frequencies
3. Get the sampling frequencies
4. Calculate the order of the filter
5. Find the filter co-efficient
6. Draw the magnitude and phase response
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the IIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Implement IIR filter for given specifications
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2. Plot the response of the same.
Applications:
1. Implementation of filter circuits
2. Speech processing and video processing
3. Radio transmission
4. Filters in various industrial applications


1. Define ? IIR Filter
2. List the steps to design a digital filter using impulse invariant method.
3. What are the disadvantages of impulse invariant method?
4. Differentiate recursive & non-recursive systems.
5. List the parameters obtained from the Chebyshev filter specifications.
6. What is warping effect? Where we found this effect? How can we eliminate warping effect
7. What is the difference between type I & type II Chebyshev filters?
8. Define ? Cut-off frequency
9. Compare direct form I and direct form II realization of IIR filter.
10. List the digital techniques used to convert analog filter to digital filter.
11. List the properties of Butterworth low pass filter.
12. What is the role of delay element in filter design?
13. List the properties of Chebyshev low pass filter.
14. What are the properties of bilinear transformations?
15. Distinguish between the frequency responses of Chebyshev type I filter for N odd and N even.
16. Write the expression of the order of the Butterworth filter.
17. Define ? Prewarping
18. List the steps to design digital filter from analog filter.
19. How many number of additions, multiplications and memory locations are required to realize a
digital system in IIR filter?
20. Write the expression of the order of the Chebyshev filter.
Viva ?voce
24 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt.No.6 MULTIRATE FILTERS
Aim
To design multirate FIR filters using interpolation and decimation and to plot the impulse responses and
the magnitude responses for all designs.
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the multirate filters was designed using interpolation and decimation for the given input.
Outcome:
After finishing this experiment the students are able to design linear-phase FIR multirate filters and do
interpolation and decimation of the signals
Applications:
1. Sampling rate conversion systems
2. Communication systems
3. Design of phase shifters
4. Antenna arrays





25 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00



1. What is the importance of decimation for a given signal/sequence?
2. What is aliasing? What is the condition to avoid aliasing for sampling?
3. Define ? Decimation
4. Define ? Interpolation
5. What is multirate signal processing?
6. Name the process of converting a signal from a given rate to a different rate
7. State the formula for poly phase interpolation using z-transform.
8. List the applications of multirate signal processing.
9. List the methods used in sampling for converting a digital signal
10. List the advantages of multirate processing.
11. What is up sampling?
12. List the applications of multirate DSP.
13. What is the need for antialiasing filter prior to down sampling?
14. List the applications of adaptive filters
15. What are the sections in Quadrature mirror filter bank?
16. What is down sampling?
17. What is the need for anti-imaging filter after up sampling a signal?
18. State the formula for poly phase decimation using z-transform.
19. What is anti-aliasing filter?
20. Define ? Sampling theorem







Viva ?voce
26 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.7 EQUALIZATION

Aim
To write MATLAB program to design adaptive channel equalization using LMS algorithm
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the adaptive channel equalization was designed using LMS algorithm.
Outcome:
After finishing this experiment the students are able to design adaptive channel equalization using LMS
algorithm.
Applications:
1. Echo cancellation in telecommunication systems
2. Digital radio communication systems
3. Channel equalization for dispersive channels
4. Signal processing and control applications








27 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


1. Define ? Equalization
2. What is recursive algorithm?
3. List the steps in the process of Adaptive equalization.
4. What is decision directed mode?
5. What are the uses of adaptive filters?
6. Define ? Rate of Convergence
7. What is the relation between analog and digital filter?
8. Name the algorithm used in Adaptive Equalization.
9. What is the impact of LMS algorithm?
10. What is the relation between analog and digital filter?
11. What is the factor for designing an equalizer?
12. What is non-recursive algorithm?
13. What is the purpose of receiver decision?
14. What is difference between recursive and non-recursive algorithm?
15. What is the maximum expected time delay spread in the channel?













Viva ?voce
28 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt. No. 8 STUDY OF DSP PROCESSOR TMS320C50
Aim:
To study the architecture overview of DSP processor TMS320C50
Introduction:
It is needless to say that in order to utilize the full feature of the DSP chip TMS320C50, the DSP
engineer must have a complete knowledge of the DSP device. This chapter is an introduction to the
hardware aspects of the TMS320C50. The important units of TMS320C50 are discussed.
The DSP Chip TMS320C50:
The TMS320C50 is a 16-bit fixed point digital signal processor that combines the flexibility of a high
speed controller with the numerical capability of an array processor, thereby offering an inexpensive
alternative to multichip bit-slice processors. The highly paralleled architecture and efficient instruction
set, provide speed and flexibility capable of executing 10 MIPS (Million Instructions Per Second). The
TMS320C50 optimizes speed by implementing functions in hardware that other processors implement
through microcode or software. This hardware intensive approach provides the design engineer with
processing power previously unavailable on a single chip.
The TMS320C50 is the third generation digital signal processor in the TMS320 family. Its powerful
instruction set, inherent flexibility, high-speed number-crunching capabilities, and innovative
architecture have made this high-performance, cost-effective processor the ideal solution to many
telecommunications, computer, commercial, industrial, and military applications.
Key Features of TMS320C50:
The key features of the Digital Signal Processor TMS320C50 are:
? 35-/50-ns single-cycle fixed-point instruction execution time (28.6/20 MIPS)
? Upward source-code compatible with all C1X and C2x devices
? RAM-based memory operation (C50)
? 9K x 16-bit single-cycle on-chip program/data RAM (C50)
? 2K x 16-bit single-cycle on-chip boot ROM (C50)
? 1056 x 16-bit dual-access on-chip data RAM
? 224K x 16-bit maximum addressable external memory space (64K program, 64K data, 64K I/O)
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DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
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5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
12 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
18 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
19 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

20 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
21 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.5 DESIGN OF IIR FILTERS
Aim:
To write a program to design and obtain the response of the given IIR filter using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
A digital filter, the derivation of the digital filter transfer function requires three steps.
1. Map the desired digital filter specification into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
The various parameters in the figure are,
?p = Pass band frequency in radians
?s = Stop band frequency in radians
?c = 3-db cutoff frequency in radians
? = Parameter specifying allowable pass band
? = Parameter specifying allowable stop band
Design of IIR filters from analog filters:
There are several methods that can be used to design digital filters having an infinite duration unit
sample response. The techniques described are all based on converting an analog filter into digital filter. If
the conversion technique is to be effective, it should posses the following desirable properties.
The j ?-axes in the s-plane should map into the unit circle in the z-plane. Thus there will be a
direct relationship between the two frequency variables in the two domains.
The left-half plane of the s-plane should map into inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
Algorithm:
Butterworth low pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
22 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Butterworth high pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band stop filter
1. Get the pass band and stop band ripples
2. Get the pass band and stop band edge frequencies
3. Get the sampling frequencies
4. Calculate the order of the filter
5. Find the filter co-efficient
6. Draw the magnitude and phase response
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the IIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Implement IIR filter for given specifications
23 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

2. Plot the response of the same.
Applications:
1. Implementation of filter circuits
2. Speech processing and video processing
3. Radio transmission
4. Filters in various industrial applications


1. Define ? IIR Filter
2. List the steps to design a digital filter using impulse invariant method.
3. What are the disadvantages of impulse invariant method?
4. Differentiate recursive & non-recursive systems.
5. List the parameters obtained from the Chebyshev filter specifications.
6. What is warping effect? Where we found this effect? How can we eliminate warping effect
7. What is the difference between type I & type II Chebyshev filters?
8. Define ? Cut-off frequency
9. Compare direct form I and direct form II realization of IIR filter.
10. List the digital techniques used to convert analog filter to digital filter.
11. List the properties of Butterworth low pass filter.
12. What is the role of delay element in filter design?
13. List the properties of Chebyshev low pass filter.
14. What are the properties of bilinear transformations?
15. Distinguish between the frequency responses of Chebyshev type I filter for N odd and N even.
16. Write the expression of the order of the Butterworth filter.
17. Define ? Prewarping
18. List the steps to design digital filter from analog filter.
19. How many number of additions, multiplications and memory locations are required to realize a
digital system in IIR filter?
20. Write the expression of the order of the Chebyshev filter.
Viva ?voce
24 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt.No.6 MULTIRATE FILTERS
Aim
To design multirate FIR filters using interpolation and decimation and to plot the impulse responses and
the magnitude responses for all designs.
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the multirate filters was designed using interpolation and decimation for the given input.
Outcome:
After finishing this experiment the students are able to design linear-phase FIR multirate filters and do
interpolation and decimation of the signals
Applications:
1. Sampling rate conversion systems
2. Communication systems
3. Design of phase shifters
4. Antenna arrays





25 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00



1. What is the importance of decimation for a given signal/sequence?
2. What is aliasing? What is the condition to avoid aliasing for sampling?
3. Define ? Decimation
4. Define ? Interpolation
5. What is multirate signal processing?
6. Name the process of converting a signal from a given rate to a different rate
7. State the formula for poly phase interpolation using z-transform.
8. List the applications of multirate signal processing.
9. List the methods used in sampling for converting a digital signal
10. List the advantages of multirate processing.
11. What is up sampling?
12. List the applications of multirate DSP.
13. What is the need for antialiasing filter prior to down sampling?
14. List the applications of adaptive filters
15. What are the sections in Quadrature mirror filter bank?
16. What is down sampling?
17. What is the need for anti-imaging filter after up sampling a signal?
18. State the formula for poly phase decimation using z-transform.
19. What is anti-aliasing filter?
20. Define ? Sampling theorem







Viva ?voce
26 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.7 EQUALIZATION

Aim
To write MATLAB program to design adaptive channel equalization using LMS algorithm
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the adaptive channel equalization was designed using LMS algorithm.
Outcome:
After finishing this experiment the students are able to design adaptive channel equalization using LMS
algorithm.
Applications:
1. Echo cancellation in telecommunication systems
2. Digital radio communication systems
3. Channel equalization for dispersive channels
4. Signal processing and control applications








27 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


1. Define ? Equalization
2. What is recursive algorithm?
3. List the steps in the process of Adaptive equalization.
4. What is decision directed mode?
5. What are the uses of adaptive filters?
6. Define ? Rate of Convergence
7. What is the relation between analog and digital filter?
8. Name the algorithm used in Adaptive Equalization.
9. What is the impact of LMS algorithm?
10. What is the relation between analog and digital filter?
11. What is the factor for designing an equalizer?
12. What is non-recursive algorithm?
13. What is the purpose of receiver decision?
14. What is difference between recursive and non-recursive algorithm?
15. What is the maximum expected time delay spread in the channel?













Viva ?voce
28 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt. No. 8 STUDY OF DSP PROCESSOR TMS320C50
Aim:
To study the architecture overview of DSP processor TMS320C50
Introduction:
It is needless to say that in order to utilize the full feature of the DSP chip TMS320C50, the DSP
engineer must have a complete knowledge of the DSP device. This chapter is an introduction to the
hardware aspects of the TMS320C50. The important units of TMS320C50 are discussed.
The DSP Chip TMS320C50:
The TMS320C50 is a 16-bit fixed point digital signal processor that combines the flexibility of a high
speed controller with the numerical capability of an array processor, thereby offering an inexpensive
alternative to multichip bit-slice processors. The highly paralleled architecture and efficient instruction
set, provide speed and flexibility capable of executing 10 MIPS (Million Instructions Per Second). The
TMS320C50 optimizes speed by implementing functions in hardware that other processors implement
through microcode or software. This hardware intensive approach provides the design engineer with
processing power previously unavailable on a single chip.
The TMS320C50 is the third generation digital signal processor in the TMS320 family. Its powerful
instruction set, inherent flexibility, high-speed number-crunching capabilities, and innovative
architecture have made this high-performance, cost-effective processor the ideal solution to many
telecommunications, computer, commercial, industrial, and military applications.
Key Features of TMS320C50:
The key features of the Digital Signal Processor TMS320C50 are:
? 35-/50-ns single-cycle fixed-point instruction execution time (28.6/20 MIPS)
? Upward source-code compatible with all C1X and C2x devices
? RAM-based memory operation (C50)
? 9K x 16-bit single-cycle on-chip program/data RAM (C50)
? 2K x 16-bit single-cycle on-chip boot ROM (C50)
? 1056 x 16-bit dual-access on-chip data RAM
? 224K x 16-bit maximum addressable external memory space (64K program, 64K data, 64K I/O)
29 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

? 32-bit arithmetic logic unit (ALU), 32-bit accumulator (ACC), and 32-bit accumulator buffer
(ACCB)
? 16-bit parallel logic unit (PLU)
? 16 x 16-bit parallel multiplier with a 32-bit product capability.
? Single-cycle multiply/accumulate instructions
? Eight auxiliary registers with a dedicated auxiliary register arithmetic unit for indirect
addressing.
? Eleven context-switch registers (shadow registers) for storing strategic CPU controlled registers
during an interrupt service routine
? Eight-level hardware stack
? 0- to 16-bit left and right data barrel-shifters and a 64-bit incremental data shifter
? Two indirectly addressed circular buffers for circular addressing
? Single-instruction repeat and block repeat operations for program code
? Block memory move instructions for better program/data management
? Full-duplex synchronous serial port for direct communication between the C5x and another
serial device
? Time-division multiple-access (TDM) serial port
? Interval timer with period, control, and counter registers for software stop, start, and reset
? 64K parallel I/O ports, 16 of which are memory mapped
? Sixteen software programmable wait-state generators for program, data, and I/O
memory spaces.





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?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
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5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
12 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
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Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
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4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
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5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

20 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
21 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.5 DESIGN OF IIR FILTERS
Aim:
To write a program to design and obtain the response of the given IIR filter using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
A digital filter, the derivation of the digital filter transfer function requires three steps.
1. Map the desired digital filter specification into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
The various parameters in the figure are,
?p = Pass band frequency in radians
?s = Stop band frequency in radians
?c = 3-db cutoff frequency in radians
? = Parameter specifying allowable pass band
? = Parameter specifying allowable stop band
Design of IIR filters from analog filters:
There are several methods that can be used to design digital filters having an infinite duration unit
sample response. The techniques described are all based on converting an analog filter into digital filter. If
the conversion technique is to be effective, it should posses the following desirable properties.
The j ?-axes in the s-plane should map into the unit circle in the z-plane. Thus there will be a
direct relationship between the two frequency variables in the two domains.
The left-half plane of the s-plane should map into inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
Algorithm:
Butterworth low pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
22 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Butterworth high pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band stop filter
1. Get the pass band and stop band ripples
2. Get the pass band and stop band edge frequencies
3. Get the sampling frequencies
4. Calculate the order of the filter
5. Find the filter co-efficient
6. Draw the magnitude and phase response
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the IIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Implement IIR filter for given specifications
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2. Plot the response of the same.
Applications:
1. Implementation of filter circuits
2. Speech processing and video processing
3. Radio transmission
4. Filters in various industrial applications


1. Define ? IIR Filter
2. List the steps to design a digital filter using impulse invariant method.
3. What are the disadvantages of impulse invariant method?
4. Differentiate recursive & non-recursive systems.
5. List the parameters obtained from the Chebyshev filter specifications.
6. What is warping effect? Where we found this effect? How can we eliminate warping effect
7. What is the difference between type I & type II Chebyshev filters?
8. Define ? Cut-off frequency
9. Compare direct form I and direct form II realization of IIR filter.
10. List the digital techniques used to convert analog filter to digital filter.
11. List the properties of Butterworth low pass filter.
12. What is the role of delay element in filter design?
13. List the properties of Chebyshev low pass filter.
14. What are the properties of bilinear transformations?
15. Distinguish between the frequency responses of Chebyshev type I filter for N odd and N even.
16. Write the expression of the order of the Butterworth filter.
17. Define ? Prewarping
18. List the steps to design digital filter from analog filter.
19. How many number of additions, multiplications and memory locations are required to realize a
digital system in IIR filter?
20. Write the expression of the order of the Chebyshev filter.
Viva ?voce
24 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt.No.6 MULTIRATE FILTERS
Aim
To design multirate FIR filters using interpolation and decimation and to plot the impulse responses and
the magnitude responses for all designs.
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the multirate filters was designed using interpolation and decimation for the given input.
Outcome:
After finishing this experiment the students are able to design linear-phase FIR multirate filters and do
interpolation and decimation of the signals
Applications:
1. Sampling rate conversion systems
2. Communication systems
3. Design of phase shifters
4. Antenna arrays





25 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00



1. What is the importance of decimation for a given signal/sequence?
2. What is aliasing? What is the condition to avoid aliasing for sampling?
3. Define ? Decimation
4. Define ? Interpolation
5. What is multirate signal processing?
6. Name the process of converting a signal from a given rate to a different rate
7. State the formula for poly phase interpolation using z-transform.
8. List the applications of multirate signal processing.
9. List the methods used in sampling for converting a digital signal
10. List the advantages of multirate processing.
11. What is up sampling?
12. List the applications of multirate DSP.
13. What is the need for antialiasing filter prior to down sampling?
14. List the applications of adaptive filters
15. What are the sections in Quadrature mirror filter bank?
16. What is down sampling?
17. What is the need for anti-imaging filter after up sampling a signal?
18. State the formula for poly phase decimation using z-transform.
19. What is anti-aliasing filter?
20. Define ? Sampling theorem







Viva ?voce
26 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.7 EQUALIZATION

Aim
To write MATLAB program to design adaptive channel equalization using LMS algorithm
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the adaptive channel equalization was designed using LMS algorithm.
Outcome:
After finishing this experiment the students are able to design adaptive channel equalization using LMS
algorithm.
Applications:
1. Echo cancellation in telecommunication systems
2. Digital radio communication systems
3. Channel equalization for dispersive channels
4. Signal processing and control applications








27 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


1. Define ? Equalization
2. What is recursive algorithm?
3. List the steps in the process of Adaptive equalization.
4. What is decision directed mode?
5. What are the uses of adaptive filters?
6. Define ? Rate of Convergence
7. What is the relation between analog and digital filter?
8. Name the algorithm used in Adaptive Equalization.
9. What is the impact of LMS algorithm?
10. What is the relation between analog and digital filter?
11. What is the factor for designing an equalizer?
12. What is non-recursive algorithm?
13. What is the purpose of receiver decision?
14. What is difference between recursive and non-recursive algorithm?
15. What is the maximum expected time delay spread in the channel?













Viva ?voce
28 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt. No. 8 STUDY OF DSP PROCESSOR TMS320C50
Aim:
To study the architecture overview of DSP processor TMS320C50
Introduction:
It is needless to say that in order to utilize the full feature of the DSP chip TMS320C50, the DSP
engineer must have a complete knowledge of the DSP device. This chapter is an introduction to the
hardware aspects of the TMS320C50. The important units of TMS320C50 are discussed.
The DSP Chip TMS320C50:
The TMS320C50 is a 16-bit fixed point digital signal processor that combines the flexibility of a high
speed controller with the numerical capability of an array processor, thereby offering an inexpensive
alternative to multichip bit-slice processors. The highly paralleled architecture and efficient instruction
set, provide speed and flexibility capable of executing 10 MIPS (Million Instructions Per Second). The
TMS320C50 optimizes speed by implementing functions in hardware that other processors implement
through microcode or software. This hardware intensive approach provides the design engineer with
processing power previously unavailable on a single chip.
The TMS320C50 is the third generation digital signal processor in the TMS320 family. Its powerful
instruction set, inherent flexibility, high-speed number-crunching capabilities, and innovative
architecture have made this high-performance, cost-effective processor the ideal solution to many
telecommunications, computer, commercial, industrial, and military applications.
Key Features of TMS320C50:
The key features of the Digital Signal Processor TMS320C50 are:
? 35-/50-ns single-cycle fixed-point instruction execution time (28.6/20 MIPS)
? Upward source-code compatible with all C1X and C2x devices
? RAM-based memory operation (C50)
? 9K x 16-bit single-cycle on-chip program/data RAM (C50)
? 2K x 16-bit single-cycle on-chip boot ROM (C50)
? 1056 x 16-bit dual-access on-chip data RAM
? 224K x 16-bit maximum addressable external memory space (64K program, 64K data, 64K I/O)
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? 32-bit arithmetic logic unit (ALU), 32-bit accumulator (ACC), and 32-bit accumulator buffer
(ACCB)
? 16-bit parallel logic unit (PLU)
? 16 x 16-bit parallel multiplier with a 32-bit product capability.
? Single-cycle multiply/accumulate instructions
? Eight auxiliary registers with a dedicated auxiliary register arithmetic unit for indirect
addressing.
? Eleven context-switch registers (shadow registers) for storing strategic CPU controlled registers
during an interrupt service routine
? Eight-level hardware stack
? 0- to 16-bit left and right data barrel-shifters and a 64-bit incremental data shifter
? Two indirectly addressed circular buffers for circular addressing
? Single-instruction repeat and block repeat operations for program code
? Block memory move instructions for better program/data management
? Full-duplex synchronous serial port for direct communication between the C5x and another
serial device
? Time-division multiple-access (TDM) serial port
? Interval timer with period, control, and counter registers for software stop, start, and reset
? 64K parallel I/O ports, 16 of which are memory mapped
? Sixteen software programmable wait-state generators for program, data, and I/O
memory spaces.





30 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Theory:
A detailed architectural block diagram of TMS320C50 is illustrated in Figure. The TMS320C50
utilizes a modified Harvard architecture for speed and flexibility. In a strict Harvard architecture,
program and data memory are in two separate spaces, permitting a full overlap of instruction fetch and
execution. The TMS320 family's modification of the Harvard architecture allows transfer between
program and data spaces, thereby increasing the flexibility of the device. This modification permits
coefficients stored in program memory to be read into the data RAM, eliminating the need for a
separate coefficient ROM. It also makes available immediate instructions and subroutines based on
computed values.
32-bit Accumulator:
The TMS320C50 contains a 32-bit ALU and accumulator for support of double- precision, two's
complement arithmetic. The ALU is a general purpose arithmetic unit that operates on 16-bit words
taken from the data RAM or derived from immediate instructions. In addition to the usual arithmetic
instructions, the ALU can perform Boolean operations, providing the bit manipulation ability required of
a high-speed controller. The accumulator stores the output from the ALU and is often an input to the
ALU. Its word length is 32-bit. The accumulator is divided into a high order word (bits 31 through 16)
and a low-order word (bits 15 through 0). Instructions are provided for storing and loading the high and
lower order accumulator words to memory.
16 X 16-bit Parallel Multiplier:
The multiplier performs a 16 x 16-bit two's complement multiplication with a 32- bit result in a single
instruction cycle. The multiplier consists of three units: the T- Register, P-Register, and multiplier array.
The 16-bit T-Register temporarily stores the multiplicand and the P-Register stores the 32-bit product.
Multiplier values either come from the data memory or are derived immediately from the MPY (multiply
immediate) instruction word. The fast on-chip multiplier allows the device to perform fundamental
operations such as convolution, correlation, and filtering. Two multiply/accumulate instructions in the
instruction set fully utilize the computational bandwidth of the multiplier, allowing both operands to be
processed simultaneously.
Shifters:
A 16-bit scaling shifter is available at the accumulator input. This shifter produces a left shift of 0 to
16-bits on the input data to accumulator. TMS320C50 also contains a shifter at the accumulator output.
This shifter provide a left shift of 0 to 7, on the data from either the ACCH or ACCL register. In addition
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DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
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5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
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Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
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6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
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4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
19 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

20 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
21 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.5 DESIGN OF IIR FILTERS
Aim:
To write a program to design and obtain the response of the given IIR filter using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
A digital filter, the derivation of the digital filter transfer function requires three steps.
1. Map the desired digital filter specification into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
The various parameters in the figure are,
?p = Pass band frequency in radians
?s = Stop band frequency in radians
?c = 3-db cutoff frequency in radians
? = Parameter specifying allowable pass band
? = Parameter specifying allowable stop band
Design of IIR filters from analog filters:
There are several methods that can be used to design digital filters having an infinite duration unit
sample response. The techniques described are all based on converting an analog filter into digital filter. If
the conversion technique is to be effective, it should posses the following desirable properties.
The j ?-axes in the s-plane should map into the unit circle in the z-plane. Thus there will be a
direct relationship between the two frequency variables in the two domains.
The left-half plane of the s-plane should map into inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
Algorithm:
Butterworth low pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
22 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Butterworth high pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band stop filter
1. Get the pass band and stop band ripples
2. Get the pass band and stop band edge frequencies
3. Get the sampling frequencies
4. Calculate the order of the filter
5. Find the filter co-efficient
6. Draw the magnitude and phase response
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the IIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Implement IIR filter for given specifications
23 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

2. Plot the response of the same.
Applications:
1. Implementation of filter circuits
2. Speech processing and video processing
3. Radio transmission
4. Filters in various industrial applications


1. Define ? IIR Filter
2. List the steps to design a digital filter using impulse invariant method.
3. What are the disadvantages of impulse invariant method?
4. Differentiate recursive & non-recursive systems.
5. List the parameters obtained from the Chebyshev filter specifications.
6. What is warping effect? Where we found this effect? How can we eliminate warping effect
7. What is the difference between type I & type II Chebyshev filters?
8. Define ? Cut-off frequency
9. Compare direct form I and direct form II realization of IIR filter.
10. List the digital techniques used to convert analog filter to digital filter.
11. List the properties of Butterworth low pass filter.
12. What is the role of delay element in filter design?
13. List the properties of Chebyshev low pass filter.
14. What are the properties of bilinear transformations?
15. Distinguish between the frequency responses of Chebyshev type I filter for N odd and N even.
16. Write the expression of the order of the Butterworth filter.
17. Define ? Prewarping
18. List the steps to design digital filter from analog filter.
19. How many number of additions, multiplications and memory locations are required to realize a
digital system in IIR filter?
20. Write the expression of the order of the Chebyshev filter.
Viva ?voce
24 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt.No.6 MULTIRATE FILTERS
Aim
To design multirate FIR filters using interpolation and decimation and to plot the impulse responses and
the magnitude responses for all designs.
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the multirate filters was designed using interpolation and decimation for the given input.
Outcome:
After finishing this experiment the students are able to design linear-phase FIR multirate filters and do
interpolation and decimation of the signals
Applications:
1. Sampling rate conversion systems
2. Communication systems
3. Design of phase shifters
4. Antenna arrays





25 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00



1. What is the importance of decimation for a given signal/sequence?
2. What is aliasing? What is the condition to avoid aliasing for sampling?
3. Define ? Decimation
4. Define ? Interpolation
5. What is multirate signal processing?
6. Name the process of converting a signal from a given rate to a different rate
7. State the formula for poly phase interpolation using z-transform.
8. List the applications of multirate signal processing.
9. List the methods used in sampling for converting a digital signal
10. List the advantages of multirate processing.
11. What is up sampling?
12. List the applications of multirate DSP.
13. What is the need for antialiasing filter prior to down sampling?
14. List the applications of adaptive filters
15. What are the sections in Quadrature mirror filter bank?
16. What is down sampling?
17. What is the need for anti-imaging filter after up sampling a signal?
18. State the formula for poly phase decimation using z-transform.
19. What is anti-aliasing filter?
20. Define ? Sampling theorem







Viva ?voce
26 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.7 EQUALIZATION

Aim
To write MATLAB program to design adaptive channel equalization using LMS algorithm
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the adaptive channel equalization was designed using LMS algorithm.
Outcome:
After finishing this experiment the students are able to design adaptive channel equalization using LMS
algorithm.
Applications:
1. Echo cancellation in telecommunication systems
2. Digital radio communication systems
3. Channel equalization for dispersive channels
4. Signal processing and control applications








27 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


1. Define ? Equalization
2. What is recursive algorithm?
3. List the steps in the process of Adaptive equalization.
4. What is decision directed mode?
5. What are the uses of adaptive filters?
6. Define ? Rate of Convergence
7. What is the relation between analog and digital filter?
8. Name the algorithm used in Adaptive Equalization.
9. What is the impact of LMS algorithm?
10. What is the relation between analog and digital filter?
11. What is the factor for designing an equalizer?
12. What is non-recursive algorithm?
13. What is the purpose of receiver decision?
14. What is difference between recursive and non-recursive algorithm?
15. What is the maximum expected time delay spread in the channel?













Viva ?voce
28 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt. No. 8 STUDY OF DSP PROCESSOR TMS320C50
Aim:
To study the architecture overview of DSP processor TMS320C50
Introduction:
It is needless to say that in order to utilize the full feature of the DSP chip TMS320C50, the DSP
engineer must have a complete knowledge of the DSP device. This chapter is an introduction to the
hardware aspects of the TMS320C50. The important units of TMS320C50 are discussed.
The DSP Chip TMS320C50:
The TMS320C50 is a 16-bit fixed point digital signal processor that combines the flexibility of a high
speed controller with the numerical capability of an array processor, thereby offering an inexpensive
alternative to multichip bit-slice processors. The highly paralleled architecture and efficient instruction
set, provide speed and flexibility capable of executing 10 MIPS (Million Instructions Per Second). The
TMS320C50 optimizes speed by implementing functions in hardware that other processors implement
through microcode or software. This hardware intensive approach provides the design engineer with
processing power previously unavailable on a single chip.
The TMS320C50 is the third generation digital signal processor in the TMS320 family. Its powerful
instruction set, inherent flexibility, high-speed number-crunching capabilities, and innovative
architecture have made this high-performance, cost-effective processor the ideal solution to many
telecommunications, computer, commercial, industrial, and military applications.
Key Features of TMS320C50:
The key features of the Digital Signal Processor TMS320C50 are:
? 35-/50-ns single-cycle fixed-point instruction execution time (28.6/20 MIPS)
? Upward source-code compatible with all C1X and C2x devices
? RAM-based memory operation (C50)
? 9K x 16-bit single-cycle on-chip program/data RAM (C50)
? 2K x 16-bit single-cycle on-chip boot ROM (C50)
? 1056 x 16-bit dual-access on-chip data RAM
? 224K x 16-bit maximum addressable external memory space (64K program, 64K data, 64K I/O)
29 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

? 32-bit arithmetic logic unit (ALU), 32-bit accumulator (ACC), and 32-bit accumulator buffer
(ACCB)
? 16-bit parallel logic unit (PLU)
? 16 x 16-bit parallel multiplier with a 32-bit product capability.
? Single-cycle multiply/accumulate instructions
? Eight auxiliary registers with a dedicated auxiliary register arithmetic unit for indirect
addressing.
? Eleven context-switch registers (shadow registers) for storing strategic CPU controlled registers
during an interrupt service routine
? Eight-level hardware stack
? 0- to 16-bit left and right data barrel-shifters and a 64-bit incremental data shifter
? Two indirectly addressed circular buffers for circular addressing
? Single-instruction repeat and block repeat operations for program code
? Block memory move instructions for better program/data management
? Full-duplex synchronous serial port for direct communication between the C5x and another
serial device
? Time-division multiple-access (TDM) serial port
? Interval timer with period, control, and counter registers for software stop, start, and reset
? 64K parallel I/O ports, 16 of which are memory mapped
? Sixteen software programmable wait-state generators for program, data, and I/O
memory spaces.





30 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Theory:
A detailed architectural block diagram of TMS320C50 is illustrated in Figure. The TMS320C50
utilizes a modified Harvard architecture for speed and flexibility. In a strict Harvard architecture,
program and data memory are in two separate spaces, permitting a full overlap of instruction fetch and
execution. The TMS320 family's modification of the Harvard architecture allows transfer between
program and data spaces, thereby increasing the flexibility of the device. This modification permits
coefficients stored in program memory to be read into the data RAM, eliminating the need for a
separate coefficient ROM. It also makes available immediate instructions and subroutines based on
computed values.
32-bit Accumulator:
The TMS320C50 contains a 32-bit ALU and accumulator for support of double- precision, two's
complement arithmetic. The ALU is a general purpose arithmetic unit that operates on 16-bit words
taken from the data RAM or derived from immediate instructions. In addition to the usual arithmetic
instructions, the ALU can perform Boolean operations, providing the bit manipulation ability required of
a high-speed controller. The accumulator stores the output from the ALU and is often an input to the
ALU. Its word length is 32-bit. The accumulator is divided into a high order word (bits 31 through 16)
and a low-order word (bits 15 through 0). Instructions are provided for storing and loading the high and
lower order accumulator words to memory.
16 X 16-bit Parallel Multiplier:
The multiplier performs a 16 x 16-bit two's complement multiplication with a 32- bit result in a single
instruction cycle. The multiplier consists of three units: the T- Register, P-Register, and multiplier array.
The 16-bit T-Register temporarily stores the multiplicand and the P-Register stores the 32-bit product.
Multiplier values either come from the data memory or are derived immediately from the MPY (multiply
immediate) instruction word. The fast on-chip multiplier allows the device to perform fundamental
operations such as convolution, correlation, and filtering. Two multiply/accumulate instructions in the
instruction set fully utilize the computational bandwidth of the multiplier, allowing both operands to be
processed simultaneously.
Shifters:
A 16-bit scaling shifter is available at the accumulator input. This shifter produces a left shift of 0 to
16-bits on the input data to accumulator. TMS320C50 also contains a shifter at the accumulator output.
This shifter provide a left shift of 0 to 7, on the data from either the ACCH or ACCL register. In addition
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one shifter at the output of P- register, can shift the product by 1 or 4-bits left or 6-bits right, before
transferring the product to accumulator.
Interrupts and Subroutines:
The TMS320C50 has three external maskable user interrupts available for external devices that
interrupt the processor. The TMS320C50 contains a eight-level hardware stack for saving the contents
of the program counter during interrupts and subroutine calls. Instructions are available for saving the
device's complete context. PUSH and POP instructions permit a level of nesting restricted only by the
amount of available RAM.
Serial Port:
A full-duplex on-chip serial port provides direct communication with serial devices such as codec?s,
serial A/D converters and other serial systems. The interface signals are compatible with codec?s and
many others serial devices with a minimum of external hardware.
Input and Output:
The 16-bit parallel data bus can be utilized to perform I/O functions in two cycles. The I/O ports are
addressed by the four LSBs on the address lines, allowing 16 input and 16 output ports. In addition,
polling input for bit test and jump operations (BIO) and three interrupt pins (INT0 - INT2) have been
incorporated for multitasking.
Result:
Thus the architecture overview of DSP processor TMS320C50 was studied.
Outcome:
After finishing this experiment the students are able to know the architecture of DSP processor
TMS320C50.
Applications:
1. Audio signal processing
2. Speech recognition
3. Biomedicine
4. Seismology
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DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
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5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

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Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
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Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
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6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
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4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
19 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

20 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
21 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.5 DESIGN OF IIR FILTERS
Aim:
To write a program to design and obtain the response of the given IIR filter using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
A digital filter, the derivation of the digital filter transfer function requires three steps.
1. Map the desired digital filter specification into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
The various parameters in the figure are,
?p = Pass band frequency in radians
?s = Stop band frequency in radians
?c = 3-db cutoff frequency in radians
? = Parameter specifying allowable pass band
? = Parameter specifying allowable stop band
Design of IIR filters from analog filters:
There are several methods that can be used to design digital filters having an infinite duration unit
sample response. The techniques described are all based on converting an analog filter into digital filter. If
the conversion technique is to be effective, it should posses the following desirable properties.
The j ?-axes in the s-plane should map into the unit circle in the z-plane. Thus there will be a
direct relationship between the two frequency variables in the two domains.
The left-half plane of the s-plane should map into inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
Algorithm:
Butterworth low pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
22 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Butterworth high pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band stop filter
1. Get the pass band and stop band ripples
2. Get the pass band and stop band edge frequencies
3. Get the sampling frequencies
4. Calculate the order of the filter
5. Find the filter co-efficient
6. Draw the magnitude and phase response
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the IIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Implement IIR filter for given specifications
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2. Plot the response of the same.
Applications:
1. Implementation of filter circuits
2. Speech processing and video processing
3. Radio transmission
4. Filters in various industrial applications


1. Define ? IIR Filter
2. List the steps to design a digital filter using impulse invariant method.
3. What are the disadvantages of impulse invariant method?
4. Differentiate recursive & non-recursive systems.
5. List the parameters obtained from the Chebyshev filter specifications.
6. What is warping effect? Where we found this effect? How can we eliminate warping effect
7. What is the difference between type I & type II Chebyshev filters?
8. Define ? Cut-off frequency
9. Compare direct form I and direct form II realization of IIR filter.
10. List the digital techniques used to convert analog filter to digital filter.
11. List the properties of Butterworth low pass filter.
12. What is the role of delay element in filter design?
13. List the properties of Chebyshev low pass filter.
14. What are the properties of bilinear transformations?
15. Distinguish between the frequency responses of Chebyshev type I filter for N odd and N even.
16. Write the expression of the order of the Butterworth filter.
17. Define ? Prewarping
18. List the steps to design digital filter from analog filter.
19. How many number of additions, multiplications and memory locations are required to realize a
digital system in IIR filter?
20. Write the expression of the order of the Chebyshev filter.
Viva ?voce
24 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt.No.6 MULTIRATE FILTERS
Aim
To design multirate FIR filters using interpolation and decimation and to plot the impulse responses and
the magnitude responses for all designs.
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the multirate filters was designed using interpolation and decimation for the given input.
Outcome:
After finishing this experiment the students are able to design linear-phase FIR multirate filters and do
interpolation and decimation of the signals
Applications:
1. Sampling rate conversion systems
2. Communication systems
3. Design of phase shifters
4. Antenna arrays





25 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00



1. What is the importance of decimation for a given signal/sequence?
2. What is aliasing? What is the condition to avoid aliasing for sampling?
3. Define ? Decimation
4. Define ? Interpolation
5. What is multirate signal processing?
6. Name the process of converting a signal from a given rate to a different rate
7. State the formula for poly phase interpolation using z-transform.
8. List the applications of multirate signal processing.
9. List the methods used in sampling for converting a digital signal
10. List the advantages of multirate processing.
11. What is up sampling?
12. List the applications of multirate DSP.
13. What is the need for antialiasing filter prior to down sampling?
14. List the applications of adaptive filters
15. What are the sections in Quadrature mirror filter bank?
16. What is down sampling?
17. What is the need for anti-imaging filter after up sampling a signal?
18. State the formula for poly phase decimation using z-transform.
19. What is anti-aliasing filter?
20. Define ? Sampling theorem







Viva ?voce
26 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.7 EQUALIZATION

Aim
To write MATLAB program to design adaptive channel equalization using LMS algorithm
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the adaptive channel equalization was designed using LMS algorithm.
Outcome:
After finishing this experiment the students are able to design adaptive channel equalization using LMS
algorithm.
Applications:
1. Echo cancellation in telecommunication systems
2. Digital radio communication systems
3. Channel equalization for dispersive channels
4. Signal processing and control applications








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1. Define ? Equalization
2. What is recursive algorithm?
3. List the steps in the process of Adaptive equalization.
4. What is decision directed mode?
5. What are the uses of adaptive filters?
6. Define ? Rate of Convergence
7. What is the relation between analog and digital filter?
8. Name the algorithm used in Adaptive Equalization.
9. What is the impact of LMS algorithm?
10. What is the relation between analog and digital filter?
11. What is the factor for designing an equalizer?
12. What is non-recursive algorithm?
13. What is the purpose of receiver decision?
14. What is difference between recursive and non-recursive algorithm?
15. What is the maximum expected time delay spread in the channel?













Viva ?voce
28 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt. No. 8 STUDY OF DSP PROCESSOR TMS320C50
Aim:
To study the architecture overview of DSP processor TMS320C50
Introduction:
It is needless to say that in order to utilize the full feature of the DSP chip TMS320C50, the DSP
engineer must have a complete knowledge of the DSP device. This chapter is an introduction to the
hardware aspects of the TMS320C50. The important units of TMS320C50 are discussed.
The DSP Chip TMS320C50:
The TMS320C50 is a 16-bit fixed point digital signal processor that combines the flexibility of a high
speed controller with the numerical capability of an array processor, thereby offering an inexpensive
alternative to multichip bit-slice processors. The highly paralleled architecture and efficient instruction
set, provide speed and flexibility capable of executing 10 MIPS (Million Instructions Per Second). The
TMS320C50 optimizes speed by implementing functions in hardware that other processors implement
through microcode or software. This hardware intensive approach provides the design engineer with
processing power previously unavailable on a single chip.
The TMS320C50 is the third generation digital signal processor in the TMS320 family. Its powerful
instruction set, inherent flexibility, high-speed number-crunching capabilities, and innovative
architecture have made this high-performance, cost-effective processor the ideal solution to many
telecommunications, computer, commercial, industrial, and military applications.
Key Features of TMS320C50:
The key features of the Digital Signal Processor TMS320C50 are:
? 35-/50-ns single-cycle fixed-point instruction execution time (28.6/20 MIPS)
? Upward source-code compatible with all C1X and C2x devices
? RAM-based memory operation (C50)
? 9K x 16-bit single-cycle on-chip program/data RAM (C50)
? 2K x 16-bit single-cycle on-chip boot ROM (C50)
? 1056 x 16-bit dual-access on-chip data RAM
? 224K x 16-bit maximum addressable external memory space (64K program, 64K data, 64K I/O)
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? 32-bit arithmetic logic unit (ALU), 32-bit accumulator (ACC), and 32-bit accumulator buffer
(ACCB)
? 16-bit parallel logic unit (PLU)
? 16 x 16-bit parallel multiplier with a 32-bit product capability.
? Single-cycle multiply/accumulate instructions
? Eight auxiliary registers with a dedicated auxiliary register arithmetic unit for indirect
addressing.
? Eleven context-switch registers (shadow registers) for storing strategic CPU controlled registers
during an interrupt service routine
? Eight-level hardware stack
? 0- to 16-bit left and right data barrel-shifters and a 64-bit incremental data shifter
? Two indirectly addressed circular buffers for circular addressing
? Single-instruction repeat and block repeat operations for program code
? Block memory move instructions for better program/data management
? Full-duplex synchronous serial port for direct communication between the C5x and another
serial device
? Time-division multiple-access (TDM) serial port
? Interval timer with period, control, and counter registers for software stop, start, and reset
? 64K parallel I/O ports, 16 of which are memory mapped
? Sixteen software programmable wait-state generators for program, data, and I/O
memory spaces.





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Theory:
A detailed architectural block diagram of TMS320C50 is illustrated in Figure. The TMS320C50
utilizes a modified Harvard architecture for speed and flexibility. In a strict Harvard architecture,
program and data memory are in two separate spaces, permitting a full overlap of instruction fetch and
execution. The TMS320 family's modification of the Harvard architecture allows transfer between
program and data spaces, thereby increasing the flexibility of the device. This modification permits
coefficients stored in program memory to be read into the data RAM, eliminating the need for a
separate coefficient ROM. It also makes available immediate instructions and subroutines based on
computed values.
32-bit Accumulator:
The TMS320C50 contains a 32-bit ALU and accumulator for support of double- precision, two's
complement arithmetic. The ALU is a general purpose arithmetic unit that operates on 16-bit words
taken from the data RAM or derived from immediate instructions. In addition to the usual arithmetic
instructions, the ALU can perform Boolean operations, providing the bit manipulation ability required of
a high-speed controller. The accumulator stores the output from the ALU and is often an input to the
ALU. Its word length is 32-bit. The accumulator is divided into a high order word (bits 31 through 16)
and a low-order word (bits 15 through 0). Instructions are provided for storing and loading the high and
lower order accumulator words to memory.
16 X 16-bit Parallel Multiplier:
The multiplier performs a 16 x 16-bit two's complement multiplication with a 32- bit result in a single
instruction cycle. The multiplier consists of three units: the T- Register, P-Register, and multiplier array.
The 16-bit T-Register temporarily stores the multiplicand and the P-Register stores the 32-bit product.
Multiplier values either come from the data memory or are derived immediately from the MPY (multiply
immediate) instruction word. The fast on-chip multiplier allows the device to perform fundamental
operations such as convolution, correlation, and filtering. Two multiply/accumulate instructions in the
instruction set fully utilize the computational bandwidth of the multiplier, allowing both operands to be
processed simultaneously.
Shifters:
A 16-bit scaling shifter is available at the accumulator input. This shifter produces a left shift of 0 to
16-bits on the input data to accumulator. TMS320C50 also contains a shifter at the accumulator output.
This shifter provide a left shift of 0 to 7, on the data from either the ACCH or ACCL register. In addition
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one shifter at the output of P- register, can shift the product by 1 or 4-bits left or 6-bits right, before
transferring the product to accumulator.
Interrupts and Subroutines:
The TMS320C50 has three external maskable user interrupts available for external devices that
interrupt the processor. The TMS320C50 contains a eight-level hardware stack for saving the contents
of the program counter during interrupts and subroutine calls. Instructions are available for saving the
device's complete context. PUSH and POP instructions permit a level of nesting restricted only by the
amount of available RAM.
Serial Port:
A full-duplex on-chip serial port provides direct communication with serial devices such as codec?s,
serial A/D converters and other serial systems. The interface signals are compatible with codec?s and
many others serial devices with a minimum of external hardware.
Input and Output:
The 16-bit parallel data bus can be utilized to perform I/O functions in two cycles. The I/O ports are
addressed by the four LSBs on the address lines, allowing 16 input and 16 output ports. In addition,
polling input for bit test and jump operations (BIO) and three interrupt pins (INT0 - INT2) have been
incorporated for multitasking.
Result:
Thus the architecture overview of DSP processor TMS320C50 was studied.
Outcome:
After finishing this experiment the students are able to know the architecture of DSP processor
TMS320C50.
Applications:
1. Audio signal processing
2. Speech recognition
3. Biomedicine
4. Seismology
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5. RADAR
6. Digital communication



1. List the types of DSP processors.
2. List the features of TMS 320C6X processor.
3. Differentiate Von Neumann architecture and Harvard architecture.
4. Name the architecture used in DSP processor
5. How many instructions can be executed per cycle in TMS320C6X DSP processor?
6. Name the register that stores the result of multiplication in DSP processor.
7. How many instruction cycles are used for 16 x 16 multiplication in TMS320C6X DSP processor?
8. What are the applications for the TMS320 DSP?s?
9. Name the software tool required to compile and run the DSP assembly program.
10. What is the functional unit adapted for transferring the data from the register in TMS 320 C6x
processor?
11. What are the difference buses of TMS320C5X and their functions?
12. What are the different stages in pipelining?
13. What is pipeline depth of TMS320C50 and TMS320C54x?
14. What is the function of parallel logic unit?
15. How many functional units are in TMS 320C6X DSP processor?
16. What are the advantages and disadvantages of VLIW architecture?
17. What is pipeline depth?
18. List the on chip peripheral in C5x.
19. What is the logical instruction of c5x?
20. What are the factors that influence selection of DSP?

Viva ?voce
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?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
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5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

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Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
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Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

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Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
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6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
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4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
19 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

20 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
21 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.5 DESIGN OF IIR FILTERS
Aim:
To write a program to design and obtain the response of the given IIR filter using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
A digital filter, the derivation of the digital filter transfer function requires three steps.
1. Map the desired digital filter specification into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
The various parameters in the figure are,
?p = Pass band frequency in radians
?s = Stop band frequency in radians
?c = 3-db cutoff frequency in radians
? = Parameter specifying allowable pass band
? = Parameter specifying allowable stop band
Design of IIR filters from analog filters:
There are several methods that can be used to design digital filters having an infinite duration unit
sample response. The techniques described are all based on converting an analog filter into digital filter. If
the conversion technique is to be effective, it should posses the following desirable properties.
The j ?-axes in the s-plane should map into the unit circle in the z-plane. Thus there will be a
direct relationship between the two frequency variables in the two domains.
The left-half plane of the s-plane should map into inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
Algorithm:
Butterworth low pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
22 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Butterworth high pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band stop filter
1. Get the pass band and stop band ripples
2. Get the pass band and stop band edge frequencies
3. Get the sampling frequencies
4. Calculate the order of the filter
5. Find the filter co-efficient
6. Draw the magnitude and phase response
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the IIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Implement IIR filter for given specifications
23 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

2. Plot the response of the same.
Applications:
1. Implementation of filter circuits
2. Speech processing and video processing
3. Radio transmission
4. Filters in various industrial applications


1. Define ? IIR Filter
2. List the steps to design a digital filter using impulse invariant method.
3. What are the disadvantages of impulse invariant method?
4. Differentiate recursive & non-recursive systems.
5. List the parameters obtained from the Chebyshev filter specifications.
6. What is warping effect? Where we found this effect? How can we eliminate warping effect
7. What is the difference between type I & type II Chebyshev filters?
8. Define ? Cut-off frequency
9. Compare direct form I and direct form II realization of IIR filter.
10. List the digital techniques used to convert analog filter to digital filter.
11. List the properties of Butterworth low pass filter.
12. What is the role of delay element in filter design?
13. List the properties of Chebyshev low pass filter.
14. What are the properties of bilinear transformations?
15. Distinguish between the frequency responses of Chebyshev type I filter for N odd and N even.
16. Write the expression of the order of the Butterworth filter.
17. Define ? Prewarping
18. List the steps to design digital filter from analog filter.
19. How many number of additions, multiplications and memory locations are required to realize a
digital system in IIR filter?
20. Write the expression of the order of the Chebyshev filter.
Viva ?voce
24 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt.No.6 MULTIRATE FILTERS
Aim
To design multirate FIR filters using interpolation and decimation and to plot the impulse responses and
the magnitude responses for all designs.
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the multirate filters was designed using interpolation and decimation for the given input.
Outcome:
After finishing this experiment the students are able to design linear-phase FIR multirate filters and do
interpolation and decimation of the signals
Applications:
1. Sampling rate conversion systems
2. Communication systems
3. Design of phase shifters
4. Antenna arrays





25 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00



1. What is the importance of decimation for a given signal/sequence?
2. What is aliasing? What is the condition to avoid aliasing for sampling?
3. Define ? Decimation
4. Define ? Interpolation
5. What is multirate signal processing?
6. Name the process of converting a signal from a given rate to a different rate
7. State the formula for poly phase interpolation using z-transform.
8. List the applications of multirate signal processing.
9. List the methods used in sampling for converting a digital signal
10. List the advantages of multirate processing.
11. What is up sampling?
12. List the applications of multirate DSP.
13. What is the need for antialiasing filter prior to down sampling?
14. List the applications of adaptive filters
15. What are the sections in Quadrature mirror filter bank?
16. What is down sampling?
17. What is the need for anti-imaging filter after up sampling a signal?
18. State the formula for poly phase decimation using z-transform.
19. What is anti-aliasing filter?
20. Define ? Sampling theorem







Viva ?voce
26 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.7 EQUALIZATION

Aim
To write MATLAB program to design adaptive channel equalization using LMS algorithm
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the adaptive channel equalization was designed using LMS algorithm.
Outcome:
After finishing this experiment the students are able to design adaptive channel equalization using LMS
algorithm.
Applications:
1. Echo cancellation in telecommunication systems
2. Digital radio communication systems
3. Channel equalization for dispersive channels
4. Signal processing and control applications








27 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


1. Define ? Equalization
2. What is recursive algorithm?
3. List the steps in the process of Adaptive equalization.
4. What is decision directed mode?
5. What are the uses of adaptive filters?
6. Define ? Rate of Convergence
7. What is the relation between analog and digital filter?
8. Name the algorithm used in Adaptive Equalization.
9. What is the impact of LMS algorithm?
10. What is the relation between analog and digital filter?
11. What is the factor for designing an equalizer?
12. What is non-recursive algorithm?
13. What is the purpose of receiver decision?
14. What is difference between recursive and non-recursive algorithm?
15. What is the maximum expected time delay spread in the channel?













Viva ?voce
28 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt. No. 8 STUDY OF DSP PROCESSOR TMS320C50
Aim:
To study the architecture overview of DSP processor TMS320C50
Introduction:
It is needless to say that in order to utilize the full feature of the DSP chip TMS320C50, the DSP
engineer must have a complete knowledge of the DSP device. This chapter is an introduction to the
hardware aspects of the TMS320C50. The important units of TMS320C50 are discussed.
The DSP Chip TMS320C50:
The TMS320C50 is a 16-bit fixed point digital signal processor that combines the flexibility of a high
speed controller with the numerical capability of an array processor, thereby offering an inexpensive
alternative to multichip bit-slice processors. The highly paralleled architecture and efficient instruction
set, provide speed and flexibility capable of executing 10 MIPS (Million Instructions Per Second). The
TMS320C50 optimizes speed by implementing functions in hardware that other processors implement
through microcode or software. This hardware intensive approach provides the design engineer with
processing power previously unavailable on a single chip.
The TMS320C50 is the third generation digital signal processor in the TMS320 family. Its powerful
instruction set, inherent flexibility, high-speed number-crunching capabilities, and innovative
architecture have made this high-performance, cost-effective processor the ideal solution to many
telecommunications, computer, commercial, industrial, and military applications.
Key Features of TMS320C50:
The key features of the Digital Signal Processor TMS320C50 are:
? 35-/50-ns single-cycle fixed-point instruction execution time (28.6/20 MIPS)
? Upward source-code compatible with all C1X and C2x devices
? RAM-based memory operation (C50)
? 9K x 16-bit single-cycle on-chip program/data RAM (C50)
? 2K x 16-bit single-cycle on-chip boot ROM (C50)
? 1056 x 16-bit dual-access on-chip data RAM
? 224K x 16-bit maximum addressable external memory space (64K program, 64K data, 64K I/O)
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? 32-bit arithmetic logic unit (ALU), 32-bit accumulator (ACC), and 32-bit accumulator buffer
(ACCB)
? 16-bit parallel logic unit (PLU)
? 16 x 16-bit parallel multiplier with a 32-bit product capability.
? Single-cycle multiply/accumulate instructions
? Eight auxiliary registers with a dedicated auxiliary register arithmetic unit for indirect
addressing.
? Eleven context-switch registers (shadow registers) for storing strategic CPU controlled registers
during an interrupt service routine
? Eight-level hardware stack
? 0- to 16-bit left and right data barrel-shifters and a 64-bit incremental data shifter
? Two indirectly addressed circular buffers for circular addressing
? Single-instruction repeat and block repeat operations for program code
? Block memory move instructions for better program/data management
? Full-duplex synchronous serial port for direct communication between the C5x and another
serial device
? Time-division multiple-access (TDM) serial port
? Interval timer with period, control, and counter registers for software stop, start, and reset
? 64K parallel I/O ports, 16 of which are memory mapped
? Sixteen software programmable wait-state generators for program, data, and I/O
memory spaces.





30 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Theory:
A detailed architectural block diagram of TMS320C50 is illustrated in Figure. The TMS320C50
utilizes a modified Harvard architecture for speed and flexibility. In a strict Harvard architecture,
program and data memory are in two separate spaces, permitting a full overlap of instruction fetch and
execution. The TMS320 family's modification of the Harvard architecture allows transfer between
program and data spaces, thereby increasing the flexibility of the device. This modification permits
coefficients stored in program memory to be read into the data RAM, eliminating the need for a
separate coefficient ROM. It also makes available immediate instructions and subroutines based on
computed values.
32-bit Accumulator:
The TMS320C50 contains a 32-bit ALU and accumulator for support of double- precision, two's
complement arithmetic. The ALU is a general purpose arithmetic unit that operates on 16-bit words
taken from the data RAM or derived from immediate instructions. In addition to the usual arithmetic
instructions, the ALU can perform Boolean operations, providing the bit manipulation ability required of
a high-speed controller. The accumulator stores the output from the ALU and is often an input to the
ALU. Its word length is 32-bit. The accumulator is divided into a high order word (bits 31 through 16)
and a low-order word (bits 15 through 0). Instructions are provided for storing and loading the high and
lower order accumulator words to memory.
16 X 16-bit Parallel Multiplier:
The multiplier performs a 16 x 16-bit two's complement multiplication with a 32- bit result in a single
instruction cycle. The multiplier consists of three units: the T- Register, P-Register, and multiplier array.
The 16-bit T-Register temporarily stores the multiplicand and the P-Register stores the 32-bit product.
Multiplier values either come from the data memory or are derived immediately from the MPY (multiply
immediate) instruction word. The fast on-chip multiplier allows the device to perform fundamental
operations such as convolution, correlation, and filtering. Two multiply/accumulate instructions in the
instruction set fully utilize the computational bandwidth of the multiplier, allowing both operands to be
processed simultaneously.
Shifters:
A 16-bit scaling shifter is available at the accumulator input. This shifter produces a left shift of 0 to
16-bits on the input data to accumulator. TMS320C50 also contains a shifter at the accumulator output.
This shifter provide a left shift of 0 to 7, on the data from either the ACCH or ACCL register. In addition
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one shifter at the output of P- register, can shift the product by 1 or 4-bits left or 6-bits right, before
transferring the product to accumulator.
Interrupts and Subroutines:
The TMS320C50 has three external maskable user interrupts available for external devices that
interrupt the processor. The TMS320C50 contains a eight-level hardware stack for saving the contents
of the program counter during interrupts and subroutine calls. Instructions are available for saving the
device's complete context. PUSH and POP instructions permit a level of nesting restricted only by the
amount of available RAM.
Serial Port:
A full-duplex on-chip serial port provides direct communication with serial devices such as codec?s,
serial A/D converters and other serial systems. The interface signals are compatible with codec?s and
many others serial devices with a minimum of external hardware.
Input and Output:
The 16-bit parallel data bus can be utilized to perform I/O functions in two cycles. The I/O ports are
addressed by the four LSBs on the address lines, allowing 16 input and 16 output ports. In addition,
polling input for bit test and jump operations (BIO) and three interrupt pins (INT0 - INT2) have been
incorporated for multitasking.
Result:
Thus the architecture overview of DSP processor TMS320C50 was studied.
Outcome:
After finishing this experiment the students are able to know the architecture of DSP processor
TMS320C50.
Applications:
1. Audio signal processing
2. Speech recognition
3. Biomedicine
4. Seismology
32 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. RADAR
6. Digital communication



1. List the types of DSP processors.
2. List the features of TMS 320C6X processor.
3. Differentiate Von Neumann architecture and Harvard architecture.
4. Name the architecture used in DSP processor
5. How many instructions can be executed per cycle in TMS320C6X DSP processor?
6. Name the register that stores the result of multiplication in DSP processor.
7. How many instruction cycles are used for 16 x 16 multiplication in TMS320C6X DSP processor?
8. What are the applications for the TMS320 DSP?s?
9. Name the software tool required to compile and run the DSP assembly program.
10. What is the functional unit adapted for transferring the data from the register in TMS 320 C6x
processor?
11. What are the difference buses of TMS320C5X and their functions?
12. What are the different stages in pipelining?
13. What is pipeline depth of TMS320C50 and TMS320C54x?
14. What is the function of parallel logic unit?
15. How many functional units are in TMS 320C6X DSP processor?
16. What are the advantages and disadvantages of VLIW architecture?
17. What is pipeline depth?
18. List the on chip peripheral in C5x.
19. What is the logical instruction of c5x?
20. What are the factors that influence selection of DSP?

Viva ?voce
33 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.9 MAC OPERATIONS USING VARIOUS
ADDRESSING MODES
Aim:
To study the various addressing modes of TMS320C50 processor using the following programs:
1. Addition of two data
2. Subtraction of two data
3. Multiplication of two data
Apparatus Required:
1. Digital signal processor - TMS320C50 Trainer kit
2. PC with C50 debugger
Algorithm:
Addition of two data:
1. Start
2. Initialize the memory mapped registers.
3. Load the data pointer with address 100H.
4. Load the first data in accumulator.
5. Add the second data with accumulator.
6. Store the result in memory.
7. Check the result in various addressing modes.
8. Stop
Subtraction of two data:
1. Start
2. Initialize the memory mapped registers.
3. Load the data pointer with address 100H.
4. Load the first data in accumulator.
5. Subtract the second data from the accumulator.
6. Store the result in memory.
7. Check the result in various addressing modes.
8. Stop

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To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
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EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
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5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

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Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
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Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
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6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
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Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
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Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
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4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
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5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

20 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
21 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.5 DESIGN OF IIR FILTERS
Aim:
To write a program to design and obtain the response of the given IIR filter using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
A digital filter, the derivation of the digital filter transfer function requires three steps.
1. Map the desired digital filter specification into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
The various parameters in the figure are,
?p = Pass band frequency in radians
?s = Stop band frequency in radians
?c = 3-db cutoff frequency in radians
? = Parameter specifying allowable pass band
? = Parameter specifying allowable stop band
Design of IIR filters from analog filters:
There are several methods that can be used to design digital filters having an infinite duration unit
sample response. The techniques described are all based on converting an analog filter into digital filter. If
the conversion technique is to be effective, it should posses the following desirable properties.
The j ?-axes in the s-plane should map into the unit circle in the z-plane. Thus there will be a
direct relationship between the two frequency variables in the two domains.
The left-half plane of the s-plane should map into inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
Algorithm:
Butterworth low pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
22 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Butterworth high pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band stop filter
1. Get the pass band and stop band ripples
2. Get the pass band and stop band edge frequencies
3. Get the sampling frequencies
4. Calculate the order of the filter
5. Find the filter co-efficient
6. Draw the magnitude and phase response
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the IIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Implement IIR filter for given specifications
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2. Plot the response of the same.
Applications:
1. Implementation of filter circuits
2. Speech processing and video processing
3. Radio transmission
4. Filters in various industrial applications


1. Define ? IIR Filter
2. List the steps to design a digital filter using impulse invariant method.
3. What are the disadvantages of impulse invariant method?
4. Differentiate recursive & non-recursive systems.
5. List the parameters obtained from the Chebyshev filter specifications.
6. What is warping effect? Where we found this effect? How can we eliminate warping effect
7. What is the difference between type I & type II Chebyshev filters?
8. Define ? Cut-off frequency
9. Compare direct form I and direct form II realization of IIR filter.
10. List the digital techniques used to convert analog filter to digital filter.
11. List the properties of Butterworth low pass filter.
12. What is the role of delay element in filter design?
13. List the properties of Chebyshev low pass filter.
14. What are the properties of bilinear transformations?
15. Distinguish between the frequency responses of Chebyshev type I filter for N odd and N even.
16. Write the expression of the order of the Butterworth filter.
17. Define ? Prewarping
18. List the steps to design digital filter from analog filter.
19. How many number of additions, multiplications and memory locations are required to realize a
digital system in IIR filter?
20. Write the expression of the order of the Chebyshev filter.
Viva ?voce
24 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt.No.6 MULTIRATE FILTERS
Aim
To design multirate FIR filters using interpolation and decimation and to plot the impulse responses and
the magnitude responses for all designs.
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the multirate filters was designed using interpolation and decimation for the given input.
Outcome:
After finishing this experiment the students are able to design linear-phase FIR multirate filters and do
interpolation and decimation of the signals
Applications:
1. Sampling rate conversion systems
2. Communication systems
3. Design of phase shifters
4. Antenna arrays





25 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00



1. What is the importance of decimation for a given signal/sequence?
2. What is aliasing? What is the condition to avoid aliasing for sampling?
3. Define ? Decimation
4. Define ? Interpolation
5. What is multirate signal processing?
6. Name the process of converting a signal from a given rate to a different rate
7. State the formula for poly phase interpolation using z-transform.
8. List the applications of multirate signal processing.
9. List the methods used in sampling for converting a digital signal
10. List the advantages of multirate processing.
11. What is up sampling?
12. List the applications of multirate DSP.
13. What is the need for antialiasing filter prior to down sampling?
14. List the applications of adaptive filters
15. What are the sections in Quadrature mirror filter bank?
16. What is down sampling?
17. What is the need for anti-imaging filter after up sampling a signal?
18. State the formula for poly phase decimation using z-transform.
19. What is anti-aliasing filter?
20. Define ? Sampling theorem







Viva ?voce
26 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.7 EQUALIZATION

Aim
To write MATLAB program to design adaptive channel equalization using LMS algorithm
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the adaptive channel equalization was designed using LMS algorithm.
Outcome:
After finishing this experiment the students are able to design adaptive channel equalization using LMS
algorithm.
Applications:
1. Echo cancellation in telecommunication systems
2. Digital radio communication systems
3. Channel equalization for dispersive channels
4. Signal processing and control applications








27 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


1. Define ? Equalization
2. What is recursive algorithm?
3. List the steps in the process of Adaptive equalization.
4. What is decision directed mode?
5. What are the uses of adaptive filters?
6. Define ? Rate of Convergence
7. What is the relation between analog and digital filter?
8. Name the algorithm used in Adaptive Equalization.
9. What is the impact of LMS algorithm?
10. What is the relation between analog and digital filter?
11. What is the factor for designing an equalizer?
12. What is non-recursive algorithm?
13. What is the purpose of receiver decision?
14. What is difference between recursive and non-recursive algorithm?
15. What is the maximum expected time delay spread in the channel?













Viva ?voce
28 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt. No. 8 STUDY OF DSP PROCESSOR TMS320C50
Aim:
To study the architecture overview of DSP processor TMS320C50
Introduction:
It is needless to say that in order to utilize the full feature of the DSP chip TMS320C50, the DSP
engineer must have a complete knowledge of the DSP device. This chapter is an introduction to the
hardware aspects of the TMS320C50. The important units of TMS320C50 are discussed.
The DSP Chip TMS320C50:
The TMS320C50 is a 16-bit fixed point digital signal processor that combines the flexibility of a high
speed controller with the numerical capability of an array processor, thereby offering an inexpensive
alternative to multichip bit-slice processors. The highly paralleled architecture and efficient instruction
set, provide speed and flexibility capable of executing 10 MIPS (Million Instructions Per Second). The
TMS320C50 optimizes speed by implementing functions in hardware that other processors implement
through microcode or software. This hardware intensive approach provides the design engineer with
processing power previously unavailable on a single chip.
The TMS320C50 is the third generation digital signal processor in the TMS320 family. Its powerful
instruction set, inherent flexibility, high-speed number-crunching capabilities, and innovative
architecture have made this high-performance, cost-effective processor the ideal solution to many
telecommunications, computer, commercial, industrial, and military applications.
Key Features of TMS320C50:
The key features of the Digital Signal Processor TMS320C50 are:
? 35-/50-ns single-cycle fixed-point instruction execution time (28.6/20 MIPS)
? Upward source-code compatible with all C1X and C2x devices
? RAM-based memory operation (C50)
? 9K x 16-bit single-cycle on-chip program/data RAM (C50)
? 2K x 16-bit single-cycle on-chip boot ROM (C50)
? 1056 x 16-bit dual-access on-chip data RAM
? 224K x 16-bit maximum addressable external memory space (64K program, 64K data, 64K I/O)
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? 32-bit arithmetic logic unit (ALU), 32-bit accumulator (ACC), and 32-bit accumulator buffer
(ACCB)
? 16-bit parallel logic unit (PLU)
? 16 x 16-bit parallel multiplier with a 32-bit product capability.
? Single-cycle multiply/accumulate instructions
? Eight auxiliary registers with a dedicated auxiliary register arithmetic unit for indirect
addressing.
? Eleven context-switch registers (shadow registers) for storing strategic CPU controlled registers
during an interrupt service routine
? Eight-level hardware stack
? 0- to 16-bit left and right data barrel-shifters and a 64-bit incremental data shifter
? Two indirectly addressed circular buffers for circular addressing
? Single-instruction repeat and block repeat operations for program code
? Block memory move instructions for better program/data management
? Full-duplex synchronous serial port for direct communication between the C5x and another
serial device
? Time-division multiple-access (TDM) serial port
? Interval timer with period, control, and counter registers for software stop, start, and reset
? 64K parallel I/O ports, 16 of which are memory mapped
? Sixteen software programmable wait-state generators for program, data, and I/O
memory spaces.





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Theory:
A detailed architectural block diagram of TMS320C50 is illustrated in Figure. The TMS320C50
utilizes a modified Harvard architecture for speed and flexibility. In a strict Harvard architecture,
program and data memory are in two separate spaces, permitting a full overlap of instruction fetch and
execution. The TMS320 family's modification of the Harvard architecture allows transfer between
program and data spaces, thereby increasing the flexibility of the device. This modification permits
coefficients stored in program memory to be read into the data RAM, eliminating the need for a
separate coefficient ROM. It also makes available immediate instructions and subroutines based on
computed values.
32-bit Accumulator:
The TMS320C50 contains a 32-bit ALU and accumulator for support of double- precision, two's
complement arithmetic. The ALU is a general purpose arithmetic unit that operates on 16-bit words
taken from the data RAM or derived from immediate instructions. In addition to the usual arithmetic
instructions, the ALU can perform Boolean operations, providing the bit manipulation ability required of
a high-speed controller. The accumulator stores the output from the ALU and is often an input to the
ALU. Its word length is 32-bit. The accumulator is divided into a high order word (bits 31 through 16)
and a low-order word (bits 15 through 0). Instructions are provided for storing and loading the high and
lower order accumulator words to memory.
16 X 16-bit Parallel Multiplier:
The multiplier performs a 16 x 16-bit two's complement multiplication with a 32- bit result in a single
instruction cycle. The multiplier consists of three units: the T- Register, P-Register, and multiplier array.
The 16-bit T-Register temporarily stores the multiplicand and the P-Register stores the 32-bit product.
Multiplier values either come from the data memory or are derived immediately from the MPY (multiply
immediate) instruction word. The fast on-chip multiplier allows the device to perform fundamental
operations such as convolution, correlation, and filtering. Two multiply/accumulate instructions in the
instruction set fully utilize the computational bandwidth of the multiplier, allowing both operands to be
processed simultaneously.
Shifters:
A 16-bit scaling shifter is available at the accumulator input. This shifter produces a left shift of 0 to
16-bits on the input data to accumulator. TMS320C50 also contains a shifter at the accumulator output.
This shifter provide a left shift of 0 to 7, on the data from either the ACCH or ACCL register. In addition
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one shifter at the output of P- register, can shift the product by 1 or 4-bits left or 6-bits right, before
transferring the product to accumulator.
Interrupts and Subroutines:
The TMS320C50 has three external maskable user interrupts available for external devices that
interrupt the processor. The TMS320C50 contains a eight-level hardware stack for saving the contents
of the program counter during interrupts and subroutine calls. Instructions are available for saving the
device's complete context. PUSH and POP instructions permit a level of nesting restricted only by the
amount of available RAM.
Serial Port:
A full-duplex on-chip serial port provides direct communication with serial devices such as codec?s,
serial A/D converters and other serial systems. The interface signals are compatible with codec?s and
many others serial devices with a minimum of external hardware.
Input and Output:
The 16-bit parallel data bus can be utilized to perform I/O functions in two cycles. The I/O ports are
addressed by the four LSBs on the address lines, allowing 16 input and 16 output ports. In addition,
polling input for bit test and jump operations (BIO) and three interrupt pins (INT0 - INT2) have been
incorporated for multitasking.
Result:
Thus the architecture overview of DSP processor TMS320C50 was studied.
Outcome:
After finishing this experiment the students are able to know the architecture of DSP processor
TMS320C50.
Applications:
1. Audio signal processing
2. Speech recognition
3. Biomedicine
4. Seismology
32 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. RADAR
6. Digital communication



1. List the types of DSP processors.
2. List the features of TMS 320C6X processor.
3. Differentiate Von Neumann architecture and Harvard architecture.
4. Name the architecture used in DSP processor
5. How many instructions can be executed per cycle in TMS320C6X DSP processor?
6. Name the register that stores the result of multiplication in DSP processor.
7. How many instruction cycles are used for 16 x 16 multiplication in TMS320C6X DSP processor?
8. What are the applications for the TMS320 DSP?s?
9. Name the software tool required to compile and run the DSP assembly program.
10. What is the functional unit adapted for transferring the data from the register in TMS 320 C6x
processor?
11. What are the difference buses of TMS320C5X and their functions?
12. What are the different stages in pipelining?
13. What is pipeline depth of TMS320C50 and TMS320C54x?
14. What is the function of parallel logic unit?
15. How many functional units are in TMS 320C6X DSP processor?
16. What are the advantages and disadvantages of VLIW architecture?
17. What is pipeline depth?
18. List the on chip peripheral in C5x.
19. What is the logical instruction of c5x?
20. What are the factors that influence selection of DSP?

Viva ?voce
33 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.9 MAC OPERATIONS USING VARIOUS
ADDRESSING MODES
Aim:
To study the various addressing modes of TMS320C50 processor using the following programs:
1. Addition of two data
2. Subtraction of two data
3. Multiplication of two data
Apparatus Required:
1. Digital signal processor - TMS320C50 Trainer kit
2. PC with C50 debugger
Algorithm:
Addition of two data:
1. Start
2. Initialize the memory mapped registers.
3. Load the data pointer with address 100H.
4. Load the first data in accumulator.
5. Add the second data with accumulator.
6. Store the result in memory.
7. Check the result in various addressing modes.
8. Stop
Subtraction of two data:
1. Start
2. Initialize the memory mapped registers.
3. Load the data pointer with address 100H.
4. Load the first data in accumulator.
5. Subtract the second data from the accumulator.
6. Store the result in memory.
7. Check the result in various addressing modes.
8. Stop

34 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Multiplication of two data:
1. Start.
2. Initialize the memory mapped registers.
3. Load the data pointer with address 100H.
4. Load the first data in accumulator.
5. Multiply the second data with accumulator.
6. Store the result in memory.
7. Check the result in various addressing modes.
8. Stop
Procedure:
1. Enter the program in work space.
2. Save and compile the program to generate ASCII codes.
3. Download the ASCII codes to micro 50 trainer kit.
4. Execute the program.
5. Observe the output in memory.
Result:
Thus the various addressing modes of TMS320C50 are studied and the outputs are verified.
Outcome:
After finishing this experiment the students are able to perform arithmetic operations using various
addressing modes in TMS320C50 Processor.
Applications:
1. Implementation of digital filters for speech and music processing
2. Correlation of signals in RADAR
3. Computing the output of LTI systems for various input sequences
4. Study of system behaviour

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?



DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
7 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
9 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
12 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
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4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
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5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

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Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
21 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.5 DESIGN OF IIR FILTERS
Aim:
To write a program to design and obtain the response of the given IIR filter using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
A digital filter, the derivation of the digital filter transfer function requires three steps.
1. Map the desired digital filter specification into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
The various parameters in the figure are,
?p = Pass band frequency in radians
?s = Stop band frequency in radians
?c = 3-db cutoff frequency in radians
? = Parameter specifying allowable pass band
? = Parameter specifying allowable stop band
Design of IIR filters from analog filters:
There are several methods that can be used to design digital filters having an infinite duration unit
sample response. The techniques described are all based on converting an analog filter into digital filter. If
the conversion technique is to be effective, it should posses the following desirable properties.
The j ?-axes in the s-plane should map into the unit circle in the z-plane. Thus there will be a
direct relationship between the two frequency variables in the two domains.
The left-half plane of the s-plane should map into inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
Algorithm:
Butterworth low pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
22 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Butterworth high pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band stop filter
1. Get the pass band and stop band ripples
2. Get the pass band and stop band edge frequencies
3. Get the sampling frequencies
4. Calculate the order of the filter
5. Find the filter co-efficient
6. Draw the magnitude and phase response
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the IIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Implement IIR filter for given specifications
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2. Plot the response of the same.
Applications:
1. Implementation of filter circuits
2. Speech processing and video processing
3. Radio transmission
4. Filters in various industrial applications


1. Define ? IIR Filter
2. List the steps to design a digital filter using impulse invariant method.
3. What are the disadvantages of impulse invariant method?
4. Differentiate recursive & non-recursive systems.
5. List the parameters obtained from the Chebyshev filter specifications.
6. What is warping effect? Where we found this effect? How can we eliminate warping effect
7. What is the difference between type I & type II Chebyshev filters?
8. Define ? Cut-off frequency
9. Compare direct form I and direct form II realization of IIR filter.
10. List the digital techniques used to convert analog filter to digital filter.
11. List the properties of Butterworth low pass filter.
12. What is the role of delay element in filter design?
13. List the properties of Chebyshev low pass filter.
14. What are the properties of bilinear transformations?
15. Distinguish between the frequency responses of Chebyshev type I filter for N odd and N even.
16. Write the expression of the order of the Butterworth filter.
17. Define ? Prewarping
18. List the steps to design digital filter from analog filter.
19. How many number of additions, multiplications and memory locations are required to realize a
digital system in IIR filter?
20. Write the expression of the order of the Chebyshev filter.
Viva ?voce
24 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt.No.6 MULTIRATE FILTERS
Aim
To design multirate FIR filters using interpolation and decimation and to plot the impulse responses and
the magnitude responses for all designs.
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the multirate filters was designed using interpolation and decimation for the given input.
Outcome:
After finishing this experiment the students are able to design linear-phase FIR multirate filters and do
interpolation and decimation of the signals
Applications:
1. Sampling rate conversion systems
2. Communication systems
3. Design of phase shifters
4. Antenna arrays





25 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00



1. What is the importance of decimation for a given signal/sequence?
2. What is aliasing? What is the condition to avoid aliasing for sampling?
3. Define ? Decimation
4. Define ? Interpolation
5. What is multirate signal processing?
6. Name the process of converting a signal from a given rate to a different rate
7. State the formula for poly phase interpolation using z-transform.
8. List the applications of multirate signal processing.
9. List the methods used in sampling for converting a digital signal
10. List the advantages of multirate processing.
11. What is up sampling?
12. List the applications of multirate DSP.
13. What is the need for antialiasing filter prior to down sampling?
14. List the applications of adaptive filters
15. What are the sections in Quadrature mirror filter bank?
16. What is down sampling?
17. What is the need for anti-imaging filter after up sampling a signal?
18. State the formula for poly phase decimation using z-transform.
19. What is anti-aliasing filter?
20. Define ? Sampling theorem







Viva ?voce
26 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.7 EQUALIZATION

Aim
To write MATLAB program to design adaptive channel equalization using LMS algorithm
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the adaptive channel equalization was designed using LMS algorithm.
Outcome:
After finishing this experiment the students are able to design adaptive channel equalization using LMS
algorithm.
Applications:
1. Echo cancellation in telecommunication systems
2. Digital radio communication systems
3. Channel equalization for dispersive channels
4. Signal processing and control applications








27 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


1. Define ? Equalization
2. What is recursive algorithm?
3. List the steps in the process of Adaptive equalization.
4. What is decision directed mode?
5. What are the uses of adaptive filters?
6. Define ? Rate of Convergence
7. What is the relation between analog and digital filter?
8. Name the algorithm used in Adaptive Equalization.
9. What is the impact of LMS algorithm?
10. What is the relation between analog and digital filter?
11. What is the factor for designing an equalizer?
12. What is non-recursive algorithm?
13. What is the purpose of receiver decision?
14. What is difference between recursive and non-recursive algorithm?
15. What is the maximum expected time delay spread in the channel?













Viva ?voce
28 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt. No. 8 STUDY OF DSP PROCESSOR TMS320C50
Aim:
To study the architecture overview of DSP processor TMS320C50
Introduction:
It is needless to say that in order to utilize the full feature of the DSP chip TMS320C50, the DSP
engineer must have a complete knowledge of the DSP device. This chapter is an introduction to the
hardware aspects of the TMS320C50. The important units of TMS320C50 are discussed.
The DSP Chip TMS320C50:
The TMS320C50 is a 16-bit fixed point digital signal processor that combines the flexibility of a high
speed controller with the numerical capability of an array processor, thereby offering an inexpensive
alternative to multichip bit-slice processors. The highly paralleled architecture and efficient instruction
set, provide speed and flexibility capable of executing 10 MIPS (Million Instructions Per Second). The
TMS320C50 optimizes speed by implementing functions in hardware that other processors implement
through microcode or software. This hardware intensive approach provides the design engineer with
processing power previously unavailable on a single chip.
The TMS320C50 is the third generation digital signal processor in the TMS320 family. Its powerful
instruction set, inherent flexibility, high-speed number-crunching capabilities, and innovative
architecture have made this high-performance, cost-effective processor the ideal solution to many
telecommunications, computer, commercial, industrial, and military applications.
Key Features of TMS320C50:
The key features of the Digital Signal Processor TMS320C50 are:
? 35-/50-ns single-cycle fixed-point instruction execution time (28.6/20 MIPS)
? Upward source-code compatible with all C1X and C2x devices
? RAM-based memory operation (C50)
? 9K x 16-bit single-cycle on-chip program/data RAM (C50)
? 2K x 16-bit single-cycle on-chip boot ROM (C50)
? 1056 x 16-bit dual-access on-chip data RAM
? 224K x 16-bit maximum addressable external memory space (64K program, 64K data, 64K I/O)
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? 32-bit arithmetic logic unit (ALU), 32-bit accumulator (ACC), and 32-bit accumulator buffer
(ACCB)
? 16-bit parallel logic unit (PLU)
? 16 x 16-bit parallel multiplier with a 32-bit product capability.
? Single-cycle multiply/accumulate instructions
? Eight auxiliary registers with a dedicated auxiliary register arithmetic unit for indirect
addressing.
? Eleven context-switch registers (shadow registers) for storing strategic CPU controlled registers
during an interrupt service routine
? Eight-level hardware stack
? 0- to 16-bit left and right data barrel-shifters and a 64-bit incremental data shifter
? Two indirectly addressed circular buffers for circular addressing
? Single-instruction repeat and block repeat operations for program code
? Block memory move instructions for better program/data management
? Full-duplex synchronous serial port for direct communication between the C5x and another
serial device
? Time-division multiple-access (TDM) serial port
? Interval timer with period, control, and counter registers for software stop, start, and reset
? 64K parallel I/O ports, 16 of which are memory mapped
? Sixteen software programmable wait-state generators for program, data, and I/O
memory spaces.





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Theory:
A detailed architectural block diagram of TMS320C50 is illustrated in Figure. The TMS320C50
utilizes a modified Harvard architecture for speed and flexibility. In a strict Harvard architecture,
program and data memory are in two separate spaces, permitting a full overlap of instruction fetch and
execution. The TMS320 family's modification of the Harvard architecture allows transfer between
program and data spaces, thereby increasing the flexibility of the device. This modification permits
coefficients stored in program memory to be read into the data RAM, eliminating the need for a
separate coefficient ROM. It also makes available immediate instructions and subroutines based on
computed values.
32-bit Accumulator:
The TMS320C50 contains a 32-bit ALU and accumulator for support of double- precision, two's
complement arithmetic. The ALU is a general purpose arithmetic unit that operates on 16-bit words
taken from the data RAM or derived from immediate instructions. In addition to the usual arithmetic
instructions, the ALU can perform Boolean operations, providing the bit manipulation ability required of
a high-speed controller. The accumulator stores the output from the ALU and is often an input to the
ALU. Its word length is 32-bit. The accumulator is divided into a high order word (bits 31 through 16)
and a low-order word (bits 15 through 0). Instructions are provided for storing and loading the high and
lower order accumulator words to memory.
16 X 16-bit Parallel Multiplier:
The multiplier performs a 16 x 16-bit two's complement multiplication with a 32- bit result in a single
instruction cycle. The multiplier consists of three units: the T- Register, P-Register, and multiplier array.
The 16-bit T-Register temporarily stores the multiplicand and the P-Register stores the 32-bit product.
Multiplier values either come from the data memory or are derived immediately from the MPY (multiply
immediate) instruction word. The fast on-chip multiplier allows the device to perform fundamental
operations such as convolution, correlation, and filtering. Two multiply/accumulate instructions in the
instruction set fully utilize the computational bandwidth of the multiplier, allowing both operands to be
processed simultaneously.
Shifters:
A 16-bit scaling shifter is available at the accumulator input. This shifter produces a left shift of 0 to
16-bits on the input data to accumulator. TMS320C50 also contains a shifter at the accumulator output.
This shifter provide a left shift of 0 to 7, on the data from either the ACCH or ACCL register. In addition
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one shifter at the output of P- register, can shift the product by 1 or 4-bits left or 6-bits right, before
transferring the product to accumulator.
Interrupts and Subroutines:
The TMS320C50 has three external maskable user interrupts available for external devices that
interrupt the processor. The TMS320C50 contains a eight-level hardware stack for saving the contents
of the program counter during interrupts and subroutine calls. Instructions are available for saving the
device's complete context. PUSH and POP instructions permit a level of nesting restricted only by the
amount of available RAM.
Serial Port:
A full-duplex on-chip serial port provides direct communication with serial devices such as codec?s,
serial A/D converters and other serial systems. The interface signals are compatible with codec?s and
many others serial devices with a minimum of external hardware.
Input and Output:
The 16-bit parallel data bus can be utilized to perform I/O functions in two cycles. The I/O ports are
addressed by the four LSBs on the address lines, allowing 16 input and 16 output ports. In addition,
polling input for bit test and jump operations (BIO) and three interrupt pins (INT0 - INT2) have been
incorporated for multitasking.
Result:
Thus the architecture overview of DSP processor TMS320C50 was studied.
Outcome:
After finishing this experiment the students are able to know the architecture of DSP processor
TMS320C50.
Applications:
1. Audio signal processing
2. Speech recognition
3. Biomedicine
4. Seismology
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5. RADAR
6. Digital communication



1. List the types of DSP processors.
2. List the features of TMS 320C6X processor.
3. Differentiate Von Neumann architecture and Harvard architecture.
4. Name the architecture used in DSP processor
5. How many instructions can be executed per cycle in TMS320C6X DSP processor?
6. Name the register that stores the result of multiplication in DSP processor.
7. How many instruction cycles are used for 16 x 16 multiplication in TMS320C6X DSP processor?
8. What are the applications for the TMS320 DSP?s?
9. Name the software tool required to compile and run the DSP assembly program.
10. What is the functional unit adapted for transferring the data from the register in TMS 320 C6x
processor?
11. What are the difference buses of TMS320C5X and their functions?
12. What are the different stages in pipelining?
13. What is pipeline depth of TMS320C50 and TMS320C54x?
14. What is the function of parallel logic unit?
15. How many functional units are in TMS 320C6X DSP processor?
16. What are the advantages and disadvantages of VLIW architecture?
17. What is pipeline depth?
18. List the on chip peripheral in C5x.
19. What is the logical instruction of c5x?
20. What are the factors that influence selection of DSP?

Viva ?voce
33 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.9 MAC OPERATIONS USING VARIOUS
ADDRESSING MODES
Aim:
To study the various addressing modes of TMS320C50 processor using the following programs:
1. Addition of two data
2. Subtraction of two data
3. Multiplication of two data
Apparatus Required:
1. Digital signal processor - TMS320C50 Trainer kit
2. PC with C50 debugger
Algorithm:
Addition of two data:
1. Start
2. Initialize the memory mapped registers.
3. Load the data pointer with address 100H.
4. Load the first data in accumulator.
5. Add the second data with accumulator.
6. Store the result in memory.
7. Check the result in various addressing modes.
8. Stop
Subtraction of two data:
1. Start
2. Initialize the memory mapped registers.
3. Load the data pointer with address 100H.
4. Load the first data in accumulator.
5. Subtract the second data from the accumulator.
6. Store the result in memory.
7. Check the result in various addressing modes.
8. Stop

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Multiplication of two data:
1. Start.
2. Initialize the memory mapped registers.
3. Load the data pointer with address 100H.
4. Load the first data in accumulator.
5. Multiply the second data with accumulator.
6. Store the result in memory.
7. Check the result in various addressing modes.
8. Stop
Procedure:
1. Enter the program in work space.
2. Save and compile the program to generate ASCII codes.
3. Download the ASCII codes to micro 50 trainer kit.
4. Execute the program.
5. Observe the output in memory.
Result:
Thus the various addressing modes of TMS320C50 are studied and the outputs are verified.
Outcome:
After finishing this experiment the students are able to perform arithmetic operations using various
addressing modes in TMS320C50 Processor.
Applications:
1. Implementation of digital filters for speech and music processing
2. Correlation of signals in RADAR
3. Computing the output of LTI systems for various input sequences
4. Study of system behaviour

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1. List the addressing modes supported by C5X processors.
2. List the on-chip peripherals in C5X.
3. What are the features of TMS320C50 processor?
4. What are the different stages in pipelining?
5. What is parallel processing?
6. What is pipelining?
7. What are the classification digital signal processors?
8. What is MAC?
9. What is barrel shifter? List its advantages in DSP processors
10. What is digital signal processor?
11. What is the expansion of MAC in TMS320C50?
12. What is the purpose of shadow registers in TMS320C50 processor?
13. Differentiate between RISC and CISC architectures.
14. What is the general purpose of I/O pins?
15. List the different quantization methods








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DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY

V SEMESTER - R 2013







Name : _______________________________________
Register No. : _______________________________________
Section : _______________________________________
LABORATORY MANUAL
1 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00




is committed to provide highly disciplined, conscientious and
enterprising professionals conforming to global standards through value based quality education and
training.

? To provide competent technical manpower capable of meeting requirements of the industry
? To contribute to the promotion of Academic Excellence in pursuit of Technical Education at different
levels
? To train the students to sell his brawn and brain to the highest bidder but to never put a price tag on
heart and soul
DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

To impart professional education integrated with human values to the younger generation, so as to
shape them as proficient and dedicated engineers, capable of providing comprehensive solutions to the
challenges in deploying technology for the service of humanity

? To educate the students with the state-of-art technologies to meet the growing challenges of the
electronics industry
? To carry out research through continuous interaction with research institutes and industry, on
advances in communication systems
? To provide the students with strong ground rules to facilitate them for systematic learning,
innovation and ethical practices
VISION
VISION
MISSION
MISSION
2 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME EDUCATIONAL OBJECTIVES (PEOs)
1. Fundamentals
To provide students with a solid foundation in Mathematics, Science and fundamentals of
engineering, enabling them to apply, to find solutions for engineering problems and use this knowledge
to acquire higher education
2. Core Competence
To train the students in Electronics and Communication technologies so that they apply their
knowledge and training to compare, and to analyze various engineering industrial problems to find
solutions
3. Breadth
To provide relevant training and experience to bridge the gap between theory and practice which
enables them to find solutions for the real time problems in industry, and to design products
4. Professionalism
To inculcate professional and effective communication skills, leadership qualities and team spirit in
the students to make them multi-faceted personalities and develop their ability to relate engineering
issues to broader social context
5. Lifelong Learning/Ethics
To demonstrate and practice ethical and professional responsibilities in the industry and society in
the large, through commitment and lifelong learning needed for successful professional career







3 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

PROGRAMME OUTCOME (POs)
a) To demonstrate and apply knowledge of Mathematics, Science and engineering fundamentals in
Electronics and Communication Engineering field
b) To design a component, a system or a process to meet the specific needs within the realistic
constraints such as economics, environment, ethics, health, safety and manufacturability
c) To demonstrate the competency to use software tools for computation, simulation and testing of
electronics and communication engineering circuits
d) To identify, formulate and solve electronic and communication engineering problems
e) To demonstrate an ability to visualize and work on laboratory and multidisciplinary tasks
f) To function as a member or a leader in multidisciplinary activities
g) To communicate in verbal and written form with fellow engineers and society at large
h) To understand the impact of Electronics and Communication Engineering in the society and
demonstrate awareness of contemporary issues and commitment to give solutions exhibiting
social responsibility
i) To demonstrate professional & ethical responsibilities
j) To exhibit confidence in self-education and ability for lifelong learning
k) To participate and succeed in competitive exams







4 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
SYLLABUS


? To implement linear and circular convolution
? To implement FIR and IIR filters
? To study the architecture of DSP processor
? To demonstrate finite word length effect
LIST OF EXPERIMENTS:
1. Generation of sequences (functional & random) & correlation
2. Linear and circular convolutions
3. Spectrum analysis using DFT
4. FIR filter design
5. IIR filter design
6. Multirate filters
7. Equalization
DSP PROCESSOR BASED IMPLEMENTATION
1. Study of architecture of digital signal processor
2. MAC operation using various addressing modes
3. Linear convolution
4. Circular convolution
5. FFT Implementation
6. Waveform generation
7. IIR and FIR implementation
8. Finite word length effect


1. Carry out simulation of DSP systems
2. Demonstrate their abilities towards DSP processor based implementation of DSP system
3. Analyze finite word length effect on DSP systems
4. Demonstrate the applications of FFT to DSP


COURSE OBJECTIVES
COURSE OUTCOME
5 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

EC6511 ? DIGITAL SIGNAL PROCESSING LABORATORY
CONTENTS
Sl.No. Name of the Experiment Page No.
CYCLE 1 ? EXPERIMENTS
1 Generation of signals 6
2 Linear and Circular Convolution of two sequences 11
3 Spectrum analysis using DFT 15
4 Design of FIR filters 17
5 Design of IIR filters 21
6 Multirate filters 24
7 Equalization 26
CYCLE 2 ? EXPERIMENTS
8 Study of DSP processor TMS320C50 28
9 MAC operations using various addressing modes 33
10 Implementation of Linear Convolution 36
11 Implementation of Circular Convolution 38
12 Calculation of FFT 40
13 Waveform Generation 43
14 Implementation of IIR and FIR filters 48
15 Analysis of Finite Word Length Effects in fixed point DSP systems 50
ADDITIONAL EXPERIMENTS BEYOND THE SYLLABUS
16 Decimation by Polyphase Decomposition 53
17 Sampling of signals 55





6 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.1 GENERATION OF SIGNALS

Aim:
To write a program to generate the following sequences and to verify using MATLAB
1. Sine wave
2. Cosine wave
3. Ramp signal
4. Exponential signal
5. Impulse signal
6. Step signal
7. Saw tooth signal
8. Rectangular signal
9. Random sequence
10. Auto correlation
11. Cross correlation
Apparatus required:
PC with MATLAB software
Theory:
A signal is defined as any physical quantity that varies with time, space or any other independent
variable or variables. Mathematically, we describe a signal as a function of one or more independent
variables.
A system may be defined as a physical device that performs an operation on a signal. It can also be
regarded as a process that transforms one signal into another. The input to a system and the output from a
system are signals.

Algorithm:
Sine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?.
3. Determine the values of the sine function.
4. Plot the graph for obtained values in continuous and discrete forms.
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5. Label the x and y axes.
6. Stop
Cosine Wave Generation:
1. Start
2. Get the frequency and range of time ?t?
3. Determine the values of the cosine function.
4. Plot the graph for obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Ramp Signal Generation:
1. Start
2. Get the amplitude and range of time ?t?.
3. Assign the ramp signal output equal to input time.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Exponential Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the exponential function.
4. Plot the graph for the obtained values in continuous and discrete forms.
5. Label the x and y axes.
6. Stop
Unit Impulse Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the unit impulse signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop

8 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Step Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the unit step signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Saw tooth Signal Generation:
1. Start
2. Get the ranges of time ?t?.
3. Determine the values of the saw tooth signal.
4. Plot the graph for the obtained values in discrete form.
5. Label the x and y axes.
6. Stop
Rectangular Signal Generation:
1. Start
2. Get the amplitude and ranges of time ?t?.
3. Determine the values of the rectangular signal.
4. Multiply the unit step signal with given amplitude.
5. Plot the graph for the resultant values in continuous and discrete forms.
6. Label the x and y axes.
7. Stop
Random Sequence:
1. Start
2. Get the length of input sequence.
3. Generate random signals of the specified length
4. Plot the graph for the resultant values in discrete form.
5. Label the x and y axes.
6. Stop
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Auto Correlation:
1. Start
2. Get the length of input sequence.
3. Generate a rectangular signal of given length.
4. Calculate auto correlation of the rectangular signal.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Cross Correlation:
1. Start
2. Get the length of input sequence.
3. Generate sinusoidal and cosine waveforms of given length.
4. Calculate cross correlation of the two signals.
5. Plot the graph for the resultant values in discrete forms.
6. Label the x and y axes.
7. Stop
Procedure:
1. Click on the MATLAB Icon on the desktop.
2. Click on the ?FILE? Menu on menu bar.
3. Click on NEW M-File from the file Menu.
4. Save the file in directory.
5. Click on DEBUG from Menu bar and Click Run.
6. Open the command window\ Figure window for the output.
Result:
Thus the signal waveforms were analyzed for the given input.
Outcome:
After finishing this experiment the students are able to
1. Generate elementary signals/ waveforms.
2. Perform arithmetic operations on signals.

10 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Aerospace and avionics
2. Defence applications
3. Wireless communication
4. Radar
5. GPS


1. Expand MATLAB
2. What is MATLAB? Where is MATLAB applicable?
3. What is MATLAB API?
4. What is M-file and MEX files in MATLAB?
5. List the operators of MATLAB.
6. Name the graphic system used in MATLAB.
7. What is disp() and input()?
8. What is the syntax to find the rank of the matrix?
9. What is the function of clear all and close all?
10. Define ? Continuous time signal
11. Define ? Discrete time signal
12. Define ? Impulse function
13. Define ? Unit step function
14. What is a ramp function?
15. Define ? Delta function
16. What are the basic operations that can be performed over the signals?
17. List the types of discrete time signals.
18. List the types of continuous time signals.
19. What is an exponential function?
20. What are the types of program files that MATLAB allows to write?


Viva-voce
11 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt. No. 2 LINEAR AND CIRCULAR CONVOLUTION OF
TWO SEQUENCES
Aim:
To write a program to perform the following convolutions and to verify using MATLAB
1. Linear convolution
2. Circular convolution
3. Linear convolution using circular convolution
4. Linear convolution using DFT
5. Circular convolution using DFT
Apparatus Required:
PC with MATLAB software
Algorithm:
Linear Convolution:
1. Start
2. Get the two input sequences.
3. Obtain the linear convolution of two sequences.
4. Calculate the lengths of input sequences.
5. Assign the length of resultant sequence(N) equal to length of first sequence (L) + length of
second sequence (M) ? 1 (N = L + M ? 1).
6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Circular Convolution:
1. Start
2. Get the two input sequences.
3. Check the maximum length (N) of two sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate circular convolution of equal length sequences.
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6. Plot the input sequences and output sequence in discrete form corresponding to their lengths.
7. Label the x and y axes.
8. Display the resultant sequence values.
9. Stop
Linear Convolution using Circular Convolution:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L)+
length of second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Obtain the circular convolution of two sequences.
7. Plot the input sequences and output sequence in discrete form corresponding to
their lengths.
8. Label the x and y axes.
9. Display the resultant sequence values.
10. Stop
Linear Convolution using DFT method:
1. Start.
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. Assign the length of resultant sequence(N) equal to length of first sequence(L) +b length of
second sequence(M) ? 1 (N = L + M - 1)
5. By zero padding, make both the two sequences with equal length (N).
6. Calculate the N-point FFT of two sequences.
7. Multiply the two FFTs.
8. Calculate the N-point IFFT of the sequence.
9. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
10. Label the x and y axes.
11. Display the resultant sequence values.
12. Stop.
13 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Circular Convolution using DFT method:
1. Start
2. Get the two input sequences.
3. Calculate the lengths of input sequences.
4. By zero padding, make both the two sequences with equal length (N).
5. Calculate the N-point FFT of two sequences.
6. Multiply the two FFTs.
7. Calculate the N-point IFFT of the sequence.
8. Plot the input sequences and output sequences FFT and IFFT in discrete form corresponding to
their lengths.
9. Label the x and y axes.
10. Display the resultant sequence values.
11. Stop.
Procedure:
1. Get two signals x(n) and h(n) in matrix form.
2. The convolute signal is denoted as y(n).
3. y(n) is given by the formula
y(n) = ?x(k)h(n-k)
4. n = length(X) returns the size of the longest dimension of X. If X is a vector, this is
the same as its length.
5. w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same
operation as multiplying the polynomials whose coefficients are the elements of u
and v. [subplot divides the current figure into rectangular panes that are numbered
row wise]
6. Stop the program
Result:
Thus the linear and circular convolution of the given two sequences was performed using MATLAB.
Outcome:
After finishing this experiment the students are able to implement convolution of given sequences
using linear, circular, linear using circular, linear convolution using DFT method, circular convolution
using DFT method.
14 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Finding the response of LTI systems
2. Spectral analysis
3. Study of system behavior
4. Frequency response of systems



1. What is the difference between linear convolution and circular convolution?
2. List the methods of linear convolution.
3. List the steps to perform linear convolution using circular convolution.
4. Define ? Convolution
5. What are the properties of convolution?
6. What is zero padding in convolution?
7. What is the difference between convolution & correlation?
8. What is half wave symmetry?
9. What is Dirchlet condition?
10. How to compute maximum length N for a circular convolution using DFT and IDFT.
11. What is sectioned convolution?
12. What is the advantage of sectioned convolution?
13. How to perform linear convolution using circular convolution?
14. What is the difference between linear shifting and circular shifting of signal?
15. What is the length of output sequence after circular convolution?
16. State the circular convolution property of DFT
17. What is circular convolution?
18. What is the length of output sequence after linear convolution?
19. List the properties of linear and circular convolution.
20. What is linear convolution?


Viva ?voce
15 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.3 SPECTRUM ANALYSIS USING DFT
Aim:
To write a program to analyze the spectrum of a signal using Discrete Fourier Transform (DFT) and
Fast Fourier Transform (FFT) using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
1. Start
2. Get the input sequence.
3. Obtain the DFT of input sequence.
4. Find the magnitude and phase response of the DFT.
5. Plot the magnitude and phase in continuous and in discrete forms.
6. Stop
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the spectrum of a signal was analyzed using Discrete Fourier Transform (DFT).
Outcome:
After finishing this experiment the students are able to analyze the spectrum of the signal using Discrete
Fourier Transform (DFT)
Applications:
1. Spectral analysis of biomedical signals, RF signals, speech and seismic signals
2. Implementation of filters using circuits
3. Correlation of sequences in RADAR

16 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

4. Digital communication systems for TV broadcasting




1. What is the difference between DFT and IDFT?
2. Differentiate DIT from DIF algorithm.
3. What is decimation in frequency?
4. What is decimation in time?
5. How many multiplication terms are required to perform DFT by expressional method and FFT
method?
6. How many stages are required for 8 point DFT?
7. Compute inverse DFT of X(k) = {3,4,5,6}.
8. How to calculate output of DFT using MATLAB?
9. List the properties of FFT.
10. List the properties of DFT
11. What is the difference between FFT and IFFT?
12. What is "Bit reversal"?
13. List the advantages of FFT.
14. When the DFT X(k) of a sequence x(n) is imaginary?
15. What?s the difference between FFT and DFT?
16. Define ? Discrete Fourier series
17. When the DFT X(k) of a sequence x(n) is real?
18. What is the importance of FFT in OFDMA technology?
19. What is the advantage of Fourier transform in DSP?
20. What is twiddle factor?




Viva ?voce
17 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.4 DESIGN OF FIR FILTERS

Aim:
To write a program to design and obtain the magnitude and phase responses for the given FIR filter
using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
Low pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
High pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band pass filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Band stop filter using rectangular window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
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4. Find the window co-efficient using the MATLAB function boxcar (N)
5. Draw the magnitude and phase response.
Low pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
High pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band pass filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Band stop filter using hamming window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function hamming (N)
5. Draw the magnitude and phase response.
Low pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
19 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

5. Draw the magnitude and phase response.
High pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band pass filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Band stop filter using Hanning window
1. Get the pass band and stop band edge frequencies.
2. Get the sampling frequency
3. Get the order of the filter (N)
4. Find the window co-efficient using the MATLAB function Hanning (N)
5. Draw the magnitude and phase response.
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the FIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Able to Implement Low pass, High pass and Band pass FIR filter for a given sequence
2. Calculate the filter coefficients.

20 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Applications:
1. Implementation of circuits for filtering operations
2. Study of frequency response of filters
3. Audio applications
4. Image processing and music processing


1. What are the specifications required to design a filter?
2. List the design techniques for linear phase FIR filter design.
3. Distinguish between FIR and IIR filter.
4. What is window method? List the steps to design FIR filter using window method.
5. What is the MATLAB command for Hamming window?
6. What are the advantages of FIR Filters (compared to IIR filters)?
7. What are the disadvantages of FIR Filters (compared to IIR filters)?
8. What are the desirable characteristics of the window?
9. What are the difference types of filters based on impulse response?
10. What are the difference types of filters based on frequency response?
11. What are the desirable and undesirable features of FIR filters?
12. Write the expression for Hamming and Blackman windows?
13. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
14. Write the expression for Bartlett and Hamming windows?
15. What is the principle of designing FIR filter using frequency sampling method?
16. What is the need for employing window technique for FIR filter design?
17. What are the advantages of Kaiser window?
18. What is Gibbs phenomenon?
19. What are the specifications required to design filter?
20. Name the window function which has less peak amplitude of side lobe.


Viva ?voce
21 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.5 DESIGN OF IIR FILTERS
Aim:
To write a program to design and obtain the response of the given IIR filter using MATLAB
Apparatus Required:
PC with MATLAB software
Algorithm:
A digital filter, the derivation of the digital filter transfer function requires three steps.
1. Map the desired digital filter specification into those for an equivalent analog filter.
2. Derive the analog transfer function for the analog prototype.
The various parameters in the figure are,
?p = Pass band frequency in radians
?s = Stop band frequency in radians
?c = 3-db cutoff frequency in radians
? = Parameter specifying allowable pass band
? = Parameter specifying allowable stop band
Design of IIR filters from analog filters:
There are several methods that can be used to design digital filters having an infinite duration unit
sample response. The techniques described are all based on converting an analog filter into digital filter. If
the conversion technique is to be effective, it should posses the following desirable properties.
The j ?-axes in the s-plane should map into the unit circle in the z-plane. Thus there will be a
direct relationship between the two frequency variables in the two domains.
The left-half plane of the s-plane should map into inside of the unit circle in the z-plane. Thus a
stable analog filter will be converted to a stable digital filter.
Algorithm:
Butterworth low pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
22 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Butterworth high pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band pass filter
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band edge frequencies.
3. Get the sampling frequency.
4. Calculate the order the filter
5. Find the filter coefficients.
6. Draw the magnitude and phase response.
Butterworth band stop filter
1. Get the pass band and stop band ripples
2. Get the pass band and stop band edge frequencies
3. Get the sampling frequencies
4. Calculate the order of the filter
5. Find the filter co-efficient
6. Draw the magnitude and phase response
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in figure window.
Result:
Thus the IIR filter was designed for the given specifications and its magnitude and phase responses
were observed.
Outcome:
After finishing this experiment the students are able to:
1. Implement IIR filter for given specifications
23 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

2. Plot the response of the same.
Applications:
1. Implementation of filter circuits
2. Speech processing and video processing
3. Radio transmission
4. Filters in various industrial applications


1. Define ? IIR Filter
2. List the steps to design a digital filter using impulse invariant method.
3. What are the disadvantages of impulse invariant method?
4. Differentiate recursive & non-recursive systems.
5. List the parameters obtained from the Chebyshev filter specifications.
6. What is warping effect? Where we found this effect? How can we eliminate warping effect
7. What is the difference between type I & type II Chebyshev filters?
8. Define ? Cut-off frequency
9. Compare direct form I and direct form II realization of IIR filter.
10. List the digital techniques used to convert analog filter to digital filter.
11. List the properties of Butterworth low pass filter.
12. What is the role of delay element in filter design?
13. List the properties of Chebyshev low pass filter.
14. What are the properties of bilinear transformations?
15. Distinguish between the frequency responses of Chebyshev type I filter for N odd and N even.
16. Write the expression of the order of the Butterworth filter.
17. Define ? Prewarping
18. List the steps to design digital filter from analog filter.
19. How many number of additions, multiplications and memory locations are required to realize a
digital system in IIR filter?
20. Write the expression of the order of the Chebyshev filter.
Viva ?voce
24 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


Expt.No.6 MULTIRATE FILTERS
Aim
To design multirate FIR filters using interpolation and decimation and to plot the impulse responses and
the magnitude responses for all designs.
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the multirate filters was designed using interpolation and decimation for the given input.
Outcome:
After finishing this experiment the students are able to design linear-phase FIR multirate filters and do
interpolation and decimation of the signals
Applications:
1. Sampling rate conversion systems
2. Communication systems
3. Design of phase shifters
4. Antenna arrays





25 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00



1. What is the importance of decimation for a given signal/sequence?
2. What is aliasing? What is the condition to avoid aliasing for sampling?
3. Define ? Decimation
4. Define ? Interpolation
5. What is multirate signal processing?
6. Name the process of converting a signal from a given rate to a different rate
7. State the formula for poly phase interpolation using z-transform.
8. List the applications of multirate signal processing.
9. List the methods used in sampling for converting a digital signal
10. List the advantages of multirate processing.
11. What is up sampling?
12. List the applications of multirate DSP.
13. What is the need for antialiasing filter prior to down sampling?
14. List the applications of adaptive filters
15. What are the sections in Quadrature mirror filter bank?
16. What is down sampling?
17. What is the need for anti-imaging filter after up sampling a signal?
18. State the formula for poly phase decimation using z-transform.
19. What is anti-aliasing filter?
20. Define ? Sampling theorem







Viva ?voce
26 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt.No.7 EQUALIZATION

Aim
To write MATLAB program to design adaptive channel equalization using LMS algorithm
Apparatus Required:
PC with MATLAB software
Procedure:
1. Enter the program in workspace.
2. Save and Run it in .m files.
3. Observe the output in command window or in figure window.
Result:
Thus the adaptive channel equalization was designed using LMS algorithm.
Outcome:
After finishing this experiment the students are able to design adaptive channel equalization using LMS
algorithm.
Applications:
1. Echo cancellation in telecommunication systems
2. Digital radio communication systems
3. Channel equalization for dispersive channels
4. Signal processing and control applications








27 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00


1. Define ? Equalization
2. What is recursive algorithm?
3. List the steps in the process of Adaptive equalization.
4. What is decision directed mode?
5. What are the uses of adaptive filters?
6. Define ? Rate of Convergence
7. What is the relation between analog and digital filter?
8. Name the algorithm used in Adaptive Equalization.
9. What is the impact of LMS algorithm?
10. What is the relation between analog and digital filter?
11. What is the factor for designing an equalizer?
12. What is non-recursive algorithm?
13. What is the purpose of receiver decision?
14. What is difference between recursive and non-recursive algorithm?
15. What is the maximum expected time delay spread in the channel?













Viva ?voce
28 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Expt. No. 8 STUDY OF DSP PROCESSOR TMS320C50
Aim:
To study the architecture overview of DSP processor TMS320C50
Introduction:
It is needless to say that in order to utilize the full feature of the DSP chip TMS320C50, the DSP
engineer must have a complete knowledge of the DSP device. This chapter is an introduction to the
hardware aspects of the TMS320C50. The important units of TMS320C50 are discussed.
The DSP Chip TMS320C50:
The TMS320C50 is a 16-bit fixed point digital signal processor that combines the flexibility of a high
speed controller with the numerical capability of an array processor, thereby offering an inexpensive
alternative to multichip bit-slice processors. The highly paralleled architecture and efficient instruction
set, provide speed and flexibility capable of executing 10 MIPS (Million Instructions Per Second). The
TMS320C50 optimizes speed by implementing functions in hardware that other processors implement
through microcode or software. This hardware intensive approach provides the design engineer with
processing power previously unavailable on a single chip.
The TMS320C50 is the third generation digital signal processor in the TMS320 family. Its powerful
instruction set, inherent flexibility, high-speed number-crunching capabilities, and innovative
architecture have made this high-performance, cost-effective processor the ideal solution to many
telecommunications, computer, commercial, industrial, and military applications.
Key Features of TMS320C50:
The key features of the Digital Signal Processor TMS320C50 are:
? 35-/50-ns single-cycle fixed-point instruction execution time (28.6/20 MIPS)
? Upward source-code compatible with all C1X and C2x devices
? RAM-based memory operation (C50)
? 9K x 16-bit single-cycle on-chip program/data RAM (C50)
? 2K x 16-bit single-cycle on-chip boot ROM (C50)
? 1056 x 16-bit dual-access on-chip data RAM
? 224K x 16-bit maximum addressable external memory space (64K program, 64K data, 64K I/O)
29 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

? 32-bit arithmetic logic unit (ALU), 32-bit accumulator (ACC), and 32-bit accumulator buffer
(ACCB)
? 16-bit parallel logic unit (PLU)
? 16 x 16-bit parallel multiplier with a 32-bit product capability.
? Single-cycle multiply/accumulate instructions
? Eight auxiliary registers with a dedicated auxiliary register arithmetic unit for indirect
addressing.
? Eleven context-switch registers (shadow registers) for storing strategic CPU controlled registers
during an interrupt service routine
? Eight-level hardware stack
? 0- to 16-bit left and right data barrel-shifters and a 64-bit incremental data shifter
? Two indirectly addressed circular buffers for circular addressing
? Single-instruction repeat and block repeat operations for program code
? Block memory move instructions for better program/data management
? Full-duplex synchronous serial port for direct communication between the C5x and another
serial device
? Time-division multiple-access (TDM) serial port
? Interval timer with period, control, and counter registers for software stop, start, and reset
? 64K parallel I/O ports, 16 of which are memory mapped
? Sixteen software programmable wait-state generators for program, data, and I/O
memory spaces.





30 Format No.: FirstRanker/Stud/LM/34/Issue: 00/Revision: 00

Theory:
A detailed architectural block diagram of TMS320C50 is illustrated in Figure. The TMS320C50
utilizes a modified Harvard architecture for speed and flexibility. In a strict Harvard architecture,
program and data memory are in two separate spaces, permitting a full overlap of instruction fetch and
execution. The TMS320 family's modification of the Harvard architecture allows transfer between
program and data spaces, thereby increasing the flexibility of the device. This modification permits
coefficients stored in program memory to be read into the data RAM, eliminating the need for a
separate coefficient ROM. It also makes available immediate instructions and subroutines based on
computed values.
32-bit Accumulator:
The TMS320C50 contains a 32-bit ALU and accumulator for support of double- precision, two's
complement arithmetic. The ALU is a general purpose arithmetic unit that operates on 16-bit words
taken from the data RAM or derived from immediate instructions. In addition to the usual arithmetic
instructions, the ALU can perform Boolean operations, providing the bit manipulation ability required of
a high-speed controller. The accumulator stores the output from the ALU and is often an input to the
ALU. Its word length is 32-bit. The accumulator is divided into a high order word (bits 31 through 16)
and a low-order word (bits 15 through 0). Instructions are provided for storing and loading the high and
lower order accumulator words to memory.
16 X 16-bit Parallel Multiplier:
The multiplier performs a 16 x 16-bit two's complement multiplication with a 32- bit result in a single
instruction cycle. The multiplier consists of three units: the T- Register, P-Register, and multiplier array.
The 16-bit T-Register temporarily stores the multiplicand and the P-Register stores the 32-bit product.
Multiplier values either come from the data memory or are derived immediately from the MPY (multiply
immediate) instruction word. The fast on-chip multiplier allows the device to perform fundamental
operations such as convolution, correlation, and filtering. Two multiply/accumulate instructions in the
instruction set fully utilize the computational bandwidth of the multiplier, allowing both operands to be
processed simultaneously.
Shifters:
A 16-bit scaling shifter is available at the accumulator input. This shifter produces a left shift of 0 to
1