DC Course File

Digital
Communications
COURSE FILE
(AY 2018-19)
ECE DEPARTMENT DIGITAL COMMUNICATIONS

Contents required for course file
Courses file contents:
1. Cover Page
2. Vision of the Institute
3. Mission of the Institute
4. Vision of the Department
5. Mission of the Department
6. PEOs and POs
7. Syllabus copy
8. Course objectives and outcomes
9. Brief note on the course & how it fits in to the curriculum
10. Prerequisite, if any.
11. Instructional Learning Outcomes
12. Course mapping with PEOs and POs
13. Class Time Table
14. Individual Time Table
15. Lecture plan with methodology being used/adopted.
16. Assignment questions
17. Tutorial problems
18. a) Unit wise question bank
b) Unit wise Quiz Questions and long answer questions
19. Detailed notes
20. Additional topics, if any.
21. Known gaps ,if any and inclusion of the same in lecture schedule
22. Discussion topics, if any
23. University Question papers of previous years
24. References, Journals, websites and E-links
25. Quality Control Sheets. a)course end survey b)Teaching Evaluation
c) CO- attainment.
26. Student List
27. Group-Wise students list for discussion topics

GEETHANJALI COLLEGE OF ENGINEERING AND TECHNOLOGY
(Autonomous)
Department of Electronics and Communication Engineering
Name of the Subject : Digital Communications

Course Code: 16EC3202 Level: UG Programme: B. Tech
Branch: ECE Year & Semester: III – II
Section:– A/B/C/D/E Document No. GCET/ECE/III-II/DC
Classification status (Unrestricted / Restricted ) : Unrestricted
Distribution List : Dept. Library, Concerned Faculty
Prepared by Updated by
1) Name : Ms.S. Krishna Priya 1) Name : D.Venkata Rami Reddy
2) Sign : 2) Sign :
3) Design : Assoc. Prof. 3) Design : Assoc. Prof.
4) Date : 02-12-2017 4) Date : 20 – 12 - 2018
Verified by : For Q.C Only.
1) Name : 1) Name :
2) Sign : 2) Sign :
3) Design : 3) Design :
4) Date : 4) Date :
Approved by : Head of the Department
1) Name : Prof. B. Hari Kumar
2) Sign :
3) Date :

2. Vision of the Institution
Geethanjali visualizes dissemination of knowledge and skills to students, who would

eventually contribute to the well being of the people of the nation and global community.
3. Mission of the Institution

i. To impart adequate fundamental knowledge in all basic sciences and engineering,
technical and inter-personal skills to students.
ii. To bring out creativity in students that would promote innovation, research and
entrepreneurship.
iii. To preserve and promote cultural heritage, humanistic and spiritual values promoting
peace and harmony in society.
4. Vision of the Department

To impart quality technical education in Electronics and Communication Engineering
emphasizing analysis, design/synthesis and evaluation of hardware/embedded software using
various Electronic Design Automation (EDA) tools with accent on creativity, innovation and
research thereby producing competent engineers who can meet global challenges with
societal commitment.
5. Mission of the Department
i. To impart quality education in fundamentals of basic sciences, mathematics, electronics

and communication engineering through innovative teaching-learning processes.
ii. To facilitate Graduates define, design, and solve engineering problems in the field of
Electronics and Communication Engineering using various Electronic Design Automation
(EDA) tools.
iii. To encourage research culture among faculty and students thereby facilitating them to be
creative and innovative through constant interaction with R & D organizations and
Industry.
iv. To inculcate teamwork, imbibe leadership qualities, professional ethics and social
responsibilities in students and faculty

6. PEOs and POs
6.1 Program Educational Objectives:
I. To prepare students with excellent comprehension of basic sciences, mathematics

and engineering subjects facilitating them to gain employment or pursue
postgraduate studies with an appreciation for lifelong learning.
II. To train students with problem solving capabilities such as analysis and design
with adequate practical skills wherein they demonstrate creativity and innovation
that would enable them to develop state of the art equipment and technologies of
multidisciplinary nature for societal development.
III. To inculcate positive attitude, professional ethics, effective communication and

interpersonal skills which would facilitate them to succeed in the chosen
profession exhibiting creativity and innovation through research and
development both as team member and as well as leader.
6.2 Program Outcomes of B.Tech ECE Program: (POs)

1. Engineering knowledge: Apply the knowledge of mathematics, science, engineering
fundamentals, and an engineering specialization to the solution of complex engineering
problems.
2. Problem analysis: Identify, formulate, research literature, and analyze complex
engineering problems reaching substantiated conclusions using first principles of
mathematics, natural sciences, and engineering sciences.
3. Design/development of solutions: Design solutions for complex engineering
problems and design system components or processes that meet the specified needs with
appropriate consideration for the public health and safety, and the cultural, societal, and
environmental considerations.
4. Conduct investigations of complex problems: The problems
 That cannot be solved by straightforward application of knowledge, theories and
techniques applicable to the engineering discipline.
 That may not have a unique solution. For example, a design problem can be solved
in many ways and lead to multiple possible solutions.
 That requires consideration of appropriate constraints/requirements not explicitly
given in the problem statement. (like: cost, power requirement, durability, product
life, etc.).
 Which need to be defined (modeled) within appropriate mathematical framework?
 That often requires use of modern computational concepts and tools.
5. Modern tool usage: Create, select, and apply appropriate techniques, resources, and
modern engineering and IT tools including prediction and modeling to complex
engineering activities with an understanding of the limitations.
6. The engineer and society: Apply reasoning informed by the contextual knowledge to
assess societal, health, safety, legal and cultural issues and the consequent
responsibilities relevant to the professional engineering practice.
7. Environment and sustainability: Understand the impact of the professional
engineering solutions in societal and environmental contexts, and demonstrate the
knowledge of, and need for sustainable development.

8. Ethics: Apply ethical principles and commit to professional ethics and responsibilities
and norms of the engineering practice.
9. Individual and team work: Function effectively as an individual, and as a member
or leader in diverse teams, and in multidisciplinary settings.
10. Communication: Communicate effectively on complex engineering activities with
the engineering community and with society at large, such as, being able to comprehend
and write effective reports and design documentation, make effective presentations, and
give and receive clear instructions.
11. Project management and finance: Demonstrate knowledge and understanding of the
engineering and management principles and apply these to one’s own work, as a member
and leader in a team, to manage projects and in multidisciplinary environments.
12. Life-long learning: Recognize the need for, and have the preparation and ability to
engage in independent and life-long learning in the broadest context of technological
change.
Program Specific Outcomes (PSOs):
1. An ability to design an Electronics and Communication Engineering system, component,

or process and conduct experiments, analyze, interpret data and prepare a report with
conclusions to meet desired needs within the realistic constraints such as economic,
environmental, social, political, ethical, health and safety, manufacturability and
sustainability.
2. An ability to use modern Electronic Design Automation (EDA) tools, software and
electronic equipment to analyze, synthesize and evaluate Electronics and Communication
Engineering systems for multidisciplinary tasks.

7. Syllabus Copy

8. COURSE OBJECTIVES AND OUTCOMES
Course Objectives:
Develop ability to:
1. Understand different digital coding techniques such as PCM, DM, DPCM and
Analyze different Digital Baseband Transmission techniques
2. Describe the concepts of information theory and source coding techniques
3. Understand the concepts of Error Correction codes namely, Block codes, Cyclic code
and convolution Codes.
4. Understand various digital modulation techniques namely ASK, FSK, BPSK and
QPSK..
5. Understand the characteristics of Spread Spectrum (SS) modulation techniques.
Course outcomes :
After the completion of the course, students will be able to:

CO1: explain the basic elements of Digital communication Systems, base band
transmission and optimum reception of Digital signals, sampling theorem and pulse code
modulation, its advanced versions
CO2: analyze the concepts of Information theory, SNR trade off with channel band
width; develop Huffman coding, Shannon-Fano coding algorithms to increase average
information per bit.
CO 3: develop linear block coding and decoding algorithms, convolutional coding and
decoding algorithms for data transmission and reception; solve numerical problems on
error detection and correction capabilities
CO 4: compare different pass band digital modulation techniques like ASK, FSK and
PSK and compute the Probability of error in each scheme; draw and analyze the eye
diagrams for ASK, FSK and PSK.
CO 5: describe the different criteria in spread spectrum modulation scheme and its
applications
9. Brief note on the course & how it fits in to the curriculum:

Introduction to the subject:
The move to digital modulation provides more information capacity, compatibility

with digital data services, higher data security, better quality communications, and quicker
system availability. Developers of communication systems face the constraints like
availability bandwidth, permissible power, inherent noise level of the system.
The RF spectrum must be shared, yet every day there are more users for that spectrum
as demand for communications services increases. Digital modulation schemes have greater
capacity to convey large amounts of information than analog modulation schemes

10. PREREQUISITES:
 Engineering Mathematics
 Basic Electronics
 Theory of Signals and Systems
 Analog Communications
11. Instructional learning outcomes:
Subject: Digital Communications

UNIT I:
Elements of Digital Communication Systems
DC1: Analyze the elements of digital communication system, the importance and
Applications of Digital Communication.
DC2: The importance and the need of sampling theorem in digital communication systems
and Digital representation of analog signals
DC3. Advantages of digital communication systems over analog systems.
Base Band Transmission and Optimal Digital Signal
DC4: Identify the need of pulse shaping for optimum transmission and get the knowledge of
Base band signal receiver model.
DC5: Explain Probability of error, optimum receiver, coherent reception, Matched filter
and understand the Signal space representation
DC6: Describe cross talk and its effect in the degradation of signal quality in digital
communication
Pulse Coded Modulation
DC7: Explain the generation and reconstruction of PCM.
DC8: Analyze the effect of Quantization noise in Digital Communication.
DC9: Analyze the different digital communication schemes like Differential PCM systems
(DPCM), Delta modulation, and adaptive delta modulation.
DC10: Compare the digital communication schemes like Differential PCM systems
(DPCM), Delta modulation, and adaptive delta modulation.
DC 11: Illustrate the effect of Noise in PCM and DM systems.

UNIT II:
Information Theory and Source coding techniques:
DC12: Identify the basic terminology used in coding of Digital signals like Information and
entropy and Conditional entropy, Mutual information and redundancy.
DC13: Compare different types of channels and explain channel capacity, Hartley-Shannon
law and Bandwidth-SNR tradeoff.
DC14. Explain the need for Source coding and types of Source coding methods.
DC15: Solve problems based on mutual information and Information loss due to noise.
DC16: Compute problems on Source coding methods like - Huffman code, Shannon-Fano
codes used in digital communication to increase average information per bit.
UNIT III:
Linear Block Codes
DC17: Illustrate the different types of codes used in digital communication and the
Matrix description of linear block codes.
DC18: Analyze the Hamming encoder and syndrome decoder and find errors, solve the
numerical problems for Error detection and error correction of Hamming codes
DC19: Construct the algebraic structure of cyclic codes and analyze how encoder and
decoders used to detect and correct errors.
DC20: Compute problems based on the representation of cyclic codes and encoding and
decoding of cyclic codes.
DC21: Solve problems to find the location of error in the codes i.e., syndrome calculation.
Convolution Codes
DC22: Identify the difference between the different codes digital communication.
DC23: Describe Encoding & decoding of Convolutional Codes.
DC24: Solve problems on error detection & correction using state Tree and trellis
diagrams.
DC25: Solve problems based on Viterbi algorithm.
DC26: Compute numerical on error calculations and compare the error rates in coded and
uncoded transmission.

UNIT IV:
Digital Modulation Techniques
DC27: Describe and differentiate the different shift keying formats used in digital
communication.
DC28: Compute the power and bandwidth requirements of modern communication systems
modulation formats like those employing ASK, PSK, FSK, and QAM.
DC29-32: Explain the different modulators like ASK Modulator, Coherent ASK detector,
non-Coherent ASK detector, Band width frequency spectrum of FSK, Non-Coherent FSK
detector, Coherent FSK detector.
DC33: Analyze the need and use of PLL in FSK Detection.
DC34: Differentiate the different keying schemes -BPSK, Coherent PSK detection, QPSK
& Differential PSK.
DC35: Draw and analyze eye diagrams and compare performance of various modulation
techniques
UNIT V:
Spread Spectrum Modulation
DC 36: Analyze the need and use of spread spectrum in digital communication and gain
knowledge of spread spectrum techniques like direct sequence spread spectrum (DSSS).
DC 37: Describe Code division multiple access, ranging using DSSS Frequency Hopping
spread spectrum.
DC38: Generate PN sequences and solve problems based on sequence generation.
DC39: Explain the need of synchronization in spread spectrum system.
DC40: Identify the Advancements in the digital communication.
12.Course mapping with PEOs and POs:

Mapping of Course with Programme Educational Objectives:
Course code course Semester PEO PEO PEO

S.No component I II III
Digital
1 Communication 16EC3202 II 2 3 2
Communications

Mapping of Course outcomes with Programme outcomes:
COMMUNICATION
POs 1 2 3 4 5 6 7 8 9 1 11 1 PSO 1 PSO 2
0 2
Digital Communications 1 2 3 2 2 2 1 2 2 3
CO1: explain the basic 1 2 3 2 2 2 1 2 2 3

elements of Digital
communication Systems,
base band transmission and
optimum reception of Digital
signals, sampling theorem
and pulse code modulation,
its advanced versions
CO2: analyze the concepts of 1 2 3 2 2 2 1 2 2 3

Information theory, SNR
trade off with channel band
width; develop Huffman
coding, Shannon-Fano coding
algorithms to increase
average information per bit.
CO 3: develop linear block 1 2 3 2 2 2 1 2 2 3

coding and decoding
algorithms, convolutional
coding and decoding
algorithms for data
transmission and reception;
solve numerical problems on
error detection and correction
capabilities
CO 4: compare different pass 1 2 3 2 2 2 1 2 2 3

band digital modulation
techniques like ASK, FSK
and PSK and compute the
Probability of error in each
scheme; draw and analyze the
eye diagrams for ASK, FSK
and PSK.
CO 5: describe the different 1 2 3 2 2 2 1 2 2 3

criteria in spread spectrum
modulation scheme and its
applications

13.Class Time Tables:
Geethanjali College of Engineering & Technology
Department of Electronics & Communication Engineering
Year/Sem/Sec: III-B.Tech-II Semester-A Section Acad Year 2018-19, WEF: 10-12-2018
Class Incharge: M.Krishna (Room No.:314) Version - 1
12.20- 1.00-
Time 09.00-09.50 09.50-10.40 10.40-11.30 11.30-12.20 01.00 1.50-2.40 2.40-3.30
1.50
Period 1 2 3 4 5 6 7
Monday PE-I VLSI& DDVH Labs / DSP Lab HVPE DC CSE
Tuesday PE-II PE-I PE-I PE-II SC-I Finishing School
LUNCH
Wednesday DSP DSP CSE DC VLSI& DDVH Labs / DSP Lab
Library/Sports/
Thursday SC-I SC-I DSP DSP* BEC/ Mentoring CACG
Friday PE-II HVPE DC PE-II* CSE Finishing School
Saturday CSE* HVPE DC* SC-I* PE-I* Remedial Classes
S.No Subject(T/P) Faculty Name Subject Code Periods/Week
1 Digital Signal processing Dr.C.Venkata Narasimhulu / K.Victor 16EC3201 3+1*
2 Digital Communications S.Krishna Priya 16EC3202 3+1*
3 Control Systems Engineering M.Krishna 16EC3203 3+1*
Professional Elective – I (PE-I)
Electronic Instrumentation and Measurements A.Subramanyam 16EC3204
4 Telecommunication Switching Systems and 3+1*
- 16EC3205
Networks
Digital Systems Design - 16EC3206
Professional Elective – II (PE-II)
Optical Communications - 16EC3207
5 3+1*
Computer Architecture and Organization M.Laxmi 16EC3208
Computer Networks - 16CS3212
Soft Core – I (SC-I) 3+1*
6 Digital Design through Verilog HDL G.SreeLakshmi 16EC3209 Room No: 314
VLSI Design Prof.OVPR.Siva Kumar / B.Mamatha 16EC3210 Room No: 320
Digital Signal Processing Lab K.Victor, Y.Siva Rama Krishna
7 16EC32L1 3
Lab Technician: D.Vivekananda
Soft Core - I lab
Digital Design through Verilog HDL Lab G.SreeLakshmi, M.Laxmi
16EC32L2
8 Lab Technician: ARL.Padmaja 3
VLSI Lab B.Mamatha, B.Jugal Kishore
16EC32L3
Lab Technician: M.Chathar Singh, K.Chalapathi Rao
9 Human Values and Professional Ethics J.Vijaya Lakshmi 16MB32P1 3
10 BEC/SoftSkills G.Karuna Kumari, Dr.B.Nagamani - 2
11 Library/Sports - - 1
12 Mentoring - - -
13 CACG Dr.K.Madhumati - -
14 Finishing School Section Incharge: B.Ramu - 4
15 Remedial Classes - - 2
* indicates tutorial hour Date: 14-11-2018
TT Coord:___________ HOD:____________ Dean Academics:___________Principal:____________

Year/Sem/Sec: III-B.Tech-II Semester-B Section Acad Year 2018-19, WEF: 10-12-2018
Class Incharge: P.Sudhakar (Room No.:320) Version - 1
12.20- 1.00-
Time 09.00-09.50 09.50-10.40 10.40-11.30 11.30-12.20 01.00 1.50-2.40 2.40-3.30
1.50
Period 1 2 3 4 5 6 7
Monday PE-II VLSI& DDVH Labs / DSP Lab DC DC CSE
Tuesday PE-I PE-I CSE CSE SC-I Finishing School
LUNCH
Wednesday DC DC HVPE HVPE VLSI& DDVH Labs / DSP Lab
Library/Sports/
Thursday SC-I SC-I DSP DSP BEC/ Mentoring CACG
Friday PE-I PE-I* PE-II PE-II DSP* Finishing School
Saturday CSE* PE-II* DSP* SC-I* HVPE Remedial Classes
1 Digital Signal processing Dr.V.Vineel Kumar 16EC3201 3+1*
2 Digital Communications P.Sudhakar 16EC3202 3+1*
3 Control Systems Engineering A.Sowjanya 16EC3203 3+1*
Electronic Instrumentation and Measurements - 16EC3204
- 16EC3205
Networks
Digital Systems Design Dr.S.Udaya Kumar / D.Sony 16EC3206
5 3+1*
Computer Architecture and Organization - 16EC3208
Computer Networks B.Sreelatha 16CS3212
Soft Core – I (SC-I) 3+1*
6 Digital Design through Verilog HDL G.SreeLakshmi 16EC3209 Room No: 314
VLSI Design Prof.OVPR.Siva Kumar / B.Mamatha 16EC3210 Room No: 320
Digital Signal Processing Lab L.Kavya, M.Anand
7 16EC32L1 3
Lab Technician: D.Venkateshwarlu
Soft Core - I lab
Digital Design through Verilog HDL Lab G.SreeLakshmi, M.Laxmi
16EC32L2
8 Lab Technician: ARL.Padmaja 3
VLSI Lab B.Mamatha, B.Jugal Kishore
16EC32L3
12 Mentoring - - -
14 Finishing School Section Incharge: B.Ramu - 4

Year/Sem/Sec: III-B.Tech-II Semester-C Section Acad Year 2018-19, WEF: 10-12-2018

Class Incharge: Dr.V.Vineel Kumar (Room No.:321) Version - 1
12.20- 1.00-
Time 09.00-09.50 09.50-10.40 10.40-11.30 11.30-12.20 01.00 1.50-2.40 2.40-3.30
1.50
Period 1 2 3 4 5 6 7
Monday PE-I PE-I HVPE DC VLSI(C1) / DSP(C2)
Tuesday PE-I DSP SC-I SC-I PE-I* Finishing School
LUNCH
Wednesday PE-II VLSI(C2) / DSP(C1) DC HVPE CSE
Library/Sports/
Thursday PE-II PE-II CSE CSE BEC/ Mentoring CACG
Friday DSP DSP SC-I SC-I* DC Finishing School
Saturday HVPE DC* CSE* PE-II* DSP* Remedial Classes
1 Digital Signal processing Dr.V.Vineel Kumar 16EC3201 3+1*
2 Digital Communications S.Krishna Priya 16EC3202 3+1*
3 Control Systems Engineering M.Krishna 16EC3203 3+1*
Professional Elective – I (PE-I) 3+1*
Electronic Instrumentation and Measurements - 16EC3204 -
4 Telecommunication Switching Systems and
Y.Naga Lakshmi 16EC3205 Room No: 322
Networks
Digital Systems Design K.Somasekhara Rao / Ch.Sandeep 16EC3206 Room No: 321
Professional Elective – II (PE-II) 3+1*
Optical Communications M.Anand 16EC3207 Room No: 321
5
Computer Architecture and Organization - 16EC3208 -
Computer Networks J.Bharathi 16CS3212 Room No: 322
Soft Core – I (SC-I)
6 Digital Design through Verilog HDL - 16EC3209 3+1*
VLSI Design B.Jugal Kishore 16EC3210
Digital Signal Processing Lab Dr.V.Vineel Kumar, B.Ramu
7 16EC32L1 3
Lab Technician: D.Vivekananda, ARL.Padmaja
Soft Core - I lab
Digital Design through Verilog HDL Lab - 16EC32L2
8 3
VLSI Lab B.Jugal Kishore, S.Yagnasree
16EC32L3
9 Human Values and Professional Ethics B.P.S.Jyothi 16MB32P1 3
12 Mentoring - - -
14 Finishing School Section Incharge: M.Anand - 4

Year/Sem/Sec: III-B.Tech-II Semester-D Section Acad Year 2018-19, WEF: 10-12-2018
Class Incharge: Y.Siva Rama Krishna (Room No.:118) Version - 1

12.20- 1.00-
Time 09.00-09.50 09.50-10.40 10.40-11.30 11.30-12.20 01.00 1.50-2.40 2.40-3.30
1.50
Period 1 2 3 4 5 6 7
Monday DSP DSP HVPE PE-I DC* SC-I CSE
Tuesday CSE DC PE-II PE-II HVPE Finishing School
LUNCH
Wednesday DC DC CSE SC-I DSP HVPE PE-I
Library/Sports/
Thursday PE-II VLSI(D1) / DSP(D2) BEC/ Mentoring CACG
Friday DSP* CSE* SC-I SC-I* PE-I Finishing School
Saturday PE-I* VLSI(D2) / DSP(D1) PE-II* Remedial Classes
1 Digital Signal processing R.Odaiah 16EC3201 3+1*
2 Digital Communications D.Venkata Rami Reddy 16EC3202 3+1*
3 Control Systems Engineering Y.Siva Rama Krishna 16EC3203 3+1*
Electronic Instrumentation and Measurements A.Subramanyam 16EC3204
- 16EC3205
Networks
Digital Systems Design - 16EC3206
5 3+1*
Computer Architecture and Organization - 16EC3208
Computer Networks B.Sreelatha 16CS3212
VLSI Design Ch.Sandeep 16EC3210
Digital Signal Processing Lab R.Odaiah, Y.Siva Rama Krishna
7 16EC32L1 3
Soft Core - I lab
8 3
VLSI Lab Ch.Sandeep, M.Chathar Singh
16EC32L3
Lab Technician: K.Chalapathi Rao
12 Mentoring - - -
14 Finishing School Section Incharge: V.Sirisha - 4

Year/Sem/Sec: III-B.Tech-II Semester-E Section Acad Year 2018-19, WEF: 10-12-2018
Class Incharge: L.Kavya (Room No.:322) Version - 1
Time 09.00-09.50 09.50-10.40 10.40-11.30 11.30-12.20 12.20- 1.00- 1.50-2.40 2.40-3.30

01.00 1.50
Period 1 2 3 4 5 6 7
Monday PE-I PE-I CSE CSE HVPE DC DSP
Tuesday PE-I VLSI(E1) / DSP(E2) PE-I* Finishing School
LUNCH
Wednesday PE-II CSE DC DC SC-I SC-I HVPE
Library/Sports/
Thursday PE-II PE-II DSP DSP BEC/ Mentoring CACG
Friday HVPE VLSI(E2) / DSP(E1) SC-I Finishing School
Saturday DC* CSE* SC-I* PE-II* DSP* Remedial Classes
1 Digital Signal processing L.Kavya 16EC3201 3+1*
2 Digital Communications B.Mamatha 16EC3202 3+1*
3 Control Systems Engineering A.Shankar 16EC3203 3+1*
Professional Elective – I (PE-I) 3+1*
MWE&DC
Electronic Instrumentation and Measurements P.Chandra Prakash Reddy 16EC3204
Lab[315]
4
Telecommunication Switching Systems and
- 16EC3205 -
Networks
Digital Systems Design K.Somasekhara Rao / Ch.Sandeep 16EC3206 Room No: 321
Professional Elective – II (PE-II) 3+1*
Optical Communications M.Anand 16EC3207 Room No: 321
5
Computer Architecture and Organization - 16EC3208 -
Computer Networks J.Bharathi 16CS3212 Room No: 322
VLSI Design M.Krishna Chaitanya 16EC3210
Digital Signal Processing Lab A.Shankar, L.Kavya, K.Victor
7 16EC32L1 3
Soft Core - I lab
8 M.Krishna Chaitanya, B.Mamatha, 3
VLSI Lab
M.Chathar Singh 16EC32L3
Lab Technician: K.Chalapathi Rao
9 Human Values and Professional Ethics B.P.S.Jyothi 16MB32P1 3
12 Mentoring - - -
14 Finishing School Section Incharge: V.Sirisha - 4

14.INDIVIDUAL TIME TABLE
Name of the Faculty: S. Krishna Priya Workload:14
09.00- 09.50- 10.40- 11.30- 12.20- 1.00- 1.50-
Time 2.40-3.30
09.50 10.40 11.30 12.20 01.00 1.50 2.40
Period 1 2 3 4 5 6 7
DC-
Monday DC-IIIC
IIIA
Tuesday SMII Lab-IIE
LUNCH
DC-
Wednesday DC-IIIA
IIIC
Thursday SMII Lab-IIE
DC-
Friday DC-IIIA
IIIC
Saturday DC-IIIC DC-IIIA Dept Meetings
Name of the Faculty: P.Sudhakar Workload: 11+3
09.00- 09.50- 10.40- 11.30- 12.20- 1.00-
Time 1.50-2.40 2.40-3.30
09.50 10.40 11.30 12.20 01.00 1.50
Period 1 2 3 4 5 6 7
DC-
Monday DC-IIIB
IIIB
Tuesday
LUNCH
Wednesday DC-IIIB DC-IIIB GS-IID
Thursday
RS- RS- RS-
Friday
IV(C&D) IV(C&D) IV(C&D)
RS-
Saturday Dept Meetings
IV(C&D)
Name of the Faculty: D. Venkata Rami Reddy Workload:13
09.00- 09.50- 10.40- 11.30- 12.20- 1.00-
Time 1.50-2.40 2.40-3.30
09.50 10.40 11.30 12.20 01.00 1.50
Period 1 2 3 4 5 6 7
DC-
Monday
IIID
LUNCH
Tuesday DC-IIID AC-IIE

Wednesday DC-IIID DC-IIID
Thursday
Friday GS-IIE AC-IIE
Saturday Dept Meetings
Name of the Faculty: B. Mamatha Workload: 20
09.00- 09.50- 10.40- 11.30- 12.20- 1.00-
Time 1.50-2.40 2.40-3.30
09.50 10.40 11.30 12.20 01.00 1.50
Period 1 2 3 4 5 6 7
Monday VLSI Lab(IIIA&B) DC-IIIE
Tuesday VLSI Lab-IIIE SC-I
LUNCH
Wednesday DC-IIIE DC-IIIE VLSI Lab(IIIA&B)

Thursday SC-I SC-I
Friday VLSI Lab-IIIE
Saturday DC-IIIE SC-I Dept Meetings
15. Lecture Plan with methodology being used/adopted

Unit No.
Sl. Total Topics to be covered Total Date Regular/ Teaching Remarks
no no of no. of Additiona aids used
Periods hours l LCD/OHP
/BB
UNIT I
1 Elements Of Digital 1 Regular OHP,BB
Communication Systems: Model
of digital communication system
2 Model of digital communication 1 Regular OHP,BB
system
Digital representation of analog
signal
3 Certain issues of digital 1 Regular OHP,BB
transmission
4 advantages of digital 1 Regular OHP,BB
communication systems,
5 Sampling theorem 1 Regular OHP,BB
09
6 A Base band signal receiver, 1 Regular OHP,BB
Different pulses and power
spectrum densities
7 Probability of error, optimum 1 Regular OHP,BB
receiver
8 Optimum of coherent reception, 1 Regular OHP,BB
9 Tutorial class-1 1 BB
10 Pulse Coded Modulation: PCM 1 Regular BB

generation and reconstruction ,
Quantization noise
11 Differential PCM systems 1 Regular OHP,BB
(DPCM), Delta modulation,
12 adaptive delta modulation, Noise in 1 Regular OHP,BB
PCM and DM systems
13 Voice Coders 1 Additional BB
07
14 Tutorial Class-2 1 Regular BB
15 Solving University papers 1 OHP,BB
16 Assignment test-1 1
UNIT II
17 Information Theory: Information 1 Regular BB

and entropy
18 Conditional entropy and 1 Regular OHP,BB
redundancy
19 Hartley Shannon law 1 Regular OHP,BB
20 10 trade off of between SNR and BW 1 Regular OHP,BB
21 Shannon Fano coding, mutual 1 Regular OHP,BB

information
22 Information loss due to noise, 1 Regular BB
23 Source codings,- Huffman code, 1 BB

variable length coding
24 Lossy source Coding , Source 1 Regular BB
coding to increase average
information per bit
25 Feedback communications, 1 Additional BB
Assignment Test -2
26 Tutorial Class-3 1 Regular OHP,BB
27 Linear Block Codes: Matrix 1 Regular BB

description of linear block codes

UNIT III
28 Matrix description of linear block 1 Regular BB
codes
29 Error detection and error correction 1 Regular BB
capabilities of linear block codes
30 Error detection and error correction 1 Regular BB
capabilities of linear block codes
31 10 Cyclic codes: algebraic structure, 1 Regular OHP,BB
encoding,
32 syndrome calculation decoding 1 Regular OHPBB
33 Turbo codes 1 Additional OHP,BB
35 Solving University papers 1 OHP,BB
37 Convolution Codes: Encoding, 1 Regular BB

decoding using state
38 Tree and trellis diagrams 1 Regular BB
39 05 Decoding using Viterbi algorithm 1 Regular BB
40 Comparison of error rates in coded 1 Regular OHP,BB

and uncoded transmission
UNIT IV
42 Digital Modulation Techniques: 1 Regular BB

introduction , ASK, ASK
Modulator
43 Coherent ASK detector, non- 1 Regular
Coherent ASK detector
44 Band width frequency spectrum of 1 Regular OHP,BB
FSK, Non-Coherent FSK detector
45 Coherent FSK detector, FSK 1 Regular OHP,BB
Detection using PLL
46 10 BPSK, Coherent PSK detection, 1 Regular BB
47 QPSK, Differential PSK 1 Regular BB
48 Regenerative Repeater 1 Additional OHP,BB
49 Tutorial class-6 1 Regular BB
50 Solving University papers 1 Regular OHP,BB
UNIT V
52 Spread Spectrum Modulation: 1 Regular OHP,BB

Use of spread spectrum, direct
sequence spread spectrum(DSSS)
53 Code division multiple access 1 Regular OHP,BB
54 Ranging using DSSS Frequency 1 Regular

Hopping spread spectrum
55 PN sequences: generation and 1 Regular BB
characteristics
56 Synchronization in spread 1 Regular BB
09 spectrum system
57 Advancements in the digital 1 Missing BB
communication
58 Tutorial Class-7 1 Regular BB

59 Solving University papers 1 Regular OHP,BB
16. Assignment topics
Unit 1:
1. Advantages of digital communication systems,

2. A Base band signal receiver,
3. Different pulses and power spectrum densities
4. Probability of error
3. PCM generation and reconstruction
4. Quantization noise, Differential PCM systems (DPCM)
5. Delta modulation
Unit 2:
1. Conditional entropy and redundancy

2. Mutual information
3. Bandwidth- S/N trade off, and Sampling theorem
4. Shannon Fano coding
5. Huffmann Coding
Unit 3:
1. Matrix description of linear block codes

2. Matrix description of linear block codes
3. Error detection and error correction capabilities of linear block codes
4. Encoding
5. Decoding using state Tree and trellis diagrams
6. Decoding using Viterbi algorithm
Unit 4:
1. Coherent ASK detector and non-Coherent ASK detector

2. Coherent FSK detector BPSK
3. Coherent PSK detection
Unit 5:
1. Use of spread spectrum

2. Direct sequence spread spectrum(DSSS),
3. Code division multiple access
4. Ranging using DSSS Frequency Hopping spread spectrum
17. Tutorial Questions
Tutorial – 1

1. Explain the basic principles of sampling, and distinguish between ideal sampling and
practical sampling.
2. State and prove sampling theorem.
3. Find the transfer function of the optimum receiver and calculate the error probability
4. Show that the impulse response of a matched filter is a time reversed and delayed
version of the input signal .and Briefly explain the properties of matched filter.
5. For the modulating signal m(t) = 2 Cos (100 t) + 18 Cos (2000 πt), determine the
allowable sampling rates and sampling intervals.

Tutorial – 2
6. Prove that the mean value of the quantization error is inversely proportional to the
Square of the number of quantization levels.
7. Explain why quantization noise could affect small amplitude signals in a PCM system
more than large signals. With the aid of sketches show how tapered quantizing level
could be used to counteract this effect.
8. Explain the working of Delta modulation system with a neat block diagram.
clearly bring out the difference between granular noise and slope overload error.
Consider a speech signal with maximum frequency of 3.4 KHz and maximum
Amplitude of 1v.This speech signal applied to a DM whose bit rate is set at
20kbps. Discuss the choice of appropriate step size for the modulator.
9. A delta modulator system is designed to operate at five times the nyquist rate for a
signal with 3KHz bandwidth. Determine the maximum amplitude of a 2KHz input
sinusoid for which the delta modulator does not have slope overload. Quantization
step size is 250mv.Derive the formula used.
10. A signal m (t) is to be encoded using either Delta modulation or PCM technique. The
signal to quantization noise ratio (So/No) ≥ 30dB.Find the ratio bandwidth required
for PCM to Delta modulation.
Tutorial – 3
11. One of four possible messages Q1, Q2, Q3, Q4 having probabilities 1/8, 3/8, 3/8, and
1/8 respectively is transmitted. Calculate average information per message.
12. An ideal channel low pass channel of bandwidth B hz with additive Gaussian white
noise is used for transmitting of digital information.
a. Plot C/B versus S/N in dB for an ideal system using this channel
b. A practical signaling scheme on this channel used one of two waveforms of
duration Tb sec to transmit binary information. The signaling scheme
transmits data at the rare of 2B bits/sec, the probability of error is given by P
(error/1sent) = Pe
c. Plot graphs of
i. C/B
ii. Dt/B where Dt is rate of information transmission over channel.
13. A DMS has an alphabet of eight letters, Xi , i=1,2,3,….,8, with probabilities

0.36,0.14,0.13,0.12,0.1,0.09,0.04,0.02.
a) Use the Huffman encoding procedure to determine a binary code for the source
output.
b) Determine the entropy of the source and find the efficiency of the code

14. A DMS has an alphabet of eight letters, Xi , i=1,2,3,….,8, with probabilities
{0.05,0.1,0.1,0.15,0.05,0.25,0.3}
i. Use the Shannon-fano coding procedure to determine a binary code
for the source output.
ii. Determine the entropy of the source and find the efficiency of the
code. An analog signal band limited to 10HKz quantize is 8levels of
PCM System with
15. Probability of 1/4/1/5, 1/5, 1/10,1/20,1/20, and 1/20 respectively. Find the entropy and
rate of information.
16. Binary data has to be transmitted over a telephone link that has a usable bandwidth of
a. 3000Hz and a maximum achievable SNR of 6dB at its output.

b. Determine the maximum signaling rate and error probability if a coherent
ASK
c. scheme is used for transmitting binary data through this channel.
d. If the data rate is maintained at 300 bits/sec. Calculate the error probability.
Tutorial – 4
17. Consider a (6,3) generator matrix
100011
G = 010101
001110
Find
a) All the code vectors of this code.
b) The parity check matrix for this code.
c) The minimum weight of the code.
18. Draw and explain a decoder diagram for a (7,4) majority logic code whose
a. generator polynomial g(x)=1+x+x3.
b. The generator polynomial of a (7,4) cyclic code is g(x)=1+x+x3.Find the 16
code words of this code in the following ways.
c. By forming the code polynomials using V(x)=D(x)g(x), where D(x) is the

message polynomial by using systematic form.
19. Design an encoder for the (7,4) binary cyclic code generated by g(x)=1+x+x 3 and
verify its operation using the message vector (0101).

20. A (7, 4) linear block code is generated according to the H matrix
1110100
H= 1101010
1011001
a. The code word received is 1000011 for a transmitted codeword C. Find the
b. Corresponding data word transmitted.
21. Consider a (6,3) generator matrix

100011
G = 010101
001110
Find
a. All the code vectors of this code.
b. The parity check matrix for this code.
c. The error syndrome of the code.
22. Draw the state diagram, tree diagram for K=3, rate1/3 code generated by
23. Design an encoder for the (7,4) binary cyclic code generated by g(x) = 1+x+x3 and
verify its operation using the message vector (0101).
Tutorial – 5
24. A convolutional encoder has two shift registers two modulo-2 adders and an output
multiplexer. The generator sequences of the encoder are as follows: g(1)=(1,0,1); g(2)=(
1,1, 1). Assuming a 5bit message sequence is transmitted. Using the state diagram
find the message sequence when the received sequence is
(11,01,00,10,01,10,11,00,00,......)
25. Find the output codeword for the following convolutional encoder for the message
sequence 10011. (as shown in the figure).
26. Construct the state diagram for the following encoder. Starting with all zero state,
trace the path that correspond to the message sequence 1011101. Given convolutional
encoder has a single shift register with two stages,(K=3) three modulo-2 adders and
an output multiplexer. The generator sequence s of the encoder are as follows.
g(1)=(1, 0, 1) ; g(2)=(1, 1, 0),g(3)=(1,1,1).

27. Draw and explain Tree diagram of convolutional encoder shown below with rate=1/3,
L=3
2.
28. For the convolutional encoder shown below draw the trellis diagram for the message
sequence 110.let the first six received bits be 11 01 11 then by using viterbi decoding
find the decoded sequence.
Tutorial – 6
29. Explain the signal space representation of QPSK .Compare QPSK with all other
digital signaling schemes.
30. Explain in detail the power spectra and bandwidth efficiency of M-ary signals.
31. Explain coherent and non-coherent detection of binary FSK waves.
32. Derive an expression for error probability of coherent ASK scheme.
33. Derive an expression for error probability of non-coherent ASK scheme.
34. Find the transfer function of the optimum receiver and calculate the error probability.
35. Derive an expression for probability of bit error of a binary coherent FSK receiver.
36. Derive an expression for probability of bit error in a PSK system.
Tutorial – 7
37. Explain the Direct sequence spread spectrum technique with neat diagram
38. Explain the Frequency hopping spread spectrum in detail.
39. Explain the properties of PN Sequences.
40. How pseudo noise sequence is generated? Explain it with example.
41. How DS-SS works? Explain it with a block diagram. Explain the operation of slow
and fast frequency hopping technique.
42. Explain about source coding of Speech for wireless communication
43. Explain TDMA system with frame structure, frame efficiency and features. Explain
CDMA system with its features and list out various problems in CDMA systems.

18. a) Unit-wise Question Bank
Part-I (2 Marks questions)

Unit-I: Elements of Digital Communication Systems
S. Question Blooms
No Taxonomy
1 Explain the simplified Block diagram of an Electronic communication Understand

system with the help of diagram.
2 List two examples each for analog and digital signals (in mathematical Remember
form).
3 Explain Shannon & Hartley’s Law. Evaluate
4 Construct the equation for Shannon limit on Information capacity. Create
5 Explain about trade-off between bandwidth and SNR in a Evaluate

Communication signal.
6 Define bandwidth. Remember
7 Define Nyquist Sampling theorem. Remember
8 Construct the mathematical expression for Minimum sampling rate (fs). Create
9 Examine Aliasing Effect (or) Fold-over distortion? How it can be Analyze

removed.
10 List the advantages of digital communication systems Analyze

11 What is a matched filter? Remember
12 List two applications for eye pattern. Analyze
Discuss the performance of data transmission system using eye
13 Create
pattern technique?
Discuss the need of optimum transmitting and receiving filter in Create

14
baseband data transmission.
What is the value of maximum signal to noise ratio of the matched Remember
15
filter? When it becomes maximum?
16 Construct the block diagram of Base band System. Create
17 Examine Crosstalk. Analyze
18 Explain Optimum Receiver. Understand
19 Explain Signal Space Representation. Understand
20 What does the width of the eye define? Remember
Make use of the eye pattern and how the sensitivity on the system Apply
21 can be determined?
UNIT-I Pulse Code Modulation
Blooms
S. No Question Taxonomy
Summarize differential encoding signaling? Explain with an
1 Understand
example.
2 Define quantization in PCM. Remember
3 Explain a simple model of non uniform quantizer. Understand
4 Define the term quantization noise. Remember
5 Compare the features of PCM and DPCM. Analyze
6 List the advantage gained by the use of robust quantization. Remember
7 Define an output signal-to-quantization ratio. Remember
Mention two major sources of noise which influence the
8 Knowledge
performance of a PCM system.
9 Discuss the advantages of DM over PCM. Create
10 Construct the block diagram of pulse code modulation. Apply
11 Define quantization noise power Remember
12 Discuss about uniform quantization? Create
13 Discuss about Quantization? Create
14 Compare uniform and non-uniform quantization Analyze
Unit-II Information Theory
Blooms
1 What is meant by distortion less transmission? Remember
2 Discuss entropy and give the expression for it. Creating
3 Explain the channel capacity theorem. Understand
Let X represents the outcome of a single roll of a fair die. What is
5 Remember
the entropy of X?

6 What is transition probability and when it does it will occur? Remember
7 Explain the two properties of Mutual information. Understand
8 State the properties of Entropy Creating
What is discrete memory less channel and give the channel matrix
9 Understand
expression
What is channel coding theorem and how it is different from
10 Understand
source coding theorem?
What is entropy? Show that the entropy is maximum when all the
11 Understand
symbols are equi probable. Assume M=2.
Define information. Show that information contained by a symbol
12 is inversely proportional to the probability of that symbol. Create
Unit-III Linear Block Codes & Cyclic Codes
Blooms
S. No Question
Taxonomy
1 What is linear code? Remember
2 Discuss code rate? Create
3 Define code efficiency. Remember
4 Explain hamming distance? Understand
5 What is meant by systematic & non-systematic code? Remember
Explain how syndrome is calculated in Hamming codes and cyclic Evaluate
6 codes?
7 What are the conditions to satisfy the hamming code? Remember
8 Define code word & block length. Remember
9 What are the advantages of cyclic codes? Remember
10 What is linear code? Remember
What is constraint length of convolution code.

Understand
11
12 List advantages of convolutional codes Understand

Discuss the difference between convolutional code and block
13 Create
code.
Construct the graphical representations of convolutional codes.
14 Create
Construct the encoding diagram for (3, 2, 1) convolutional

15 Apply
encode.
What is sequential decoding? Understand
16
Explain about the Convolutional interleaving.

17 Evaluate
18 Compare coded and uncoded transmission techniques with respect to Evaluate

Probability of error.
What is the code length of a convolution code?
19 Understand
Examine the time-domain approach in convolution code.

20 Analyze
What is the importance of code tree? Understand
21
Define the term trellis in convolution code. Remember
22
Explain Viterbi algorithm. Create

23
Discuss maximum-likelihood decoding rule for the binary Create

24
Symmetric channel.
Unit-IV Digital Modulation Techniques
Blooms
S. No Question
Taxonomy
1 Construct the ASK and FSK waveforms for 011011. Apply
2 Remember
Sketch the block diagram of ASK generation.
Examine how does pulse shaping reduce inter symbol
3 Analyze
interference?
4 Show the space representation of BPSK and QPSK Remember
5 Explain the Bandwidth, power and energy calculations for PSK Evaluate
signal.
6 Explain why PSK is always preferable over ASK in coherent Evaluate
detection?

7 Distinguish between Coherent and Non coherent detection? Analyze
8 Understand
Explain Phase shift keying with relevant equations and waveforms.
9 Estimate the band width required for frequency shift keying and Create
draw its spectrum.
10 Explain non coherent detection of Amplitude shift keying. Understand
Construct the constellation diagram for Quadrature phase shift

11
keying. Apply
Explain coherent detection of frequency shift keying .what should

12 Evaluate
be the relationship between bit rate and frequency shift for a better
performance?
13 Construct the FSK waveforms for a given input data “1101”. Apply
14 Define the probability of error. Remember

Unit-V Spread Spectrum Modulation
Blooms
S. No Question
Taxonomy
1 Define spread spectrum communication Remember
2 Explains pseudo noise sequence? Understand
3 Discuss direct sequence spread spectrum modulation Create
4 What is frequency hap spread spectrum modulation? Remember
5 What is processing gain? Understand
6 State four applications of spread spectrum. Create
7 When is the PN sequence called as maximal length sequence? Anal yze
8 What is meant by processing gain of DS spread spectrum system? Understand
9 Discuss the applications of spread spectrum modulation? Create
10 Define frequency hopping. Understand
11 What are the Advantages of DS-SS systems? Remember
12 What are the Disadvantages of DS-SS systems? Remember
13 List the Advantages of FH-SS System Analyze
14 List the Disadvantages of FH-SS System Analyze
Part-II (10 Marks questions)

Unit-I: Elements of Digital Communication Systems
S. No Blooms
Question Taxonomy
1 Explain the different types of Sampling. Understand
Distinguish between natural sampling and flat top sampling with
2 neat schematics, listing out their merits and demerits. Analyze
Explain the principle of working a sample and hold circuit. List

3 out its applications with neat diagrams. Evaluate
Define the sampling theorem as applicable to voice signals on
4 Remember
telephone lines.
Discuss the advantages and disadvantages of digital
5 communication system. Create
6 Discuss and prove sampling theorem in time domain. Create
7 Define natural sampling? Explain it with sketches. Remember
8 Discuss the Hartley-Shannon law. Create

Explain the Model of Digital Communication Systems with neat
9 diagrams. Understand
10 Explain Bandwidth-S/N Tradeoff Understand
11 Explain A Baseband Signal Receiver. Understand

12 Explain Optimum Receiver Evaluate
13 Explain Optimal of Coherent Reception Evaluate
UNIT-I Pulse Code Modulation

Blooms
Illustrate the working of DPCM transmitter and receiver with
1 Understand
the help of diagram.
2 Enumerate the quantization error in delta modulation. Application
3 List the comparison between PCM and DM systems. Remember
4 Elaborate how to avoid slope overload distortion in DM. Create

Illustrate the working of Adaptive DPCM with the help of
5 Understand
diagram.
Illustrate the working of Adaptive DM with the help of
6 Understand
diagram.
Evaluate
7 Explain the Companding.
Explain
a)Channel Noise Evaluate
8 b) Quantization noise in DM and derive expression for them?
Explain the need for non-uniform quantization in digital
9 Comprehension
communications.
10 Explain the Block diagram of DPCM system. Understand
Discuss quantization error? How does it depend upon the step
11 size? Suggest some methods to overcome the difficulties Create
encountered depending on the modulating Amplitude
swing?
Unit-II Information Theory
Blooms
Show that the entropy for a discrete source is a maximum when
1 Understand
the output symbols are equally probable.
Show that the mutual information of a channel is related to the
2 Understand
joint entropy of the channel input and channel output.
3 Explain Shannon-fano coding algorithm using an example. Creating
5 Explain the Huffman coding algorithm using an example. Analyse
6 Explain the Conditional Entropy. Apply
7 Explain the Redundancy. Evaluate
8 Explain the Mutual Information Creating
9 Explain the Variable length Coding with an example Analyse
Creating
10 Explain the Lossy Source Coding
& Analyse
Unit-III Linear Block Codes & Cyclic Codes

Blooms
Creating &
1 Explain syndrome decoding for cyclic code expression.
analyze

2 What is parity check matrix and how it is used? Remember
C eating &
3 Explain systematic cyclic code generation formula
analyze
4 What are minimum distance considerations? Remember
Show that the syndrome depends only on the error pattern, and n
t
5 Remember
on the transmitted code word.
Show that the minimum distance of a linear block code is equal to
6 Remember
the minimum number of rows of HT that sum to zero.
7 Analyze and prove the fundamental properties of cyclic code. Analyze
Show that if ci and cj are two code vectors in an (n,k) linear block
8 Remember
code, then their sum is also a code vector.
The generator polynomial of (15,11) cyclic code is g(x)= 1+x+x4.
9 Evaluate
Determine the parity polynomial h(x) of this code.
Unit-III Convolutional Code
Blooms
S. No Question
Taxonomy
Compare the linear block codes, cyclic codes and the
1 convolutional codes? Evaluate
Draw an (n-k) syndrome calculation circuit for an (n, k)cyclic code?
2 Understand
What is meant by random errors and burst errors? Explain about a
coding technique which can be used to correct both the burst and Remember
3 random errors simultaneously.
Discuss about the various decoders for convolutional codes. Create
4
Explain how the channel coding reduces the probability of error Evaluate
5
Evaluate
6 Explain the systematic code form for the binary cyclic codes?
Explain about block codes in which each block of k message bits encoded Evaluate
7 into block of n>k bits with an example.
Demonstrate the Viterbi algorithm for maximum-likelihood decoding of Create
8 convolutional codes.
Remember
9 What is a convolutional code? How is it different from a block code?
Unit-IV Digital Modulation Techniques

Blooms
Explain in detail about
i. i)FSK
1 ii. ii)PSK with waveforms and equations Understand

Determine probability of error for
a) ASK and
2 b) PSK systems. Create
a) Explain the demodulation of FSK using coherent detection.
3 Create
b) Draw the block diagram of QPSK receiver.
4 Explain the generation of PSK signals. Understand

a) Discuss QPSK signaling.
5 Create
b) Derive the bit error probability due to PSK receiver.
Solve that the maximum output signal to noise ratio of a matched
6 Create
filter is (SNR) = 2E/N0
Explain Differential phase shift keying modulation with neat
7 Create
block diagram. Draw the wave forms.
8 Show that the probability of error for phase shift keying is Pe= Q Remember
(2Sav Tb/ N0)1/2 and the threshold levelis zero.
The bit stream 11011100101 is to be transmitted using DPSK.
9 Determine the encoded sequence and the transmitted phase Evaluate
sequence.
10 Explain the working of DPSK modulator and demodulator.
Unit-V Spread Spectrum Modulation
S. No Question Blooms Taxonomy
1 Explain the spread spectrum modulation and demodulation. Create
2 Explain the frequency hopping spread spectrum modulation. Evaluate
3 Examine spread spectrum modulation using DSSS. Analyze
4 Explain the spread spectrum modulation. Understand
5 Explain the frequency hopping spread spectrum modulation. Evaluate
a) Explain how PN sequences are generated. that are maximal-length

sequences? What are their properties and why are they preferred?
b) With the help of a neat block diagram, explain the working of a DS

6 spread spectrum based CDMA system. Understand
a)What are the advantages of spread spectrum technique.
7 b) Compare direct sequence spread spectrum and frequency Remember
Hopped spread spectrum techniques and draw the important
a) What the PN sequences? Discuss the characteristics.
8 b) What are the two basic types of spread-spectrums systems? Analyze
9 Explain the spread spectrum modulation. Understand
10 Explain the frequency hopping spread spectrum modulation. Understand

18 b) Unit wise Quiz Questions and long answer questions
CHOOSE THE CORRECT ANSWER
1.A source is transmitting six messages with probabilities,1/2,1/4,1/8,1/16,1/32,and1/32.Then
(a) Source coding improves the error performance of the communication system.
(b) Channel coding will reduce the average source code word length.
(c) Two different source codeword sets can be obtained using Huffman coding.
(d) Two different source codeword sets can be obtained using Shanon-Fano coding
2. A memory less source emits 2000binarysymbols/sec and each symbol has a Probability of
0.25 to be equal to 1and 0.75 to be equal to 0.The minimum number of bits/sec required for
error free transmission of this source is
(a) 1500
(b) 1734
(c) 1885
(d) 162213.
3. A system has a bandwidth of 3 KHz and an S/N ratio of 29dB at the input of the receiver .If
the bandwidth of the channel gets doubled, then
(a) Its capacity gets doubled

(b) Its capacity gets halved
(c) The corresponding S/N ratio gets doubled
(d) The corresponding S/N ratio gets halved
4. The capacity of a channel with infinite bandwidth is
(a) finite because of increase in noise power

(b) finite because of finite message word length
(c) Infinite because of infinite noise power
(d) Infinite because of infinite bandwidth
5. Which of the following is correct?
(a) Channel coding is an efficient way of representing the output of a source

(b) ARQ scheme of error control is applied after the receiver makes a decision about the
received bit
(c)ARQ scheme of error control is applied when the receiver is unable to make a decision
about the received bit.
(d) Source coding introduces redundancy

(a) Source encoding reduces the probability of transmission errors
(b) In an (n, k) systematic cyclic code, the sum of two code words is another codeword of the
code.

(c) In a convolution encoder, the constraint length of the encoder is equal to the tail of the
message sequence+ 1.
(d) In an (n ,k) blockcode, each codeword is the cyclic shift of an another codeword of the
code.
7. Automatic Repeat Request is a
(a) error correction scheme

(b) Source coding scheme
(c) Error control scheme
(d) Data conversion scheme
8. The fundamental limit on the average number of bits/source symbolis
(a) Channel capacity

(b) Entropy of the source
(c) Mutual Information
(d) Information content of the message
9. The Memory length of a convolutional encoder is 5. If a 6 bit message sequence is applied

as the input for the encoder, then for the last message bit to come out of the encoder, the
number of extra zeros to be applied to the encoder is
(a) 6
(b) 4
(c) 3
(d) 5
Answers
1.C 2.D 3.B 4.A 5.C 6.C 7.C 8.B 9.A 10.D
1. The cascade of two Binary Symmetric Channels is a
(a) Symmetric Binary channel
(b) Asymmetric Binary channel
(c) Asymmetric quaternary channel
(d) Symmetric quaternary channel
(a) Source coding introduces redundancy
(b) ARQ scheme of error control is applied after the receiver makes a decision about the
received bit
(c) Channel coding is an efficient way of representing the output of a source
(d) ARQ scheme of error control is applied when the receiver is unable to make a decision

about the received bit.
3. A linear block code with Hamming distance 5 is
(a) Triple error correcting code
(b) Single error correcting and double error detecting code
(c) Double error detecting code
(d) Double error correcting code
4. In a Linear Block code
(a) the encoder satisfies superposition principle
(b) the communication channel is a linear system
(c) parity bits of the code word are the linear combination of the message bits
(d) the received power varies linearly with that of the transmitted power
5. The fundamental limit on the average number of bits/source symbol is
(a) Channel capacity
(b) Information content of the message
(c) Mutual Information
(d) Entropy of the source
6. Which of the following involves the effect of the communication channel?
(a) Entropy of the source
(b) Information content of a message
(c) Mutual information
(d) Information rate of the source
7. Whichofthefollowingprovidesthefacilitytorecognizetheerroratthereceiver?
(a) Shannon -Fano Encoding
(b) differential encoding
(c) Parity Check codes
(d) Huffman encoding
8. A system has a bandwidth of 3 KHz and an S/N ratio of 29dB at the input of the receiver. If
the bandwidth of the channel gets doubled, then
(a) the corresponding S/N ratio gets doubled

(b) its capacity gets doubled
(c) its capacity gets halved
(d) the corresponding S/N ratio gets halved
9. Information rate of a source can be used to
(a) design the matched filter for the receiver
(b) differentiate between two sources
(c) Correct the errors at the receiving side
(d) to find the entropy in bits/message of a source
10. In a communication system, the average amount of uncertainty associated with the
Source, sink, source and sink jointly in bits/message are1.0613,1.5 and2.432 respectively.
Then the information transferred by the channel connecting the source and sink in bit sis
(a) 1.945
(b) 4.9933
(c) 2.8707
(d) 0.1293
11.ABS Chasa transition probability of P. The cascade of two such channel sis
(a) asymmetric channel with transition probability2P (1-P)
(b) an asymmetric channel with transition probability2P
(c) an asymmetric channel with transition probability(1-P)
(d) asymmetric channel with transition probability P2s.
Answers
1.A 2.D 3.D 4.C 5.D 6. C 7. A 8.C 9.D 10.D 11.A
1. Information rate of a source is
(a) maximum when the source is continuous
(b) the entropy of the source measured in bits/message
(c) a measure of the uncertainity of the communication system
(d) the entropy of the source measured in bits/sec.
2. A Field is
(a) a group with 0 as the multiplicative identity for its members

(b) a group with 0 as the additive inverse for its members
(c) a group with 1 as the additive identity for its members
(d) an Abelian group under addition
3.Under error free reception, the syndrome vector computed for the received cyclic code
word consists of
(a) all ones
(b) alternate0‘s and1‘s starting with a 0
(c) all zeros
(d) alternate 1‘s and 0‘sstarting with a 1
4. The Memory length of a convolutional encoder is 3. If a 5 bit message sequence is applied

as the input for the encoder, then forth last message bit to come out of the encoder. The
(a) 5
(b) 4
(c) 3
(d) 6
5. The cascade of two binary symmetric channels is a
(A) Symmetric binary channel
(B) Symmetric quaternary channel
(C) Asymmetric quaternary channel
(D) Asymmetric binary channel
6. There are four binary words given as 0000, 0001,0011, 0111.Which of these cannot beam
member of the parity check matrix of a(15,11 ) linear Block code?
(a) 0011
(b) 0000,0001
(c) 0000
(d) 0111
7. The encoder of a (7,4) systematic cyclic encoder with generating polynomial g(x)=1+x2
+x3 is basically a
(a) 3 stage shift register
(b) 22 stage shift register

(c) 11 stage shift register
(d) 4 stage shift register
8. A source X with entropy 2 bits/message is connected to the receiver Y through a Noise

free channel. The conditional probability of the source, given the receiver is H(X/Y) and the
joint entropy of the source and the receiver H(X,Y) .Then
(a) H(X/Y)=2 bits/message
(b) H(X,Y)=2bits/message
(c) H(X/Y)=1bit/message
(d) H(X,Y)=0bits/message
9. A system has a bandwidth of 4 KHz and an S/N ratio of 28 at the input to the Receiver. If
the bandwidth of the channel is doubled , then
(a) S/N ratio at the input of the received gets halved
(b) Capacity of the channel gets doubled
(c) Capacity of the channel gets squared
(d) S/N ratio at the input of the received gets doubled
10. The Parity Check Matrix of a(6,3) Systematic Linear Block code is
a)101100
b) 011010
c)110001
11.If the Syndrome vector computed for the received code word is [010], then for error
correction, which of the bits of the received code word is to be complemented?
(a)3
(b) 4
(c) 5
(d) 2
Answers
1.D 2.D 3.C 4.B 5.A 6.C 7.A 8.B 9.A 10.C
1. The minimum number of bits per message required to encode the output of source
transmitting four different messages with probabilities 0.5,0.25,0.125and0.125is
(a) 1.5 (b) 1.75 (c) 2 (d) 1

2. A Binary Erasure channel has P(0/0)=P(1/1)=p; P(k/0)=P(k/1)=q.Its Capacity in
bits/symbol is
(a) p/q (b) p q (c) p (d) q
3. The syndromes(x) o f a cyclic code is given by Reminder of the division [{V(x)+E(x)}

/g(x)], where V (x) is the transmitted code polynomial(x)is the error polynomial and g(x)
is the generator polynomial. The S(x) is also equal to
(a) Reminder of [V(x).E(x)]/g(x)
(b) Reminder of V(x)/g(x)
(c) Reminder of E(x)/g(x)
(d) Remaindering(x)/V(x)
5. The output of a continuous source is a uniform random variable in the range 0 ≤ x ≤ 4.

The entropy of the source in bits/samples
(a) 2 (b) 8 (c) 4 (d) 1
6. In a (6,3) systematic Linear Block code, the number of‘6‘bit code words that are not
useful is
(a) 45 (b) 64 (c) 8 (d) 56
7. The output of a source is band limited to 6KHz.It is sampled at a rate of 2KHz above the
nyquist rate. If the Entropy of the source is 2bits/sample, then the entropy of the source in
bits/sec is
(a) 12Kbps (b) 32Kbps (c) 28Kbps (d) 24Kbps
8. The channel capacity of a BSC with transition probability ½ is
(a) 2bits (b) 0bits (c) 1bit (d) infinity
9. A communication channel is fed with an input signal x(t) and the noise in the channel is
negative. The Power received at the receiver input is
(a) Signal power-Noise power
(b) Signal power +Noise Power
(c) Signal power x Noise Power
(d) Signal power /Noise power
10. White noise of PSD η/2 is applied to an ideal LPF with one sided bandwidth of BHz.
The filter provides again of 2. If the output power of the filter is 8η, then the value of B in Hz
is
(a) 8 (b) 2 (c) 6 (d) 4

Answers
1.B 2.C 3.C 4.A 5.D 6.C 7.B 8.B 9.B 10.A
(a) The syndrome of a received Block coded word depends on the received codeword
(b) The syndrome for a received Block coded word under error free reception consists of all
1‘s.
(c) The syndrome of a received Block coded word depends on the transmitted codeword.
(d) The syndrome of a received Block coded word depends on the error pattern
2. A Field is
(a) a group with 0 as the multiplicative identity for its members
(b) a group with 0 as the additive inverse for its members
(c) a group with 1 as the additive identity for its members
(d) an abelian group under addition
3. Variable length source coding provides better coding efficiency, if all the messages of the
source are
(a) Equi probable
(b) continuously transmitted
(c) discretely transmitted
(d) with different transmission probability
(a) FEC and ARQ are not used for error correction
(b) ARQ is used for error control after receiver makes a decision about the received bit
(c) FEC is used for error control when the receiver is unable to make a decision about the
received bit
(d) FEC is used for error control after receiver makes a decision about the received bit
5. The source coding efficiency can be increased by
(a) using source extension
(b) decreasing the entropy of the source
(c) using binary coding

(d) increasing the entropy of the source
6. a discrete source X is transmitting m messages and is connected to the receiver Y

through asymmetric channel. The capacity of the channel is given as
(a) log m bits/symbol
(b) H(X)+H(Y)-H(X,Y)bits/symbol
(c) log m-H(X/Y)bits/symbol
(d) log m-H(Y/X)bits/symbol
7. The time domain behavior of a convolutional encoder of code rate 1/3 is defined in terms
of a set of
(a) 3 ramp responses
(b) 3 step responses
(c) 3 sinusoidal responses
(d) 3 impulse responses
8. A source X with entropy 2bits/message is connected to the receiver Y through a Noise free
channel. The conditional probability of the source, given the receiver is H(X/Y) and the joint
entropy of the source and the receiver H(X, Y).Then
(a) H(X,Y)=2 bits/message
(b) H(X/Y)=1 bit/message
(c) H(X,Y)=0 bits/message
(d) H(X/Y)=2 bits/message
9. The fundamental limit on the average number of bits/source symbol is
(a) Mutual Information
(b) Entropy of the source
(c) Information content of the message
(d) Channel capacity
10. The Memory length of a convolutional encoder is 5. If a 6bit message sequence is applied
as the input for the Encoder ,then for the last message bit to come out of the encoder, the
(a) 4 (b) 6 (c) 3 (d) 5
Answers
1.D 2.D 3.D 4.D 5.A 6.D 7.D 8.A 9.B 10.B

1. If ‘a‘ is an element of a Field ‘F ‘, then its additive inverse is
(a) –a (b) 0 (c) a (d) 1
2. Relative to Hard decision decoding, soft decision decoding results in
(a) better coding gain
(b) lesser coding gain
(c) less circuit complexity
(d) better bit error probability
3. . Under error free reception, the syndrome vector computed for the received cyclic
codeword consists of
(a) alternate 0‘s and1‘s starting with a 0
(b) all zeros
(c) all ones
(d) alternate 1‘s and 0‘s starting with a 1
4. Error free communication may be possible by
(a) increasing transmission power to the required level
(b) providing redundancy during transmission
(c) increasing the channel bandwidth
(d) reducing redundancy during transmission
5.A discrete source X is transmitting m messages and is connected to the receiver Y through
asymmetric channel. The capacity of the channel is given as
(a) H(X)+H(Y)-H(X,Y)bits/symbol
(b) log m-H(X/Y)bits/symbol
(c) log m-H(Y/X)bits/symbol
(d) log m bits/symbol
6. The encoder of a (7,4) systematic cyclic encoder with generating polynomial g(x)=1+x2
+x3 is basically a
(a) 11 stage shift register
(b) 4 stage shift register

(c) 3 stage shift register
(d) 22 stage shift register
7. A channel within dependent input and output acts as
(a) Gaussian channel
(b) channel with maximum capacity
(c) lossless network
(d) resistive network
8.A system has a bandwidth of 4KHz and an S/N ratio of 28 at the input to the Receiver. If
the bandwidth of the channel is doubled, then
(a) S/N ratio at the input of the received gets halved
(b) Capacity of the channel gets doubled
(c) S/N ratio at the input of the received gets doubled
(d) Capacity of the channel gets squared
9.A source is transmitting four messages with equal probability. Then, for optimum Source
coding efficiency ,
(a) necessarily, variable length coding schemes should be used
(b) Variable length coding schemes need not necessarily be used
(c) Convolutional codes should be used
(d) Fixed length coding schemes should not be used
10.The maximum average amount of information content measured in bits/sec associated

with the output of a discrete information source transmitting 8 messages and
2000messages/sec is
(a) 16Kbps (b) 4Kbps (c) 3Kbps (d) 6Kbps
Answers
1.A 2.A 3.B 4.B 5.C 6.C 7.D 8.A 9.B10. D
19. Detailed Notes

UNIT I :
Elements Of Digital Communication Systems

 Model of digital communication system,
 Digital representation of analog signal,
 Certain issues of digital transmission,
 advantages of digital communication systems,
 Sampling theorem
What Does Communication (or Telecommunication) Mean?

The term communication (or telecommunication) means the transfer of some form of
information from one place (known as the source of information) to another place
(known as the destination of information) using some system to do this function (known
as a communication system).
So What Will we Study in This Course?
In this course, we will study the basic methods that are used for communication in
today’s world and the different systems that implement these communication methods.
Upon the successful completion of this course, you should be able to identify the
different communication techniques, know the advantages and disadvantages of each
technique, and show the basic construction of the systems that implement these
communication techniques.
Old Methods of Communication
 Pigeons
 Horseback
 Smoke
 Fire
 Post Office
 Drums
 … etc
Problems with Old Communication Methods
 Slow
 Difficult and relatively expensive
 Limited amount of information can be sent
 Some methods can be used at specific times of the day
 Information is not secure.
Examples of Today’s Communication Methods
All of the following are electric (or electromagnetic) communication systems
 Satellite (Telephone, TV, Radio, Internet, … )

 Microwave (Telephone, TV, Data, …)
 Optical Fibers (TV, Internet, Telephone, … )
 Copper Cables (telephone lines, coaxial cables, twisted pairs, … etc)
Advantages of Today’s Communication Systems
 Fast
 Easy to use and very cheap
 Huge amounts of information can be transmitted
 Secure transmission of information can easily be achieved
 Can be used 24 hours a day.
Basic Construction of Electrical Communication System

A communication system may transmit information in one direction such as TV and
radio (simplex), two directions but at different times such as the CB (half-duplex), or two
directions simultaneously such as the telephone (full-duplex).
Basic Terminology Used in this Communications Course
A Signal: is a function that specifies how a specific variable changes versus an

independent variable such as time, location, height (examples: the age of
people versus their coordinates on Earth, the amount of money in your
bank account versus time).
A System: operates on an input signal in a predefined way to generate an output

signal.
Analog Signals: are signals with amplitudes that may take any real value out of an infinite
number of values in a specific range (examples: the height of mercury in
a 10cm–long thermometer over a period of time is a function of time that
may take any value between 0 and 10cm, the weight of people setting in a
class room is a function of space (x and y coordinates) that may take any
real value between 30 kg to 200 kg (typically)).
Digital Signals: are signals with amplitudes that may take only a specific number of
values (number of possible values is less than infinite) (examples: the
number of days in a year versus the year is a function that takes one of
two values of 365 or 366 days, number of people sitting on a one-person

chair at any instant of time is either 0 or 1, the number of students
registered in different classes at KFUPM is an integer number between 1
and 100).
Noise: is an undesired signal that gets added to (or sometimes multiplied with) a
desired transmitted signal at the receiver. The source of noise may be
external to the communication system (noise resulting from electric
machines, other communication systems, and noise from outer space) or
internal to the communication system (noise resulting from the collision
of electrons with atoms in wires and ICs).
Signal to Noise Ratio (SNR):is the ratio of the power of the desired signal to the power of
the noise signal.
Bandwidth (BW): is the width of the frequency range that the signal occupies. For example
the bandwidth of a radio channel in the AM is around 10 kHz and the
bandwidth of a radio channel in the FM band is 150 kHz.
Rate of Communication: is the speed at which DIGITAL information is transmitted. The

maximum rate at which most of today’s modems receive digital
information is around 56 k bits/second and transmit digital information is
around 33 k bits/second. A Local Area Network (LAN) can theoretically
receive/transmit information at a rate of 100 M bits/s. Gigabit networks
would be able to receive/transmit information at least 10 times that rate.
Modulation: is changing one or more of the characteristics of a signal (known as the

carrier signal) based on the value of another signal (known as the
information or modulating signal) to produce a modulated signal.
Analog and Digital Communications
Since the introduction of digital communication few decades ago, it has been gaining
a steady increase in use. Today, you can find a digital form of almost all types of analog
communication systems. For example, TV channels are now broadcasted in digital form
(most if not all Ku–band satellite TV transmission is digital). Also, radio now is being
broadcasted in digital form (see sirus.com and xm.com). Home phone systems are starting to
go digital (a digital phone system is available at KFUPM). Almost all cellular phones are now
digital, and so on. So, what makes digital communication more attractive compared to analog
communication?
Advantages of Digital Communication over Analog Communication
 Immunity to Noise (possibility of regenerating the original digital signal if signal

power to noise power ratio (SNR) is relatively high by using of devices called
repeaters along the path of transmission).
 Efficient use of communication bandwidth (through use of techniques like
compression).
 Digital communication provides higher security (data encryption).
 The ability to detect errors and correct them if necessary.

 Design and manufacturing of electronics for digital communication systems is
much easier and much cheaper than the design and manufacturing of electronics
for analog communication systems.
Modulation
Famous Types
 Amplitude Modulation (AM): varying the amplitude of the carrier based on the
information signal as done for radio channels that
are transmitted in the AM radio band.
 Phase Modulation (PM): varying the phase of the carrier based on the
information signal.
 Frequency Modulation (FM): varying the frequency of the carrier based on the
information signal as done for channels transmitted
in the FM radio band.
Purpose of Modulation
 For a signal (like the electric signals coming out of a microphone) to be

transmitted by an antenna, signal wavelength has to be comparable to the length of
the antenna (signal wavelength is equal to 0.1 of the antenna length or more). If
the wavelength is extremely long, modulation must be used to reduce the
wavelength of the signal to make the length of the required antenna practical.
 To receive transmitted signals from multiple sources without interference between

them, they must be transmitted at different frequencies (frequency multiplexing)
by modulating carriers that have different frequencies with the different
information signals.
Exercise 1–1: Specify if the following communication systems are (A)nalog or (D)igital:
a) TV in the 1970s:
b) TV in the 2030s:
c) Fax machines
d) Local area networks (LANs):
e) First–generation cellular phones
f) Second–generation cellular phones
g) Third–generation cellular phones

These are the basic elements of any digital communication system and It gives a basic
understanding of communication systems.
1. Information Source and Input Transducer:

The source of information can be analog or digital, e.g. analog: audio or video
signal, digital: like teletype signal. In digital communication the signal produced by
this source is converted into digital signal consists of 1′s and 0′s. For this we need
source encoder.
2. Source Encoder
In digital communication we convert the signal from source into digital signal
as mentioned above. The point to remember is we should like to use as few binary
digits as possible to represent the signal. In such a way this efficient representation of
the source output results in little or no redundancy. This sequence of binary digits is
called information sequence.
Source Encoding or Data Compression: the process of efficiently converting the
output of wither analog or digital source into a sequence of binary digits is known as
source encoding.
3. Channel Encoder:
The information sequence is passed through the channel encoder. The purpose of
the channel encoder is to introduced, in controlled manner, some redundancy in the
binary information sequence that can be used at the receiver to overcome the effects

of noise and interference encountered in the transmission on the signal through the
channel.
e.g. take k bits of the information sequence and map that k bits to unique n bit
sequence called code word. The amount of redundancy introduced is measured by the
ratio n/k and the reciprocal of this ratio (k/n) is known as rate of code or code rate.
4. Digital Modulator:
The binary sequence is passed to digital modulator which in turns convert the
sequence into electric signals so that we can transmit them on channel (we will see
channel later). The digital modulator maps the binary sequences into signal wave
forms , for example if we represent 1 by sin x and 0 by Cos(x) then we will transmit
sin x for 1 and Cos(x) for 0. ( a case similar to BPSK)
5. Channel:
The communication channel is the physical medium that is used for
transmitting signals from transmitter to receiver. In wireless system, this channel
consists of atmosphere , for traditional telephony, this channel is wired , there are
optical channels, under water acoustic channels etc. we further discriminate this
channels on the basis of their property and characteristics, like AWGN channel etc.
6. Digital Demodulator:
The digital demodulator processes the channel corrupted transmitted waveform and
reduces the waveform to the sequence of numbers that represents estimates of the
transmitted data symbols.
7. Channel Decoder:
This sequence of numbers then passed through the channel decoder which attempts to
reconstruct the original information sequence from the knowledge of the code used by the
channel encoder and the redundancy contained in the received data
The average probability of a bit error at the output of the decoder is a measure of the
performance of the demodulator – decoder combination. THIS IS THE MOST
IMPORTANT POINT, We will discuss a lot about this BER (Bit Error Rate) stuff in
coming posts.
8. Source Decoder
At the end, if an analog signal is desired then source decoder tries to decode the
sequence from the knowledge of the encoding algorithm. And which results in the
approximate replica of the input at the transmitter end
9. Output Transducer:
Finally we get the desired signal in desired format analog or digital.
The points worth noting are :
1. the source coding algorithm plays important role in higher code rate
2. the channel encoder introduced redundancy in data
3. the modulation scheme plays important role in deciding the data rate and immunity of
signal towards the errors introduced by the channel
4. channel introduced many types of errors like multi path, errors due to thermal noise

etc.
5. the demodulator and decoder should provide high BER.
Some more explanation of advantages and disadvantages of analog vs digital

communications.
1.The first advantage of digital communication against analog is it’s noise immunity. In any
transmission path some unwanted voltage or noise is always present which cannot be
eliminated fully. When signal is transmitted this noise gets added to the original signal
causing the distortion of the signal. However in a digital communication at the receiving end
this additive noise can be eliminated to great extent easily resulting in better recovery of
actual signal. In case of analog communication it’s difficult to remove the noise once added
to the signal.
2.Security is another priority of messaging services in modern days. Digital communication

provides better security to messages than the analog communication. It can be achieved
through various coding techniques available in digital communication.
3. In a digital communication the signal is digitized to a stream of 0 s and 1 s.So at the

receiver side a simple decision has to me made whether received signal is a 0 or a
1.Accordingly the receiver circuit becomes simpler as compared to the analog receiver
circuit.
4. Signal when travelling through it’s transmission path gets faded gradually. So on it’s path it
needs to be reconstructed to it’s actual form and re-transmitted many times. For that reason
AMPLIFIERS are used for analog communication and REPEATERS are used in digital
communication. Amplifiers are needed every 2 to 3 Kms apart where as repeaters are neede
every 5 to 6 Kms apart. So definitely digital communication is cheaper. Amplifiers also often
add non-linearity’s that distort the actual signal.
5. Bandwidth is another scarce resource. Various Digital communication

techniques are available that use the available bandwidth much efficiently than analog
communication techniques.
6. when audio and video signals are transmitted digitally an AD(Analog to Digital) converter
is needed at transmitting side and a DA(Digital to Analog) converter is again needed at
receiver side. While transmitted in analog communication these devices are not needed.
7. Digital signals are often an approximation of the analog data (like voice
or video) that is obtained through a process called quantization. The digital representation is
never the exact signal but it’s most closely approximated digital form. So it’s accuracy
depends on the degree of approximation taken in quantization process.

Sampling Theorem:
There are 3 cases of sampling:
Ideal impulse sampling

Consider an arbitrary lowpass signal x (t ) shown in Fig. 6.2(a). Let

Base Band Transmission And Optimal Digital Signal
 pulse shaping for optimum transmission,
 A Base band signal receiver,
 Different pulses and power spectrum densities,
 Probability of error, optimum receiver,
 Optimum of coherent reception,
 Signal space representation and probability of error,
 Eye diagram,
 cross talk.
BASEBAND FORMATTING TECHNIQUES
CORRELATIVE LEVEL CODING:
 Correlative-level coding (partial response signaling)

– adding ISI to the transmitted signal in a controlled manner
 Since ISI introduced into the transmitted signal is known, its effect can be interpreted at
the receiver
 A practical method of achieving the theoretical maximum signaling rate of 2W symbol
per second in a bandwidth of W Hertz
 Using realizable and perturbation-tolerant filters
Duo-binary Signaling :
Duo : doubling of the transmission capacity of a straight binary system
 Binary input sequence {bk} : uncorrelated binary symbol 1, 0
1 if symbol bk is 1 ck  ak  ak 1
ak  
1 if symbol bk is 0

H I ( f )  H Nyquist ( f )[1  exp( j 2fTb )]
 H Nyquist ( f )[exp( jfTb )  exp( jfTb )] exp( jfTb )
 2 H Nyquist ( f ) cos(fTb ) exp( jfTb )
1, | f | 1 / 2Tb
H Nyquist ( f )  
0, otherwise
 2 cos( fTb ) exp(  j fTb ), | f | 1/ 2Tb

HI ( f )  
sin(t / Tb ) sin[ (t  Tb ) / Tb0,] otherwise
hI (t )  
t / Tb  (t  Tb ) / Tb
Tb2 sin(t / Tb )

t (Tb  t )
 The tails of h(t) decay as 1/|t|2, which is a faster rate of decay than 1/|t| encountered in
the ideal Nyquist channel.
 Let represent the estimate of the original pulse ak as conceived by the receiver at
time t=kTb
 Decision feedback : technique of using a stored estimate of the previous symbol
 Propagate : drawback, once error are made, they tend to propagate through the output
 Precoding : practical means of avoiding the error propagation phenomenon before the
duo binary coding
d k  bk  d k 1
 symbol 1 if either symbol bk or d k 1 is 1

dk  
symbol 0 otherwise
 {dk} is applied to a pulse-amplitude modulator, producing a corresponding two-level

sequence of short pulse {ak}, where +1 or –1 as before
ck  ak  ak 1
 0 if data symbol bk is 1
ck  
2 if data symbol bk is 0
 |ck|=1 : random guess in favor of symbol 1 or 0

Modified Duo-binary Signaling :
 Nonzero at the origin : undesirable

 Subtracting amplitude-modulated pulses spaced 2Tb second
ck  ak  ak 1
H IV ( f )  H Nyquist ( f )[1  exp( j 4 fTb )]

 2 jH Nyquist ( f )sin(2 fTb )exp(  j 2 fTb )
2 j sin(2 fTb ) exp( j 2 fTb ), | f | 1/ 2Tb

H IV ( f )  
 0, elsewhere
sin( t / Tb ) sin[ (t  2Tb ) / Tb ]

hIV (t )  
 t / Tb  (t  2Tb ) / Tb
2Tb2 sin( t / Tb )

 t (2Tb  t )

 precoding
d k  bk  d k 2
 symbol 1 if either symbol bk or d k 2 is 1

symbol 0 otherwise

If | ck | 1, say symbol bk is 1
If | ck | 1, say symbol bk is 0
Generalized form of correlative-level coding:
N 1
 t 
h(t )   wn sin c  n 
n  Tb 
Baseband M-ary PAM Transmission:

 Produce one of M possible amplitude level
 T : symbol duration
 1/T: signaling rate, symbol per second, bauds
– Equal to log2M bit per second
 Tb : bit duration of equivalent binary PAM :
 To realize the same average probability of symbol error, transmitted power must be
increased by a factor of M2/log2M compared to binary PAM
Tapped-delay-line equalization :
 Approach to high speed transmission

– Combination of two basic signal-processing operation
– Discrete PAM
– Linear modulation scheme
 The number of detectable amplitude levels is often limited by ISI
 Residual distortion for ISI : limiting factor on data rate of the system

 Equalization : to compensate for the residual distortion
 Equalizer : filter
– A device well-suited for the design of a linear equalizer is the tapped-delay-
line filter
– Total number of taps is chosen to be (2N+1)
N
h (t )   w  (t  kT )
k  N
k
 P(t) is equal to the convolution of c(t) and h(t)
 nT=t sampling time, discrete convolution sum
N
p(t )  c(t )  h(t )  c(t )   w  (t  kT )
k  N
k
 N N
  wk c(t )   (t  kT ) 
k  N
 w c(t  kT )
k  N
k
N

p (nT )   wk c((n  k )T )
 k  N

 Nyquist criterion for distortionless transmission, with T used in place of Tb,
normalized condition p(0)=1
1, n  0 1, n0

p (nT )   
0, n  0 0, n  1,  2, ....., N
 Zero-forcing equalizer
– Optimum in the sense that it minimizes the peak distortion(ISI) – worst case
– Simple implementation
– The longer equalizer, the more the ideal condition for distortionless
transmission
Adaptive Equalizer :
 The channel is usually time varying

– Difference in the transmission characteristics of the individual links that may
be switched together
– Differences in the number of links in a connection
 Adaptive equalization
– Adjust itself by operating on the the input signal
 Training sequence
– Precall equalization

– Channel changes little during an average data call
 Prechannel equalization
– Require the feedback channel
 Postchannel equalization
 synchronous
– Tap spacing is the same as the symbol duration of transmitted signal
Least-Mean-Square Algorithm:
 Adaptation may be achieved

– By observing the error b/w desired pulse shape and actual pulse shape
– Using this error to estimate the direction in which the tap-weight should be
changed
 Mean-square error criterion
– More general in application
– Less sensitive to timing perturbations
 : desired response, : error signal, : actual response
 Mean-square error is defined by cos function
  E en2 
 Ensemble-averaged cross-correlation
  e   y 
 2 E en n   2 E  en n   2 E  en xn  k   2 Rex (k )
wk  wk   wk 
Rex (k )  E  en xn  k 
 Optimality condition for minimum mean-square error

 0 for k  0,  1,....,  N


Mean-square error is a second-order and a parabolic function of tap weights as a multi
w k
dimensional bowl-shaped surface
 Adaptive process is a successive adjustments of tap-weight seeking the bottom of the
bowl(minimum value )
 Steepest descent algorithm
– The successive adjustments to the tap-weight in direction opposite to the
vector of gradient )
– Recursive formula (m : step size parameter)
1 
wk ( n  1)  wk ( n)  m , k  0,  1,....,  N
2 wk
 wk (n)  m Rex (k ), k  0,  1,....,  N
 Least-Mean-Square Algorithm
– Steepest-descent algorithm is not available in an unknown environment
– Approximation to the steepest descent algorithm using instantaneous estimate
)
Rex ( k )  en xn k
) )
wk (n  1)  wk ( n)  m en xn  k
 LMS is a feedback system
 In the case of small m, roughly similar to steepest descent algorithm

Operation of the equalizer:
 square error Training mode

– Known sequence is transmitted and synchronized version is generated in the
receiver
– Use the training sequence, so called pseudo-noise(PN) sequence
 Decision-directed mode
– After training sequence is completed
– Track relatively slow variation in channel characteristic
 Large m : fast tracking, excess mean
Implementation Approaches:
 Analog
– CCD, Tap-weight is stored in digital memory, analog sample and
multiplication
– Symbol rate is too high
 Digital
– Sample is quantized and stored in shift register
– Tap weight is stored in shift register, digital multiplication
 Programmable digital
– Microprocessor
– Flexibility
– Same H/W may be time shared
Decision-Feedback equalization:

 Baseband channel impulse response : {hn}, input : {xn}
yn   hk xn  k
k
 h0 xn   hk xn  k   hk xn  k
k 0 k 0
 Using data decisions made on the basis of precursor to take care of the postcursors
– The decision would obviously have to be correct
 Feed forward section : tapped-delay-line equalizer

 Feedback section : the decision is made on previously detected symbols of the input
sequence
– Nonlinear feedback loop by decision device
) ) )
 wn(1)  x  wn(1)1  wn(1)1  m1en xn
cn   ) (2)  vn   )n  ) ) )
 wn  an  en  an  cnT vn wn(2)1  wn(2)1  m1en an
Pulse Coded Modulation

 PCM generation and reconstruction ,
 Quantization noise,
 Differential PCM systems (DPCM),
 Delta modulation, adaptive delta modulation,
 Noise in PCM and DM systems
Digital Transmission of Analog Signals:

PCM, DPCM and DM
6.1 Introduction

Quite a few of the information bearing signals, such as speech, music, video, etc., are
analog in nature; that is, they are functions of the continuous variable t and for any t = t1,
their value can lie anywhere in the interval, say − A to A. Also, these signals are of the
baseband variety. If there is a channel that can support baseband transmission, we can easily
set up a baseband communication system. In such a system, the transmitter could be as
simple as just a power amplifier so that the signal that is transmitted could be received at the
destination with some minimum power level, even after being subject to attenuation during
propagation on the channel. In such a situation, even the receiver could have a very simple
structure; an appropriate filter (to eliminate the out of band spectral components) followed by
an amplifier. If a baseband channel is not available but have access to a passband channel,
(such as ionosphere channel, satellite channel etc.) an appropriate CW modulation scheme
discussed earlier could be used to shift the baseband spectrum to the passband of the given
channel. Interesting enough, it is possible to transmit the analog information in a digital
format. Though there are many ways of doing it, in this chapter, we shall explore three such
techniques, which have found widespread acceptance. These are: Pulse Code Modulation
(PCM), Differential Pulse Code Modulation (DPCM)n and Delta Modulation (DM). Before
we get into the details of these techniques, let us summarize the benefits of digital
transmission. For simplicity, we shall assume that information is being transmitted by a
sequence of binary pulses
.i) during the course of propagation on the channel, a transmitted pulse becomes
gradually distorted due to the non-ideal transmission characteristic of the channel. Also,
various unwanted signals (usually termed interference and noise) will cause further
deterioration of the information bearing pulse. However, as there are only two types of
signals that are being transmitted, it is possible for us to identify (with a very high
probability) a given transmitted pulse at some appropriate intermediate point on the channel
and regenerate a clean pulse. In this way, be completely eliminating the effect of distortion
and noise till the point of regeneration. (In long-haul PCM telephony, regeneration is done
every few kilometers, with the help of regenerative repeaters.) Clearly, such an operation is
not possible if the transmitted signal was analog because there is nothing like a reference
waveform that can be regenerated.
ii) Storing the messages in digital form and forwarding or redirecting them at a later point in
time is quite simple.
iii) Coding the message sequence to take care of the channel noise, encrypting for secure
communication can easily be accomplished in the digital domain.
iv)Mixing the signals is easy. All signals look alike after conversion to digital form
independent of the source (or language!). Hence they can easily be multiplexed (and
demultiplexed)
6.2 The PCM system

Two basic operations in the conversion of analog signal into the digital is time
discretization and amplitude discretization. In the context of PCM, the former is
accomplished with the sampling operation and the latter by means of quantization. In
addition, PCM involves another step, namely, conversion of quantized amplitudes into a
sequence of simpler pulse patterns (usually binary), generally called as code words.

(The word code in pulse code modulation refers to the fact that every quantized
sample is converted to an R -bit code word.)
Fig. 6.1 illustrates a PCM system.

Here, m(t ) is the information bearing message signal that is to be transmitted
digitally. m(t ) is first sampled and then quantized. The output of the sampler is
Ts is the sampling period and n is the appropriate integer.
is called the sampling rate or sampling frequency.
The quantizer converts each sample to one of the values that is closest to it from
among a pre-selected set of discrete amplitudes. The encoder represents each one of these
quantized samples by an R -bit code word. This bit stream travels on the channel and reaches
the receiving end. With fs as the sampling rate and R -bits per code word, the bit rate of the
PCM System is
The decoder converts the R -bit code words into the corresponding (discrete)
amplitudes. Finally, the reconstruction filter, acting on these discrete amplitudes, produces the
analog signal, denoted by m’(t ) . If there are no channel errors, then m’(t ) approx= m(t ) .
777773333333333333333333333333333333333333333333333333
444444477744444444ggggggggggggggggggg77777777774444477777777777

Pulse-Amplitude Modulation :
Let s (t ) denote the sequence of flat - top pulses as


s (t )   m(nT ) h(t  nT )
n  
s s (3.10)
 1, 0 t  T
1
h (t )   , t  0, t  T (3.11)
2
 0, otherwise

The instantaneously sampled version of m(t ) is

m ( t )   m(nT ) (t  nT )
n  
s s (3.12)

m ( t )  h ( t )   
m ( )h (t   )d
 
   m(nT ) (  nT )h(t   )d

n  
s s
 
  m(nT ) 
n  
s

 (  nTs )h (t   )d (3.13)
Using the sifting property , we have


m ( t )  h ( t )   m(nT )h(t  nT )
n  
s s (3.14)
The PAM signal s (t ) is

s (t )  m (t )  h(t ) (3.15)
 S ( f )  Mδ ( f ) H ( f ) (3.16)

Recall (3.2) g  (t )  fs  G( f
m  
 mf s ) (3.2)

M ( f )  f s M( f
k  
 k fs ) (3.17)

S( f )  fs M( f
k  
 k f s )H ( f ) (3.18)

Pulse Amplitude Modulation – Natural and Flat-Top Sampling:
 The most common technique for sampling voice in PCM systems is to a sample-and-
hold circuit.
 The instantaneous amplitude of the analog (voice) signal is held as a constant charge
on a capacitor for the duration of the sampling period Ts.
 This technique is useful for holding the sample constant while other processing is
taking place, but it alters the frequency spectrum and introduces an error, called
aperture error, resulting in an inability to recover exactly the original analog signal.
 The amount of error depends on how mach the analog changes during the holding
time, called aperture time.
 To estimate the maximum voltage error possible, determine the maximum slope of the
analog signal and multiply it by the aperture time DT

Recovering the original message signal m(t) from PAM signal :
Where the filter bandwidth is W

The filter output is f s M ( f ) H ( f ) . Note that the
Fourier transform of h (t ) is given by
H ( f )  T sinc( f T ) exp( j f T ) (3.19)
amplitude distortion delay  T
2
 aparture effect
Let the equalizer response is
1 1 f
  (3.20)
H ( f ) T sinc( f T ) sin( f T )
Ideally the original signal m(t ) can be recovered completely.
Other Forms of Pulse Modulation:
In pulse width modulation (PWM), the width of each pulse is made directly proportional
to the amplitude of the information signal.
 In pulse position modulation, constant-width pulses are used, and the position or time of
occurrence of each pulse from some reference time is made directly proportional to the
amplitude of the information signal.

Pulse Code Modulation (PCM) :
 Pulse code modulation (PCM) is produced by analog-to-digital conversion process.
 As in the case of other pulse modulation techniques, the rate at which samples are
taken and encoded must conform to the Nyquist sampling rate.
 The sampling rate must be greater than, twice the highest frequency in the analog
signal, fs > 2fA(max)
Quantization Process:
Define partition cell

J k :  mk  m  mk 1 , k  1,2,, L (3.21)
Where mk is the decision level or the decision threshold.
Amplitude quantization : The process of transforming the
sample amplitude m( nTs ) into a discrete amplitude
 ( nTs ) as shown in Fig 3.9
If m(t )  J k then the quantizer output is νk where νk , k  1,2,, L
are the representation or reconstruction levels , mk 1  mk is the step size.
The mapping   g(m ) (3.22)
is called the quantizer characteristic, which is a staircase function.
Figure 3.10 Two types of quantization: (a) midtread and (b) midrise.

Quantization Noise:
Figure 3.11 Illustration of the quantization process

Let the quantization error be denoted by the random
variable Q of sample value q
q  m  (3.23)
Q  M  V , ( E[ M ]  0) (3.24)
Assuming a uniform quantizer of the midrise type
2m max
the step - size is   (3.25)
L
 m max  m  m max , L : total number of levels
 1  
 ,  q
f Q (q)    2 2 (3.26)

 0, otherwise
 
1
 Q2  E[Q 2 ]  
2

2
q 2 f Q ( q ) dq 
 

2

2
q 2 dq
2
 (3.28)
12
When the quatized sample is expressed in binary form,
L  2R (3.29)
where R is the number of bits per sample
R  log 2 L (3.30)
2m max
 (3.31)
2R
1 2
 Q2  mmax 22 R (3.32)
3
Let P denote the average power of m(t )
P
 ( SNR ) o 
 Q2
3P
( 2
)2 2 R (3.33)
mmax
(SNR)o increases exponentially with increasing R (bandwidth).

Pulse Code Modulation (PCM):
Figure 3.13 The basic elements of a PCM system

Quantization (nonuniform quantizer):
Compression laws. (a) m -law. (b) A-law.
m - law
log(1  m m )
  (3.48)
log(1  m )
dm log(1  m )
 (1  m m ) (3.49)
d m
A - law
 A(m) 1
 0 m 
 1  log A A
  (3.50)
1  log( A m ) 1
  m 1

 1  log A A
 1  log A 1
dm  0 m 
 A A (3.51)
d 1
(1  A) m  m 1
 A

Figure 3.15 Line codes for the electrical representations of binary data.
(a) Unipolar NRZ signaling. (b) Polar NRZ signaling.
(c) Unipolar RZ signaling. (d) Bipolar RZ signaling.
(e) Split-phase or Manchester code.
Noise consideration in PCM systems:
(Channel noise, quantization noise)

Time-Division Multiplexing(TDM):
Digital Multiplexers :
Virtues, Limitations and Modifications of PCM:
Advantages of PCM
1. Robustness to noise and interference
2. Efficient regeneration
3. Efficient SNR and bandwidth trade-off
4. Uniform format
5. Ease add and drop
6. Secure

Delta Modulation (DM) :
Let m n   m(nTs ) , n  0,1,2,

where Ts is the sampling period and m( nTs ) is a sample of m(t ).
The error signal is
e n   m n  mq  n  1 (3.52)
eq  n   sgn(e n ) (3.53)
mq  n  mq  n  1  eq  n (3.54)
where mq  n is the quantizer output , eq  n is
the quantized version of e n , and  is the step size
The modulator consists of a comparator, a quantizer, and an accumulator
The output of the accumulator is
n
mq  n     sgn(e i  )
i 1
n
  eq  i  (3.55)
i 1

Two types of quantization errors:
Slope Overload Distortion and Granular Noise:

Denote the quantization error by q n  ,
mq  n  m n   q n (3.56)
Recall (3.52) , we have
e n  m n  m n  1  q n  1 (3.57)
Except for q n  1, the quantizer input is a first
backward difference of the input signal
To avoid slope - overload distortion , we require
 dm(t )
(slope)  max (3.58)
Ts dt
On the other hand, granular noise occurs when step size
 is too large relative to the local slope of m(t ).
Delta-Sigma modulation (sigma-delta modulation):
The    modulation which has an integrator can relieve the draw back of delta
modulation (differentiator).
Beneficial effects of using integrator:
1. Pre-emphasize the low-frequency content
2. Increase correlation between adjacent samples
(reduce the variance of the error signal at the quantizer input)
3. Simplify receiver design
Because the transmitter has an integrator, the receiver consists simply of a low-pass filter.
(The differentiator in the conventional DM receiver is cancelled by the integrator)

Linear Prediction (to reduce the sampling rate):
Consider a finite-duration impulse response (FIR) discrete-time filter which consists of three
blocks :
1. Set of p ( p: prediction order) unit-delay elements (z-1)
2. Set of multipliers with coefficients w1,w2,…wp
3. Set of adders (  )
Figure 3.27 Block diagram illustrating the linear adaptive prediction process
The filter o utput (Th

p
ˆ
x n    wk x
k 1
The prediction er ror

e n   x n   ˆ
x
Let the index of per
J  E e  2
 n  
Find w1 , w2 ,  , w p t
Fr om (3 .5 9) (3.60 ) a
 
p
J  E x 2
 n   2 w
k 1
p p
   w j wk E 
j 1 k 1

Linear adaptive prediction :
The predictor is adaptive in the follow sense

1. Compute wk , k  1,2,  , p, starting any initial values
2. Do iteration using the method of steepest descent
Define the gradient vector
J
gk  , k  1,2 , ,p (3.68)
wk
wk  n denotes the value at iteration n . Then update wk  n  1
1
wk  n  1  wk  n 
mg k , k  1,2 , ,p (3.69)
2
1
where m is a step - size parameter and is for convenience
2
of presentation.
Assume X (t ) is stationary process with zero mean ( E[ x[n]]  0)

 X2  E  x 2  n    ( E  x n   ) 2

 E x 2  n 
The autocorrelation
R X (  kTs )  R X  k   E  x n x n  k  
We may simplify J as
p p p
J   X2  2 wk R X  k    w j wk R X  k  j  (3.63)
k 1 j 1 k 1
J p
 2 R X  k   2 w j R X  k  j   0
wk j 1
p
 w R k  j  R k   R  k 
j 1
j X X X , k  1,2 , ,p (3.64)
(3.64) are called Wiener - Hopf equations
as , if R X1 exists w 0  R X1rX (3.66)

where 
w 0  w1 , w2 ,  , w p  T
rX  [ R X [1], R X [2],..., R X [ p ]] T
 R X  0 R X 1  R X  p  1 
 R 1 R X  0  R X  p  2 
RX  X
    
 
 R X  p  1 R X  p  2  R X  0 
R X  0 , R X 1 ,  , R X  p 
Substituting (3.64) into (3.63) yields
p p
J min   X2  2 wk R X  k    wk R X  k 
k 1 k 1
p
  X2   wk R X  k 
k 1
 2
X  rXT w 0   X2  rXT R X1rX (3.67)
rXT R X1rX  0, J min is always less than  X2
Differential Pulse-Code Modulation (DPCM):
Usually PCM has the sampling rate higher than the Nyquist rate .The encode signal
contains redundant information. DPCM can efficiently remove this redundancy.
Figure 3.28 DPCM system. (a) Transmitter. (b) Receiver.
Input signal to the quantizer is defined by:
e n   m n   m ˆ  n (3.74)
mˆ  n  is a prediction value.
The quantizer output is
eq  n   e n   q n  (3.75)
where q n  is quantization error.
The prediction filter input is
mq  n   m
ˆ  n   e n   q n  (3.77)
m n 
 mq  n   Gain:
Processing m n   q n  (3.78)

Adaptive Differential Pulse-Code Modulation (ADPCM):
Need for coding speech at low bit rates , we have two aims in mind:
1. Remove redundancies from the speech signal as far as possible.
2. Assign the available bits in a perceptually efficient manner.
The (SNR)o of the DPCM system is

 M2
(SNR)o  2 (3.79)
Q
where  M2 and  Q2 are variances of m n ( E[ m[ n]]  0) and q n
 M2  E2
(SNR)o  ( )( )
 E2  Q2
 G p (SNR )Q (3.80)
where  E2 is the variance of the predictions error
and the signal - to - quantization noise ratio is
 E2
(SNR )Q  (3.81)
 Q2
 M2
Processing Gain, G p  (3.82)
 E2
Design a prediction filter to maximize G p (minimize  E2 )
Figure 3.29 Adaptive quantization with backward estimation (AQB).
Figure 3.30 Adaptive prediction with backward estimation (APB).

UNIT II:
Information Theory
 Information and entropy,
 Conditional entropy and redundancy,
 Bandwidth- S/N trade off,
 Hartley Shannon law,
 Shannon Fano coding,

 mutual, information,
 Information loss due to noise,
 Source codings,- Huffman code, variable length coding
 Source coding to increase average information per bit,
 Lossy source Coding.
INFORMATION THEORY AND CODING TECHNIQUES
Information sources
Definition:
The set of source symbols is called the source alphabet, and the elements of the set are
called the symbols or letters.
The number of possible answers ‘ r ’ should be linked to “information.”
“Information” should be additive in some sense.
We define the following measure of information:
Where ‘ r ’ is the number of all possible outcome so far an do m message U.
Using this definition we can confirm that it has the wanted property of additivity:
The basis ‘b’ of the logarithm b is only a change of units without actually changing
the amount of information it describes.
Classification of information sources
1. Discrete memory less.

2. Memory.

Discrete memory less source (DMS) can be characterized by “the list of the symbols, the
probability assignment to these symbols, and the specification of the rate of generating these
symbols by the source”.
1. Information should be proportion to the uncertainty of an outcome.

2. Information contained in independent outcome should add.
Information content of a symbol:
Let us consider a discrete memory less source (DMS) denoted by X and having the
alphabet {U1, U2, U3, ……Um}. The information content of the symbol xi, denoted by I(x i) is
defined as
I(U) = logb = - log b P(U)
Where P(U) is the probability of occurrence of symbol U
Units of I(xi):
For two important and one unimportant special cases of b it has been agreed to use the
following names for these units:
b =2(log2): bit,
b = e (ln): nat (natural logarithm),
b =10(log10): Hartley.
The conversation of these units to other units is given as
log2a=
Uncertainty or Entropy (i.e Average information)
Definition:
In order to get the information content of the symbol, the flow information on the
symbol can fluctuate widely because of randomness involved into the section of symbols.
The uncertainty or entropy of a discrete random variable (RV) ‘U’ is defined as
H(U)= E[I(u)]=

where PU(·)denotes the probability mass function (PMF)2 of the RV U, and where the
support of P U is defined as
We will usually neglect to mention “support” when we sum over PU(u) · log b PU(u), i.e., we
implicitly assume that we exclude all u
With zero probability PU(u)=0.
Entropy for binary source
It may be noted that for a binary source U which genets independent symbols 0 and 1
with equal probability, the source entropy H(u) is
H(u) = - log2 - log2 = 1 b/symbol
Bounds on H(U)
If U has r possible values, then 0 ≤ H(U) ≤ log r,
Where
H(U)=0 if, and only if, PU(u)=1 for some u,
H(U)=log r if, and only if, PU(u)= 1/r ∀ u.
Hence, H(U) ≥ 0.Equalitycanonlybeachievedif −PU(u)log2 PU(u)=0
For all u ∈ supp(PU),i.e., PU(u)=1forall u ∈ supp(PU).
To derive the upper bound we use at rick that is quite common in.
Formation theory: We take the deference and try to show that it must be non positive.

Equality can only be achieved if
1. In the IT Inequality ξ =1,i.e.,if 1r·PU(u)=1=⇒ PU(u)= 1r ,for all u;

2. |supp(PU)| = r.
Note that if Condition1 is satisfied, Condition 2 is also satisfied.

Conditional Entropy
Similar to probability of random vectors, there is nothing really new about

conditional probabilities given that a particular event Y = y has occurred.
The conditional entropy or conditional uncertainty of the RV X given the event Y = y

is defined as
Note that the definition is identical to before apart from that everything is conditioned
on the event Y = y
Note that the conditional entropy given the event Y = y is a function of y. Since Y is
also a RV, we can now average over all possible events Y = y according to the
probabilities of each event. This will lead to the averaged.
• Forward Error Correction (FEC)

– Coding designed so that errors can be corrected at the receiver
– Appropriate for delay sensitive and one-way transmission (e.g., broadcast TV)
of data
– Two main types, namely block codes and convolution codes. We will only
look at block codes

UNIT III :
Linear Block Codes
 Matrix description of linear block codes,

 Matrix description of linear block codes,
 Error detection and error correction capabilities of linear block codes
 Cyclic codes: algebraic structure, encoding, syndrome calculation, decoding
Block Codes:
• We will consider only binary data

• Data is grouped into blocks of length k bits (dataword)
• Each dataword is coded into blocks of length n bits (codeword), where in general n>k
• This is known as an (n,k) block code
• A vector notation is used for the datawords and codewords,
– Dataword d = (d1 d2….dk)
– Codeword c = (c1 c2……..cn)
• The redundancy introduced by the code is quantified by the code rate,
– Code rate = k/n
– i.e., the higher the redundancy, the lower the code rate
Hamming Distance:
• Error control capability is determined by the Hamming distance

• The Hamming distance between two code words is equal to the number of differences
between them, e.g.,
10011011
11010010 have a Hamming distance = 3
• Alternatively, can compute by adding codeword (mod 2)

=01001001 (now count up the ones)
• The maximum number of detectable errors is

d min  1
• That is the maximum number of correctable errors is given by,

 d  1
t   min 
 2 
Where d min is the minimum Hamming distance between 2 codeword’s and means
the smallest integer

Linear Block Codes:
• As seen from the second Parity Code example, it is possible to use a table to hold all
the codewords for a code and to look-up the appropriate codeword based on the
supplied dataword
• Alternatively, it is possible to create codewords by addition of other codewords. This
has the advantage that there is now no longer the need to held every possible
codeword in the table.
• If there are k data bits, all that is required is to hold k linearly independent codewords,
i.e., a set of k codewords none of which can be produced by linear combinations of 2
or more codewords in the set.
• The easiest way to find k linearly independent codewords is to choose those which
have ‘1’ in just one of the first k positions and ‘0’ in the other k-1 of the first k
positions.
• For example for a (7,4) code, only four codewords are required, e.g.,
1 0 0 0 1 1 0
0 1 0 0 1 0 1
0 0 1 0 0 1 1
0 0 0 1 1 1 1
• So, to obtain the codeword for dataword 1011, the first, third and fourth codewords in
the list are added together, giving 1011010
• This process will now be described in more detail
• An (n,k) block code has code vectors

d=(d1 d2….dk) and
c=(c1 c2……..cn)
• The block coding process can be written as c=dG
where G is the Generator Matrix
 a11 a12 ... a1n   a 1 

a a22 ... a2 n  a 2 
G   21 
 . . ... .   . 
   
 ak 1 ak 2 ... akn  a k 
• Thus,
k
c   dia i
i 1

•ai must be linearly independent, i.e.,
Since codewords are given by summations of the ai vectors, then to avoid 2 datawords
having the same codeword the ai vectors must be linearly independent.
• Sum (mod 2) of any 2 codewords is also a codeword, i.e.,

Since for datawords d1 and d2 we have;
d 3  d1  d 2
k k k k
So, c3   d 3i a i   (d1i  d 2i )a i  d1i a i   d 2i a i
i 1 i 1 i 1 i 1
c 3  c1  c 2
Error Correcting Power of LBC:
• The Hamming distance of a linear block code (LBC) is simply the minimum
Hamming weight (number of 1’s or equivalently the distance from the all 0 codeword)
of the non-zero codewords
• Note d(c1,c2) = w(c1+ c2) as shown previously
• For an LBC, c1+ c2=c3
• So min (d(c1,c2)) = min (w(c1+ c2)) = min (w(c3))
• Therefore to find min Hamming distance just need to search among the 2k codewords
to find the min Hamming weight – far simpler than doing a pair wise check for all
possible codewords.
•
Linear Block Codes – example 1:
• For example a (4,2) code, suppose;

1 0 1 1
G  
0 1 0 1

a1 = [1011]
a2 = [0101]
• For d = [1 1], then;
1 0 1 1
 0 1 0 1
c 
_ _ _ _
 1 1 1 0
Linear Block Codes – example 2:
• A (6,5) code wit h 1 0 0 0 0 1

0 1 0 0 0 1

G  0 0 1 0 0 1
 
0 0 0 1 0 1
0 0 0 0 1 1

• Is an even single parity code
Systematic Codes:
• For a systematic block code the dataword appears unaltered in the codeword – usually
at the start
• The generator matrix has the structure,
1 0 .. 0 p11 p12 .. p1R 

0 1 .. 0 p21 p22 .. p2 R 
G   I | P
.. .. .. .. .. .. .. .. 
 
R=n-k 0 0 .. 1 pk 1 pk 2 .. pkR 
• is often referred to as parity bits

I is k*k identity matrix. Ensures data word appears as beginning of codeword P is k*R matrix.
Decoding Linear Codes:
• One possibility is a ROM look-up table

• In this case received codeword is used as an address
• Example – Even single parity check code;
Address Data
000000 0
000001 1
000010 1
000011 0
……… .
• Data output is the error flag, i.e., 0 – codeword ok,

• If no error, data word is first k bits of codeword
• For an error correcting code the ROM can also store data words
• Another possibility is algebraic decoding, i.e., the error flag is computed from the
received codeword (as in the case of simple parity codes)
• How can this method be extended to more complex error detection and correction
codes?
Parity Check Matrix:
• A linear block code is a linear subspace S sub of all length n vectors (Space S)
• Consider the subset S null of all length n vectors in space S that are orthogonal to all
length n vectors in S sub
• It can be shown that the dimensionality of S null is n-k, where n is the dimensionality of
S and k is the dimensionality of S sub
• It can also be shown that S null is a valid subspace of S and consequently S sub is also
the null space of S null

• S null can be represented by its basis vectors. In this case the generator basis vectors (or
‘generator matrix’ H) denote the generator matrix for S null - of dimension n-k = R
• This matrix is called the parity check matrix of the code defined by G, where G is
obviously the generator matrix for S sub - of dimension k
• Note that the number of vectors in the basis defines the dimension of the subspace
• So the dimension of H is n-k (= R) and all vectors in the null space are orthogonal to
all the vectors of the code
• Since the rows of H, namely the vectors bi are members of the null space they are
orthogonal to any code vector
• So a vector y is a codeword only if yHT=0
• Note that a linear block code can be specified by either G or H
Parity Check Matrix:
 b11 b12 ... b1n   b1 
b b22 ... b2 n   b 2 
H   21 
R = n - k . . ... .   . 
   
bR1 bR 2 ... bRn  b R 
• So H is used to check if a codeword is valid,

• The rows of H, namely, bi, are chosen to be orthogonal to rows of G, namely ai
• Consequently the dot product of any valid codeword with any bi is zero
This is so since, k
c   di a i
i 1
and so, k k
b j .c  b j . d i a i   d i (a i .b j )  0
i 1 i 1
• This means that a codeword is valid (but not necessarily correct) only if cHT = 0. To
ensure this it is required that the rows of H are independent and are orthogonal to the
rows of G
• That is the bi span the remaining R (= n - k) dimensions of the codespace
• For example consider a (3,2) code. In this case G has 2 rows, a1 and a2
• Consequently all valid codewords sit in the subspace (in this case a plane) spanned by
a1 and a2
• In this example the H matrix has only one row, namely b1. This vector is orthogonal
to the plane containing the rows of the G matrix, i.e., a1 and a2
• Any received codeword which is not in the plane containing a1 and a2 (i.e., an invalid
codeword) will thus have a component in the direction of b1 yielding a nonzero dot
product between itself and b1.
Error Syndrome:
• For error correcting codes we need a method to compute the required correction
• To do this we use the Error Syndrome, s of a received codeword, cr
s = crHT
• If cr is corrupted by the addition of an error vector, e, then

cr = c + e
and s = (c + e) HT = cHT + eHT
s = 0 + eHT

Syndrome depends only on the error
• That is, we can add the same error pattern to different code words and get the same
syndrome.
– There are 2(n - k) syndromes but 2n error patterns
– For example for a (3,2) code there are 2 syndromes and 8 error patterns
– Clearly no error correction possible in this case
– Another example. A (7,4) code has 8 syndromes and 128 error patterns.
– With 8 syndromes we can provide a different value to indicate single errors in
any of the 7 bit positions as well as the zero value to indicate no errors
• Now need to determine which error pattern caused the syndrome
• For systematic linear block codes, H is constructed as follows,

G = [ I | P] and so H = [-PT | I]
where I is the k*k identity for G and the R*R identity for H
• Example, (7,4) code, dmin= 3
1 0 0 0 0 1 1
0 1 0 0 1 0 1 0 1 1 1 1 0 0
G   I | P  
0 0 1 0 1 1 0 
H  - P | I  1
T
 0 1 1 0 1 0
 
0 0 0 1 1 1 1 1 1 0 1 0 0 1
Error Syndrome – Example:
• For a correct received codeword cr = [1101001]

In this case,
0 1 1
1 0 1

1 1 0
 
s  c r H T  1 1 0 1 0 0 1 1 1 1   0 0 0
1 0 0
 
0 1 0
0 0 1

Standard Array:

• The Standard Array is constructed as follows,
c1 (all zero) c2 …… cM s0
e1 c2+e1 …… cM+e1 s1
e2 c2+e2 …… cM+e2 s2
e3 c2+e3 …… cM+e3 s3
… …… …… …… …
eN c2+eN …… cM+eN sN
• The array has 2k columns (i.e., equal to the number of valid codewords) and 2R rows
(i.e., the number of syndromes)
•
Hamming Codes:
• We will consider a special class of SEC codes (i.e., Hamming distance = 3) where,
– Number of parity bits R = n – k and n = 2R – 1
– Syndrome has R bits
– 0 value implies zero errors
– 2R – 1 other syndrome values, i.e., one for each bit that might need to be
corrected
– This is achieved if each column of H is a different binary word – remember s
= eHT
• Systematic form of (7,4) Hamming code is,
1 0 0 0 0 1 1
0 0 1 1 1 1 0 0
1 0 0 1 0 1 
G   I | P  
0 0 1 0 1 1 0
 T

H  - P | I  1 0 1 1 0 1 0
  1 1 0 1 0 0 1
0 0 0 1 1 1 1
• The original form is non-systematic,
1 1 1 0 0 0 0
1 0 0 0 1 1 1 1
0 0 1 1 0 0
G  H 
0 1 1 0 0 1 1

0 1 0 1 0 1 0
  
1 0 1 0 1 0 1

1 1 0 1 0 0 1
• Compared with the systematic code, the column orders of both G and H are swapped
so that the columns of H are a binary count
• The column order is now 7, 6, 1, 5, 2, 3, 4, i.e., col. 1 in the non-systematic H is col. 7
in the systematic H.

Convolution Codes
 Encoding,
 Decoding using state Tree and trellis diagrams,
 Decoding using Viterbi algorithm,
 Comparison of error rates in coded and uncoded transmission.
Introduction:
• Convolution codes map information to code bits sequentially by convolving a

sequence of information bits with “generator” sequences
• A convolution encoder encodes K information bits to N>K code bits at one time step
• Convolutional codes can be regarded as block codes for which the encoder has a
certain structure such that we can express the encoding operation as convolution
• Convolutional codes are applied in applications that require good performance with
low implementation cost. They operate on code streams (not in blocks)
• Convolution codes have memory that utilizes previous bits to encode or decode
following bits (block codes are memoryless)
• Convolutional codes achieve good performance by expanding their memory depth
• Convolutional codes are denoted by (n,k,L), where L is code (or encoder) Memory
depth (number of register stages)
• Constraint length C=n(L+1) is defined as the number of encoded bits a message bit
can influence to
• Convolutional encoder, k = 1, n = 2, L=2

– Convolutional encoder is a finite state machine (FSM) processing
information bits in a serial manner
– Thus the generated code is a function of input and the state of the FSM
– In this (n,k,L) = (2,1,2) encoder each message bit influences a span of C=
n(L+1)=6 successive output bits = constraint length C
– Thus, for generation of n-bit output, we require n shift registers in k = 1
convolutional encoders

x ' j  m j 3  m j  2  m j
x '' j  m j3  m j1  m j
x ''' j  m j 2  m j

Here each message bit influencesn a span of C = n(L+1)=3(1+1)=6
successive output bits.
Convolution point of view in encoding and generator matrix:
Example: Using generator matrix

 g  [1 0 11] 
(1)
 g ( 2 )  [111 1] 
 
Representing convolutional codes: Code tree:
(n,k,L) = (2,1,2) encoder

 x ' j  m j 2  m j 1  m j

 x '' j  m j 2  m j
xout  x '1 x ''1 x '2 x ''2 x '3 x ''3 ...
Representing convolutional codes compactly: code trellis and state diagram:
State diagram

Inspecting state diagram: Structural properties of convolutional codes:
• Each new block of k input bits causes a transition into new state
• Hence there are 2k branches leaving each state
• Assuming encoder zero initial state, encoded word for any input of k bits can thus be
obtained. For instance, below for u=(1 1 1 0 1), encoded word v=(1 1, 1 0, 0 1, 0 1, 1
1, 1 0, 1 1, 1 1) is produced:
- encoder state diagram for (n,k,L)=(2,1,2) code
- note that the number of states is 2L+1 = 8
Distance for some convolutional codes:

THE VITERBI ALGORITHEM:
• Problem of optimum decoding is to find the minimum distance path from the initial
state back to initial state (below from S0 to S0). The minimum distance is the sum of
all path metrics that is maximized by the correct path.
.
ln p (y, x m )   j0 ln p ( y j | xmj )
• Exhaustive maximum likelihood method must search all the paths in phase trellis (2k
paths emerging/entering from 2 L+1 states for an (n,k,L) code)
• The Viterbi algorithm gets its efficiency via concentrating into survivor paths of the
trellis
THE SURVIVOR PATH:
• Assume for simplicity a convolutional code with k=1, and up to 2k = 2 branches can
enter each state in trellis diagram
• Assume optimal path passes S. Metric comparison is done by adding the metric of S
into S1 and S2. At the survivor path the accumulated metric is naturally smaller
(otherwise it could not be the optimum path)

• For this reason the non-survived path can be discarded -> all path alternatives need
not to be considered
• Note that in principle whole transmitted sequence must be received before decision.
However, in practice storing of states for input length of 5L is quite adequate

The maximum likelihood path:
The decoded ML code sequence is 11 10 10 11 00 00 00 whose Hamming distance to the

received sequence is 4 and the respective decoded sequence is 1 1 0 0 0 0 0 (why?). Note that
this is the minimum distance path.
(Black circles denote the deleted branches, dashed lines: '1' was applied)
How to end-up decoding?
• In the previous example it was assumed that the register was finally filled with zeros
thus finding the minimum distance path
• In practice with long code words zeroing requires feeding of long sequence of zeros to
the end of the message bits: this wastes channel capacity & introduces delay
• To avoid this path memory truncation is applied:
– Trace all the surviving paths to the depth where they merge
– Figure right shows a common point at a memory depth J
– J is a random variable whose applicable magnitude shown in the figure (5L)
has been experimentally tested for negligible error rate increase
– Note that this also introduces the delay of 5L!

J  5L stages of the trellis
Hamming Code Example:
• H(7,4)
• Generator matrix G: first 4-by-4 identical matrix
• Message information vector p
• Transmission vector x
• Received vector r
and error vector e
• Parity check matrix H

Error Correction:
• If there is no error, syndrome vector z=zeros
• If there is one error at location 2
• New syndrome vector z is

Example of CRC:
Example: Using generator matrix:
 g  [1 0 11] 
(1)
 g ( 2 )  [111 1] 
 
11
12 3 00  01
1 2311  01
1 411 4 2 4 410 3
01

correct:1+1+2+2+2=8;8  ( 0.11)  0.88
false:1+1+0+0+0=2;2  ( 2.30)  4.6
total path metric:  5.48

Turbo Codes:
• Back ground
– Turbo codes were proposed by Berrou and Glavieux in the 1993 International
Conference in Communications.
– Performance within 0.5 dB of the channel capacity limit for BPSK was
demonstrated.
• Features of turbo codes
– Parallel concatenated coding
– Recursive convolutional encoders
– Pseudo-random interleaving
– Iterative decoding
Motivation: Performance of Turbo Codes
• Comparison:
– Rate 1/2 Codes.
– K=5 turbo code.
– K=14 convolutional code.
• Plot is from:
– L. Perez, “Turbo Codes”, chapter 8 of Trellis Coding by C. Schlegel. IEEE
Press, 1997

Pseudo-random Interleaving:
• The coding dilemma:

– Shannon showed that large block-length random codes achieve channel
capacity.
– However, codes must have structure that permits decoding with reasonable
complexity.
– Codes with structure don’t perform as well as random codes.
– “Almost all codes are good, except those that we can think of.”
• Solution:
– Make the code appear random, while maintaining enough structure to permit
decoding.
– This is the purpose of the pseudo-random interleaver.
– Turbo codes possess random-like properties.
– However, since the interleaving pattern is known, decoding is possible.
Why Interleaving and Recursive Encoding?
• In a coded systems:
– Performance is dominated by low weight code words.
• A “good” code:
– will produce low weight outputs with very low probability.
• An RSC code:
– Produces low weight outputs with fairly low probability.
– However, some inputs still cause low weight outputs.
• Because of the interleaver:
– The probability that both encoders have inputs that cause low weight outputs
is very low.
– Therefore the parallel concatenation of both encoders will produce a “good”
code.
Iterative Decoding:

• There is one decoder for each elementary encoder.
• Each decoder estimates the a posteriori probability (APP) of each data bit.
• The APP’s are used as a priori information by the other decoder.
• Decoding continues for a set number of iterations.
– Performance generally improves from iteration to iteration, but follows a law
of diminishing returns
The Turbo-Principle:
Turbo codes get their name because the decoder uses feedback, like a turbo engine
Performance as a Function of Number of Iterations:
0
10
-1
10
1 iteration
-2
10
-3
2 iterations
10
BER
-4
10 6 iterations 3 iterations
-5
10 10 iterations
-6
10 18 iterations
-7
10
0.5 1 1.5 2
E b/No in dB
Turbo Code Summary:
• Turbo code advantages:

– Remarkable power efficiency in AWGN and flat-fading channels for
moderately low BER.
– Deign tradeoffs suitable for delivery of multimedia services.
• Turbo code disadvantages:
– Long latency.
– Poor performance at very low BER.
– Because turbo codes operate at very low SNR, channel estimation and tracking
is a critical issue.
• The principle of iterative or “turbo” processing can be applied to other problems.
– Turbo-multiuser detection can improve performance of coded multiple-access
systems.

UNIT IV
Digital Modulation Techniques
 Introduction, ASK, ASK Modulator, Coherent ASK detector, non-Coherent ASK
detector,
 Band width frequency spectrum of FSK,
 Non-Coherent FSK detector,
 Coherent FSK detector,
 FSK Detection using PLL,
 BPSK, Coherent PSK detection, QPSK, Differential PSK

ASK, OOK, MASK:
• The amplitude (or height) of the sine wave varies to transmit the ones and zeros
• One amplitude encodes a 0 while another amplitude encodes a 1 (a form of amplitude

modulation)
Binary amplitude shift keying, Bandwidth:
• d ≥ 0-related to the condition of the line
B = (1+d) x S = (1+d) x N x 1/r
implementation of binary ASK:
Frequency Shift Keying:

• One frequency encodes a 0 while another frequency encodes a 1 (a form of frequency
modulation)

 A cos 2f 2t  binary 1
s t   

A cos 2f 2t 
 binary 0
FSK Bandwidth:
• Limiting factor: Physical capabilities of the carrier

• Not susceptible to noise as much as ASK
• Applications
– On voice-grade lines, used up to 1200bps

– Used for high-frequency (3 to 30 MHz) radio transmission
– used at higher frequencies on LANs that use coaxial cable
DBPSK:
• Differential BPSK
– 0 = same phase as last signal element
– 1 = 180º shift from last signal element
  
A cos 2f c t 
  11
 4 

 3 

A cos 2f c t   01
s t   
 4 
 3 
A cos 2f c t   00
  4 


 
A cos 2f c t  
 4
10
Concept of a constellation :

M-ary PSK:
Using multiple phase angles with each angle having more than one amplitude, multiple
signals elements can be achieved
R R
D 
L log 2 M
– D = modulation rate, baud

– R = data rate, bps
– M = number of different signal elements = 2L
– L = number of bits per signal element
QAM:
– As an example of QAM, 12 different phases are combined with two different

amplitudes
– Since only 4 phase angles have 2 different amplitudes, there are a total of 16
combinations
– With 16 signal combinations, each baud equals 4 bits of information (2 ^ 4 =
16)
– Combine ASK and PSK such that each signal corresponds to multiple bits
– More phases than amplitudes

– Minimum bandwidth requirement same as ASK or PSK
QAM and QPR:
• QAM is a combination of ASK and PSK

– Two different signals sent simultaneously on the same carrier frequency
– M=4, 16, 32, 64, 128, 256

• Quadrature Partial Response (QPR)
– 3 levels (+1, 0, -1), so 9QPR, 49QPR

Offset quadrature phase-shift keying (OQPSK):
• QPSK can have 180 degree jump, amplitude fluctuation

• By offsetting the timing of the odd and even bits by one bit-period, or half a symbol-
period, the in-phase and quadrature components will never change at the same time.

Generation and Detection of Coherent BPSK:
Figure 6.26 Block diagrams for (a) binary FSK transmitter and (b) coherent binary FSK
receiver.
6.28
Figure 6.30 (a) Input binary sequence. (b) Waveform of scaled time
function s1f1(t). (c) Waveform of scaled time function s2f2(t). (d)
Waveform of the MSK signal s(t) obtained by adding s1f1(t) and
s2f2(t) on a bit-by-bit basis.

Figure 6.29 Signal-space diagram for MSK system.
Generation and Detection of MSK Signals:
Figure 6.31 Block diagrams for (a) MSK transmitter and (b) coherent MSK receiver

Eye Pattern:
 Experimental tool for such an evaluation in an insightful manner

– Synchronized superposition of all the signal of interest viewed within a
particular signaling interval
 Eye opening : interior region of the eye pattern

 In the case of an M-ary system, the eye pattern contains (M-1) eye opening, where M
is the number of discrete amplitude levels

Interpretation of Eye Diagram:

UNIT V :
Spread Spectrum Modulation
 Use of spread spectrum,

 direct sequence spread spectrum(DSSS),
 Code division multiple access,
 Ranging using DSSS Frequency Hopping spread spectrum,
 PN sequences: generation and characteristics,
 Synchronization in spread spectrum system,
 Advancements in the digital communication
SPREAD SPECTRUM MODULATION
• Spread data over wide bandwidth

• Makes jamming and interception harder
• Frequency hoping
– Signal broadcast over seemingly random series of frequencies
• Direct Sequence
– Each bit is represented by multiple bits in transmitted signal
– Chipping code
Spread Spectrum Concept:
• Input fed into channel encoder

– Produces narrow bandwidth analog signal around central frequency
• Signal modulated using sequence of digits
– Spreading code/sequence
– Typically generated by pseudo noise/pseudorandom number generator
• Increases bandwidth significantly
– Spreads spectrum
• Receiver uses same sequence to demodulate signal
• Demodulated signal fed into channel decoder
General Model of Spread Spectrum System:

Gains:
• Immunity from various noise and multipath distortion

– Including jamming
• Can hide/encrypt signals
– Only receiver who knows spreading code can retrieve signal
• Several users can share same higher bandwidth with little interference
– Cellular telephones
– Code division multiplexing (CDM)
– Code division multiple access (CDMA)
Pseudorandom Numbers:
• Generated by algorithm using initial seed

• Deterministic algorithm
– Not actually random
– If algorithm good, results pass reasonable tests of randomness
• Need to know algorithm and seed to predict sequence
Frequency Hopping Spread Spectrum (FHSS):
• Signal broadcast over seemingly random series of frequencies

• Receiver hops between frequencies in sync with transmitter
• Eavesdroppers hear unintelligible blips
• Jamming on one frequency affects only a few bits
Basic Operation:
• Typically 2k carriers frequencies forming 2k channels

• Channel spacing corresponds with bandwidth of input
• Each channel used for fixed interval
– 300 ms in IEEE 802.11
– Some number of bits transmitted using some encoding scheme
• May be fractions of bit (see later)
– Sequence dictated by spreading code
Frequency Hopping Example:

Frequency Hopping Spread Spectrum System (Transmitter):
Frequency Hopping Spread Spectrum System (Receiver):
Slow and Fast FHSS:
• Frequency shifted every Tc seconds

• Duration of signal element is Ts seconds
• Slow FHSS has Tc  Ts
• Fast FHSS has Tc < Ts
• Generally fast FHSS gives improved performance in noise (or jamming)

Slow Frequency Hop Spread Spectrum Using MFSK (M=4, k=2)
Fast Frequency Hop Spread Spectrum Using MFSK (M=4, k=2)

FHSS Performance Considerations:
• Typically large number of frequencies used

– Improved resistance to jamming
Direct Sequence Spread Spectrum (DSSS):
• Each bit represented by multiple bits using spreading code

• Spreading code spreads signal across wider frequency band
– In proportion to number of bits used
– 10 bit spreading code spreads signal across 10 times bandwidth of 1 bit code
• One method:
– Combine input with spreading code using XOR
– Input bit 1 inverts spreading code bit
– Input zero bit doesn’t alter spreading code bit
– Data rate equal to original spreading code
• Performance similar to FHSS
Direct Sequence Spread Spectrum Example:

Direct Sequence Spread Spectrum Transmitter:
Direct Sequence Spread Spectrum Receiver:

Direct Sequence Spread Spectrum Using BPSK Example:
Code Division Multiple Access (CDMA):
• Multiplexing Technique used with spread spectrum

• Start with data signal rate D
– Called bit data rate
• Break each bit into k chips according to fixed pattern specific to each user
– User’s code
• New channel has chip data rate kD chips per second
• E.g. k=6, three users (A,B,C) communicating with base receiver R
• Code for A = <1,-1,-1,1,-1,1>
• Code for B = <1,1,-1,-1,1,1>
• Code for C = <1,1,-1,1,1,-1>
CDMA Example:

•Consider A communicating with base
•Base knows A’s code
•Assume communication already synchronized
•A wants to send a 1
– Send chip pattern <1,-1,-1,1,-1,1>
• A’s code
• A wants to send 0
– Send chip[ pattern <-1,1,1,-1,1,-1>
• Complement of A’s code
• Decoder ignores other sources when using A’s code to decode
– Orthogonal codes
CDMA for DSSS:
•n users each using different orthogonal PN sequence

•Modulate each users data stream
– Using BPSK
• Multiply by spreading code of user
CDMA in a DSSS Environment:
20.Additional Topics:
Voice coders
Regenerative repeater
Feedback communications
Advancements in the digital communication
Signal space representation
Turbo codes

Voice coders
A vocoder (short for voice encoder) is an analysis/synthesis system, used to
reproduce human speech. In the encoder, the input is passed through a multiband filter, each
band is passed through an envelope follower, and the control signals from the envelope
followers are communicated to the decoder. The decoder applies these (amplitude) control
signals to corresponding filters in the (re)synthesizer.
It was originally developed as a speech coder for telecommunications applications in
the 1930s, the idea being to code speech for transmission. Its primary use in this fashion is for
secure radio communication, where voice has to be encrypted and then transmitted. The
advantage of this method of "encryption" is that no 'signal' is sent, but rather envelopes of the
band pass filters. The receiving unit needs to be set up in the same channel configuration to
resynthesize a version of the original signal spectrum. The vocoder as
both hardware and software has also been used extensively as an electronic.
Whereas the vocoder analyzes speech, transforms it into electronically transmitted
information, and recreates it, The Voder (from Voice Operating Demonstrator) generates
synthesized speech by means of a console with fifteen touch-sensitive keys and a pedal,
basically consisting of the "second half" of the vocoder, but with manual filter controls,
needing a highly trained operator.
The human voice consists of sounds generated by the opening and closing of
the glottis by the vocal cords, which produces a periodic waveform with many harmonics.
This basic sound is then filtered by the nose and throat (a complicated resonant piping
system) to produce differences in harmonic content (formants) in a controlled way, creating
the wide variety of sounds used in speech. There is another set of sounds, known as
the unvoiced and plosive sounds, which are created or modified by the mouth in different
fashions.
The vocoder examines speech by measuring how its spectral characteristics change
over time. This results in a series of numbers representing these modified frequencies at any
particular time as the user speaks. In simple terms, the signal is split into a number of
frequency bands (the larger this number, the more accurate the analysis) and the level of
signal present at each frequency band gives the instantaneous representation of the spectral
energy content. Thus, the vocoder dramatically reduces the amount of information needed to
store speech, from a complete recording to a series of numbers. To recreate speech, the
vocoder simply reverses the process, processing a broadband noise source by passing it
through a stage that filters the frequency content based on the originally recorded series of
numbers. Information about the instantaneous frequency (as distinct from spectral
characteristic) of the original voice signal is discarded; it wasn't important to preserve this for
the purposes of the vocoder's original use as an encryption aid, and it is this "dehumanizing"
quality of the vocoding process that has made it useful in creating special voice effects in
popular music and audio entertainment.
Since the vocoder process sends only the parameters of the vocal model over the
communication link, instead of a point by point recreation of the waveform, it allows a
significant reduction in the bandwidth required to transmit speech.

Modern vocoder implementations
Even with the need to record several frequencies, and the additional unvoiced sounds,
the compression of the vocoder system is impressive. Standard speech-recording systems
capture frequencies from about 500 Hz to 3400 Hz, where most of the frequencies used in
speech lie, typically using a sampling rate of 8 kHz (slightly greater than the Nyquist rate).
The sampling resolution is typically at least 12 or more bits per sample resolution (16 is
standard), for a final data rate in the range of 96-128 kbit/s. However, a good vocoder can
provide a reasonable good simulation of voice with as little as 2.4 kbit/s of data.
'Toll Quality' voice coders, such as ITU G.729, are used in many telephone networks.
G.729 in particular has a final data rate of 8 kbit/s with superb voice quality. G.723 achieves
slightly worse quality at data rates of 5.3 kbit/s and 6.4 kbit/s. Many voice systems use even
lower data rates, but below 5 kbit/s voice quality begins to drop rapidly.
Several vocoder systems are used in NSA encryption systems:
 LPC-10, FIPS Pub 137, 2400 bit/s, which uses linear predictive coding
 Code-excited linear prediction (CELP), 2400 and 4800 bit/s, Federal Standard 1016,
used in STU-III
 Continuously variable slope delta modulation (CVSD), 16 kbit/s, used in wide band
encryptors such as the KY-57.
 Mixed-excitation linear prediction (MELP), MIL STD 3005, 2400 bit/s, used in the
Future Narrowband Digital Terminal FNBDT, NSA's 21st century secure telephone.
 Adaptive Differential Pulse Code Modulation (ADPCM), former ITU-T G.721, 32

kbit/s used in STE secure telephone
(ADPCM is not a proper vocoder but rather a waveform codec. ITU has gathered G.721
along with some other ADPCM codecs into G.726.)
Vocoders are also currently used in developing psychophysics, linguistics, computational
neuroscience and cochlear implant research.
Modern vocoders that are used in communication equipment and in voice storage devices
today are based on the following algorithms:
 Algebraic code-excited linear prediction (ACELP 4.7 kbit/s – 24 kbit/s)[5]

 Mixed-excitation linear prediction (MELPe 2400, 1200 and 600 bit/s)[6]
 Multi-band excitation (AMBE 2000 bit/s – 9600 bit/s)[7]
 Sinusoidal-Pulsed Representation (SPR 300 bit/s – 4800 bit/s)[8]
 Tri-wave excited linear prediction (TWELP 2400 – 3600 bit/s)[9]
Linear prediction-based vocoders

Main article: Linear predictive coding
Since the late 1970s, most non-musical vocoders have been implemented using linear
prediction, whereby the target signal's spectral envelope (formant) is estimated by an all-
pole IIR filter. In linear prediction coding, the all-pole filter replaces the bandpass filter bank
of its predecessor and is used at the encoder to whiten the signal (i.e., flatten the spectrum)
and again at the decoder to re-apply the spectral shape of the target speech signal.

One advantage of this type of filtering is that the location of the linear predictor's
spectral peaks is entirely determined by the target signal, and can be as precise as allowed by
the time period to be filtered. This is in contrast with vocoders realized using fixed-width
filter banks, where spectral peaks can generally only be determined to be within the scope of
a given frequency band. LP filtering also has disadvantages in that signals with a large
number of constituent frequencies may exceed the number of frequencies that can be
represented by the linear prediction filter. This restriction is the primary reason that LP
coding is almost always used in tandem with other methods in high-compression voice
coders.
RAWCLI vocoder
Robust Advanced Low Complexity Waveform Interpolation (RALCWI) technology
uses proprietary signal decomposition and parameter encoding methods to provide high voice
quality at high compression ratios. The voice quality of RALCWI-class vocoders, as
estimated by independent listeners, is similar to that provided by standard vocoders running
at bit rates above 4000 bit/s. TheMean Opinion Score (MOS) of voice quality for this
Vocoder is about 3.5-3.6. This value was determined by a paired comparison method,
performing listening tests of developed and standard voice Vocoders.
The RALCWI vocoder operates on a “frame-by-frame” basis. The 20ms source voice
frame consists of 160 samples of linear 16-bit PCM sampled at 8 kHz. The Voice Encoder
performs voice analysis at the high time resolution (8 times per frame) and forms a set of
estimated parameters for each voice segment. All of the estimated parameters are quantized to
produce 41-, 48- or 55-bit frames, using vector quantization (VQ) of different types. All of
the vector quantizers were trained on a mixed multi-language voice base, which contains
voice samples in both Eastern and Western languages.
Waveform-Interpolative (WI) vocoder was developed in AT&T Bell
Laboratories around 1995 by W.B. Kleijn, and subsequently a low- complexity version was
developed by AT&T for the DoD secure vocoder competition. Notable enhancements to the
WI coder were made at the University of California, Santa Barbara. AT&T holds the core
patents related to WI, and other institutes hold additional patents. Using these patents as a
part of WI coder implementation requires licensing from all IPR holders.
The product is the result of a co-operation between CML Microcircuits and SPIRIT
DSP. The co-operation combines CML’s 39-year history of developing mixed-signal
semiconductors for professional and leisure communication applications, with SPIRIT’s
experience in embedded voice products.
Regenerative repeater
Introduction of on-board regeneration alleviates the conflict between enhanced traffic
capacity and moderate system cost by reducing the requirements of the radio front-ends, by
simplifying the ground station digital equipment and the satellite communication payload in
TDMA and Satellite-Switched-TDMA systems. Regenerative satellite repeaters can be
introduced in an existing system with only minor changes at the ground stations. In cases
where one repeater can be dedicated to each station a more favorable arrangement of the
information data than in SS-TDMA can be conceived, which eliminates burst transmission
while retaining full interconnectivity among spot-beam areas.
FEED BACK COMMUNICATIONS
Abstract: The concept of generalized decision feedback equalization is introduced.

Nonlinear equalizers operating according to this principle can outperform the decision

feedback equalizer (d.f.e.) when the linear feed forward prefilter does not quite compensate
for channel precursors. Two g.d.f.e. structures are derived from the Kalman filter equalizer .
During the last few years, a number of new equalization algorithms for digital
communication systems have been developed with the scope of achieving higher information
rates over band limited communication channels. 1 Various efforts had already been made to
achieve optimization, in some sense or other, of the parameters which characterize equalizers
with predetermined structures, such as the tapped delay line equalizer (t.d.l.e.). 2 " 7 and the
decision feedback equalizer (d.f.e.). 8 " 20 Successively, experts have focused their attention
on the problem of determining the optimal structure of the equaliser within a determined
class. In this connection, the Kalman filter equalizer ( k.f.e.) 21 "32 and the Viterbi algorithm
equalizer (v.a.e.)* ( 33~ 44 are the best equalizers that can be derived by means of a
statistical state-variable approach to the equalization problem. Thek.f.e. yields an unbiased
channel-state filtered estimate, which is optimal in the minimum mean-square-error (mjn.s.e.)
sense among the linear estimates. The v.a.e. yields the maximum-likelihood estimate of the
whole transmitted sequence when the transmission channel is modeled as a finite-state
machine.
ADVANCEMENTS IN DIGITAL COMMUNICATIONS
Novel Robust, Narrow-band PSK Modes for HF Digital Communications
Some items that I wrote that may be of general interest:
The well-known Shannon-Hartley law tells us that there is an absolute limit on the
error-free bit rate that can be transmitted within a certain bandwidth at a given signal to noise
ratio (SNR). Although it is not obvious, this law can be restated (given here without proof) by
saying that for a given bit rate, one can trade off bandwidth and power. On this basis then, a
certain digital communications system could be either bandwidth limited or power limited,
depending on its design criteria.
Practice also tells us that digital communication systems designed for HF are
necessarily designed with two objectives in mind; slow and robust to allow communications
with weak signals embedded in noise and adjacent channel interference, or fast and somewhat
subject to failing under adverse conditions, however being able to best utilize the HF medium
with good prevailing conditions.
Taken that the average amateur radio transceiver has limited power output, typically
20 - 100 Watts continuous duty, poor or restricted antenna systems, fierce competition for a
free spot on the digital portion of the bands, adjacent channel QRM, QRN, and the marginal
condition of the HF bands, it is evident that for amateur radio, there is a greater need for a
weak signal, spectrally-efficient, robust digital communications mode, rather than another
high speed, wide band communications method.
Recent Developments using PSK on HF
It is difficult to understand that true coherent demodulation of PSK could ever be

achieved in any non-cabled system since random phase changes would introduce
uncontrolled phase ambiguities. Presently, we have the technology to match and track carrier
frequencies exactly, however tracking carrier phase is another matter. As a matter of
practicality thus, we must revert to differentially coherent phase demodulation (DPSK).

Another practical matter concerns that of symbol, or baud rate; conventional RTTY
runs at 45.45 baud (a symbol time of about 22 ms.) This relatively-long symbol time have
been favored as being resistant to HF multipath effects and thus attributed to its robustness.
Symbol rate also plays an important part in determining spectral occupancy. In the case of a
45.45 baud RTTY waveform, the expected spectral occupancy is some 91 Hz for the major
lobe, or +/- 45.45 on each side of each the two data tones. For a two tone FSK signaling
system of continuous-phase frequency-shift keying (CPFSK) paced at 170 Hz, this system
would occupy approximately 261 Hz.
Signal space representation
• Bandpass Signal
• Real valued signal S(f) Ù S* (-f)
• finite bandwidth B Ù infinite time span
• fc denotes center frequency
• Negative Frequencies contain no Additional Info
Characteristics:
• Complex valued signal
• No information loss, truely equivalent
Let us consider DN = {(xi , yi) : i = 1, .., N} iid realizations of the joint observation-class
phenomenon (X(u), Y (u)) with true probability measure PX,Y defined on (X ×Y, σ(FX ×
FY )). In addition, let us consider a family of measurable representation functions D, where
any f(·) ∈ D is defined in X and takes values in Xf . Let us assume that any representation
function f(·) induces an empirical istribution Pˆ Xf ,Y on (Xf ×Y, σ(Ff ×FY )), based on the
training data and an implicit learning approach, where the empirical Bayes classification rule
is given by: gˆf (x) = arg maxy∈Y Pˆ Xf ,Y (x, y).
Turbo codes
In information theory, turbo codes (originally in French Turbocodes) are a class of
high-performance forward error correction (FEC) codes developed in 1993, which were the
first practical codes to closely approach the channel capacity, a theoretical maximum for
the code rate at which reliable communication is still possible given a specific noise level.
Turbo codes are finding use in 3G mobile communications and (deep
space) satellite communications as well as other applications where designers seek to achieve
reliable information transfer over bandwidth- or latency-constrained communication links in
the presence of data-corrupting noise. Turbo codes are nowadays competing with LDPC
codes, which provide similar performance.
Prior to turbo codes, the best constructions were serial concatenated codes based on
an outer Reed-Solomon error correction code combined with an inner Viterbi-decoded short
constraint lengthconvolutional code, also known as RSV codes.
In 1993, turbo codes were introduced by Berrou, Glavieux, and Thitimajshima (from
Télécom Bretagne, former ENST Bretagne, France) in their paper: "Near Shannon Limit
Error-correcting Coding and Decoding: Turbo-codes" published in the Proceedings of IEEE
International Communications Conference.[1] In a later paper, Berrou gave credit to the

"intuition" of "G. Battail, J. Hagenauer and P. Hoeher, who, in the late 80s, highlighted the
interest of probabilistic processing.". He adds "R. Gallager and M. Tanner had already
imagined coding and decoding techniques whose general principles are closely related,"
although the necessary calculations were impractical at that time.[2]
The first class of turbo code was the parallel concatenated convolutional code
(PCCC). Since the introduction of the original parallel turbo codes in 1993, many other
classes of turbo code have been discovered, including serial versions and repeat-accumulate
codes. Iterative Turbo decoding methods have also been applied to more conventional FEC
systems, including Reed-Solomon corrected convolutional codes
There are many different instantiations of turbo codes, using different component
encoders, input/output ratios, interleavers, and puncturing patterns. This example encoder
implementation describes a 'classic' turbo encoder, and demonstrates the general design of
parallel turbo codes.
This encoder implementation sends three sub-blocks of bits. The first sub-block is
the m-bit block of payload data. The second sub-block is n/2 parity bits for the payload data,
computed using a recursive systematic convolutional code (RSC code). The third sub-block
is n/2 parity bits for a known permutation of the payload data, again computed using an RSC
convolutional code. Thus, two redundant but different sub-blocks of parity bits are sent with
the payload. The complete block has m+n bits of data with a code rate of m/(m+n).
The permutation of the payload data is carried out by a device called an interleaver.
Hardware-wise, this turbo-code encoder consists of two identical RSC coders, С 1 and
C2, as depicted in the figure, which are connected to each other using a concatenation scheme,
called parallel concatenation:
In the figure, M is a memory register. The delay line and interleaver force input bits
dk to appear in different sequences. At first iteration, the input sequence dk appears at both
outputs of the encoder,xk and y1k or y2k due to the encoder's systematic nature. If the
encoders C1 and C2 are used respectively in n1 and n2 iterations, their rates are respectively
equal to

The decoder is built in a similar way to the above encoder - two elementary decoders
are interconnected to each other, but in serial way, not in parallel. The decoder
operates on lower speed (i.e. ), thus, it is intended for the encoder, and is
for correspondingly. yields a soft decision which causes delay. The same
delay is caused by the delay line in the encoder. The 's operation causes delay.
An interleaver installed between the two decoders is used here to scatter error bursts
coming from output. DI block is a demultiplexing and insertion module. It works as
a switch, redirecting input bits to at one moment and to at another. In OFF
state, it feeds both and inputs with padding bits (zeros).
Consider a memoryless AWGN channel, and assume that at k-th iteration, the decoder
receives a pair of random variables:
where and are independent noise components having the same variance .
is a k-th bit from encoder output.
Redundant information is demultiplexed and sent through DI to
(when ) and to (when ).
yields a soft decision, i.e.:
and delivers it to . is called the logarithm of the likelihood

ratio (LLR). is the a posteriori probability (APP) of the data bit
which shows the probability of interpreting a received bit as . Taking the LLR into
account, yields a hard decision, i.e. a decoded bit.
It is known that the Viterbi algorithm is unable to calculate APP, thus it cannot be
used in . Instead of that, a modified BCJR algorithm is used. For , the Viterbi
algorithm is an appropriate one.
However, the depicted structure is not an optimal one, because uses only a
proper fraction of the available redundant information. In order to improve the structure, a
feedback loop is used (see the dotted line on the figure).

21. Known gaps
Known gaps:
1. The DC as per the curriculum is not matching with the real time applications
2. This subject is not matching with the coding techniques presently using.
Action to be taken: following additional topics are taken to fill the known gaps
1. Real rime applications

2. Draw backs of the each coding technique
22. Discussion Topics: -Nil-
23. University Question papers of previous years:

R13 Code No: 126AN
JAWAHARLAL NEHRU TECHNOLOGICAL UNIVERSITY HYDERABAD
B. Tech III Year II Semester Examinations, May - 2017
DIGITAL COMMUNICATIONS
Time: 3 hours Max. Marks: 75
Part A is compulsory which carries 25 marks. Answer all questions in Part A. Part B consists of 5
Units. Answer any one full question from each unit. Each question carries 10 marks and may
have a, b, c as sub questions.
PART - A
(25 Marks)
1.a) Compare the performance of PCM and DM system. [2]
b) What is slope overload distortion? Explain. [3]
c) Write the expression for baud rate of BPSK system. [2]
d) Explain advantages of coherent digital modulation schemes. [3]
e) Sketch the wave form of the FSK signal for the input binary sequence 1100100010. [2]
f) Define entropy and conditional entropy. [3]
g) Define code rate of block code. [2]
h) Mention various types of errors caused by noise in communication channel. [3]
i) Define processing gain and jamming margin [2]
j) Explain the generation of PN sequence. [3]
PART - B
(50 Marks)
2.a) A voice frequency signal band limited to 3kHz is transmitted with the use of the DM system.
The pulse repetition frequency is 30,000 pulses per second, and the step size is 40mV. Determine
the permissible speech signal amplitude to avoid slope overload.
b) Derive the expression for overall SNR in a ADM system. [5+5]
OR
3.a) In a binary PCM system, the output signal to quantizing noise ratio is to be held to a
minimum of 40dB. Determine the number of required levels and find the corresponding output
signal to quantization noise ratio.
b) Explain the modulation and demodulation procedure in DPCM system. [5+5]

4.a) Explain frequency shift keying. Describe coherent detection of FSK signals. What should be
the relationship between bit-rate and frequency-shift for a better performance?
b) Explain non coherent detection method of binary frequency shift keying scheme. [5+5]
OR
5.a) Explain coherent detection of PSK signals and derive probability of error.
b) Differentiate coherent and non-coherent detection techniques. [5+5]
6.a) Derive the bit error probability of a coherent ASK signaling scheme.
b) Apply Shannon-Fano coding procedure of M=2 and M=3 [x]=[x 1, x2, x3, x4, x5, x6, x7, x8] with
probability [P]=[1/4, 1/8, 1/16, 1/16, 1/4, 1/16, 1/8 ,1/16]. [5+5]
OR
7.a) Compare code efficiency of Shanon Fano coding and Huffman coding when five source
messages have probabilities m1=0.4, m2=0.15, m3=0.15, m4=0.15, m5=0.15.
b) Obtain the probability of bit error for coherently detected BPSK. [5+5]
8.a) We transmit either a 1 or a 0, and add redundancy by repeating the bit. (i) Show that if we
transmit 11111 or 00000, then 2 errors can be corrected. (ii) Show that in general if we transmit
the same bit 2t+1 times we can correct upto t errors.
b) What are code tree, code trellis and state diagrams for convolution encoders? [5+5]
OR
9.a) Design the encoder for the (7, 4) cyclic code generated by G(p)=p 3+p2+1 and also verify the
operation for any message vector.
b) Derive the steps involved in generation of linear block codes. Define and explain the
properties of syndrome. [6+4]
10.a) Derive the necessity of DSSS techniques. Draw the transmitter and receiver block diagram
and explain.
b) Write a note on CDMA. [6+4]
OR
11.a) Explain the advantages and applications of spread spectrum modulation.
b) Discuss the frequency hopping spread spectrum technique in detail. [4+6]
R13
Code No: 126AN
B.Tech III Year II Semester Examinations, October/November-2016
(Electronics and Communication Engineering)
Note: This question paper contains two parts A and B.

Part A is compulsory which carries 25 marks. Answer all questions in Part A. Part B consists of 5
Units. Answer any one full question from each unit. Each question carries 10 marks and may
have a, b, c as sub questions.
PART - A
(25 Marks)
1.a) Compare PCM and DM. [2]
b) Write the advantages of digital communication. [3]
c) Define QPSK. [2]
d) Draw the block diagram of the PLL. [3]
e) Define Baseband transmission. [2]
f) Define conditional entropy. [3]
g) Mention the properties of cyclic code. [2]
h) Write the advantages of convolution codes. [3]
i) List out the applications of CDMA. [2]
j) Define spread spectrum. List its uses. [3]

PART - B
(50 Marks)
2.a) What is Hartley Shannon law? And explain sampling theorem.

b) With a neat sketch describe ADPCM concept. [5+5]
OR
3.a) Explain the tradeoff between bandwidth and signal to noise ratio.
b) Distinguish between analog communication and digital communication. [5+5]
4.a) Draw and explain the operating principle of ASK Modulator.

b) Describe the BPSK modulation technique with the help of a neat diagram. [5+5]
OR
5.a) Explain the DPSK modulation technique with the help of a neat sketch.
b) Explain the working of non-coherent FSK detector. [5+5]
6.a) Draw and explain the working of optimum receiver with a neat diagram.
b) Define eye diagram. Draw the eye diagram for FSK. [5+5]
OR
7.a) Explain Huffman coding with an example.
b) Explain crosstalk concept. [5+5]
8.a) Describe the algebraic structure of cyclic codes.

b) Explain how to encode cyclic codes. [5+5]
OR
9.a) Give the matrix description for linear block codes.
b) Decode convolution process using viterbi algorithm. [5+5]
10.a) What are the characteristics of PN sequences? Explain

b) Describe the process of code division multiple access in detail. [5+5]
OR
11.a) Describe with a neat sketch the direct sequence Spread spectrum technique.
b) Describe the concept of Ranging using DSSS. [5+5]
Code No: 126AN

B.Tech III Year II Semester Examinations, May - 2016
(Electronics and Communication Engineering)
Note: This question paper contains two parts A and B.

Part A is compulsory which carries 25 marks. Answer all questions in Part A. Part B
consists of 5 Units. Answer any one full question from each unit. Each question carries
10 marks and may have a, b, c as sub questions.
PART - A
(25 Marks)
1.a) What are the drawbacks of delta modulation? [2]
b) Explain the need for non-uniform quantization in digital communication. [3]
c) Draw the Signal space Diagram of ASK. [2]
d) List out the Advantages of Pass band Transmission over Baseband transmission. [3]
e) Define Entropy. [2]
f) Derive the Expression for the Information Rate. [3]
g) Explain in one sentence about (i) Block Size (ii) Linear block codes. [2]

h) List out Properties of Cyclic Codes. [3]
i) Briefly explain about “Spread spectrum.” [2]
j) What is Frequency hopping spread spectrum? [3]
PART – B
(50 Marks)
2.a) With neat block diagram, Explain the process of Sampling and Quantization in digital
communication.
b) Derive the expression for the Quantization error. [5+5]
OR
3.a) Explain about the noise in PCM systems.
b) Write the comparison between PCM and Analog modulation techniques. [5+5]
4.a) With neat diagrams and equations, explain about PSK system.
b) Draw the space representation of BPSK. And also draw its waveforms? [5+5]
OR
5.a) The bit stream 1011100011 is to be transmitted using DPSK. Determine the encoded
sequence and transmitted phase sequence.
b) Explain about DPSK system. And also give the comparison between DPSK and PSK.
[5+5]
6.a) What is the need of pulse shaping for optimum transmission in baseband transmission?
Explain.
b) What is meant by Cross talk? Explain in detail about the causes for cross talk. [5+5]
OR
7.a) Briefly explain about Variable length coding.
b) Explain in detail about Huffman coding and Lossy source code. [5+5]
8.a) Write short notes on Hamming codes.

b) Explain about Error detection and Correction capabilities of Hamming codes. [5+5]
OR
9.a) Explain how Parity checking can be used for error detection or error correction.
b) For a linear block code, prove with example that:
i) The Syndrome depends only on error pattern and not on transmitted code word?
ii) All error patterns that differ by a codeword have the same syndrome? [5+5]
10.a) Explain the role of code division multiple access technique in present generation?
b) Give a brief history about direct sequence spread spectrum. [5+5]
OR
11.a) Explain about PN-Sequences generation and their characteristics.
b) What is meant by Synchronization? Why we require synchronization in spread
spectrum? Explain in detail. [5+5]

Code No: 115BF Set No. 1
III B.Tech. I Sem., I Mid-Term Examinations, August-2015
DIGITAL COMUNICATIONS
Objective Exam
Answer All Questions. All Questions Carry Equal Marks. Time: 20
Min. Marks: 10.
I. Choose the correct alternative:
1. The effect of distortion, noise and interference is less in a digital communication system.
This is because []
a) The disturbance must be large enough to change the pulse from one state to the other.
b) The disturbance must be small enough to change the pulse from one state to the other.
c) The disturbance must be large enough to change the step signal from one state to the other.
d) The disturbance must be small enough to change the step signal from one state to the other.
2. A signal g(t) having the Upper Cutoff frequency, fu = 120KHz and the Lower Cutoff
frequency fl =70KHz. The ratio of upper cutoff frequency to bandwidth of the signal g(t) is [ ]
a) 2.1 b) 2.2 c) 2.3 d) 2.4
3. The quantization process presents an error defined as the difference between the input
signal, x(t) andthe output signal, y(t). This error is called the __________. [ ]
a) Quantization Noise b) Quantization c) Signal to Noise ratio d) Error
4. In delta modulation the step-size is made arbitrarily large to avoid _________________, it

may lead to
_____________. [ ]
a) Slope-Overload Distortion, Granular Noise b) Slope-Overload Distortion, Slope-Overload
Distortionn. c) Granular Noise, Granular Noise d) Granular Noise, Slope-Overload Distortion
5. Aliasing can be overcome by using ________ as antialiasing filter. [ ]

a) BPF b) HPF c) LPF d) APF
6. The spectrum of band pass signal g(t) has bandwidth of 0.6 kHz centered around ±12 kHz.
The Nyquistrate is _________ [ ]
a) 0.3 kHz b) 0.6 kHz c) 1.2 kHz d) 2.4 kHz
7. 18. The signal x(t) =sinc2(200t) the Nyquist interval for x(t) is _______________ [ ]
a) 5 msec b) 2.5 msec c) 1.25 msec d) 0.625 msec
8. In which the carrier used in the receiver is of same frequency and phaseof the transmitted
one is called__________. [ ]
a) Non coherent receivers. b) Coherent receivers. c) ASK d) FSK
9. Probability of error depends on _______________ [ ]

a) Carrier signal power b) Message signal power c) Error signal power d) Signal to noise ratio
10. In BPSK modulation, the modulating signal shifts the phase of the waveform to one of
two states,____________ [ ]
a) 0 b) π c) Either 0 or π d) Neither 0 nor π

II Fill in the Blanks:
11. An input transducer is used for converting the message to a time – varying electrical
quantity called __________
12. Signal processing functions like ________, ___________can be employed to maintain the
secrecy of
the information.
13. The process of transforming sampled amplitude values of a message signal into a discrete
amplitude
value is referred to as _______________.
14. __________ is used for uniform sampling of band passsignals
15. In _______ format binary ‘0’ represents the first half bit duration negative pulse and the
second half
Bit duration positive pulse
16. In coherent detection the local carrier generated at the receiver is phase locked with the
carrier at the
transmitter. Hence it is also called ___________
17. ________________________ consists of tapped delay line filter with set of delay
elements, set
of adjustable multipliers connected to the delay line taps and a summer for adding multiplier
outputs.
18. _____________ is the advantage of non-coherent over coherent systems
19. FSK needs __________ the bandwidth of PSK.
20. Transfer function of the optimum filter is selected to maximize ___________
Code No: 56026 Set No. 1

III B.Tech. II Sem., I Mid-Term Examinations, February – 2014
Objective Exam
Answer All Questions.
All Questions Carry Equal Marks.Time: 20 Min. Marks: 10.
I Choose the correct alternative:
1. The minimum band width required to multiplex 12 different message signals each of band
width 10KHz is [ ]
A) 60KHz B) 120KHz C) 180KHz D) 160KHz
2. In 8-PSK system, adjacent phasors differ by an angle given by ( in degrees) [ ]
A) n/4 B) n/8 C) n/6 D) n/2
3. Band Width efficiency of a Digital Modulation Method is [ ]
A) (Minimum Band width)/ (Transmission Bit Rate)
B) (Power required)/( Minimum Band width)
C) (Transmission Bit rate)/ (Minimum Band width)
D) (Power Saved during transmission)/(Minimum Band width)
4. The Auto-correlation function of White Noise is [ ]
A) Impulse function B) Constant C) Sampling function D) Step function
5. The minimum band width required for a BPSK signal is equal to [ ]
A) one fourth of bit rate B) twice the bit rate C) half of the bit rate D) bit rate

6. Companding results in [ ]
A)More S/N ratio at higher amplitudes of the base band signal
B) More S/N ratio at lower amplitudes of the base band signal
C) Uniform S/N ratio throughout the base band signal
D) Better S/N ratio at lower frequencies
7. A uniform quantizer is having a step size of .05 volts. This quantizer suffers from a
maximum quantization error of [ ]
A) 0.1V B) 0.025 V C) 0.8 V D) 0.05 V
8. In Non-Coherent demodulation, the receiver [ ]
A) relies on carrier phase B) relies on the carrier amplitude
C) makes an error with less probability D) uses a carrier recovery circuit
9. The advantage of Manchester encoding is [ ]
A) less band width requirement B) less bit energy required for transmission
C) less probability of error D) less bit duration
10. Granular Noise in Delta Modulation system can be reduced by
A) using a square law device B) increasing the step size
C) decreasing the step size D) adjusting the rate of rise of the base band signal
II Fill in the blanks

11. Non-coherent detection of FSK signal results in ____________________
12. _____________ is used as a Predictor in a DPCM transmitter.
13. The Nyquist's rate of sampling of an analog signal S(t) for alias free reconstruction is
5000samples/sec. For a signal x(t) = [S(t)]2 ,the corresponding sampling rate in
samples/sec is __________________
14. A Matched filter is used to __________________________
15. A signal extending over -4v to +4v is quantized into 8 levels. The maximum possible
quantization error obtainable is _____________V.
16. The advantage of DPCM over Delta Modulation is _________________________
17. The phases in a QPSK system can be expressed as ______________________
18. The Synchronization is defined as _______________________
19. The sampling rate in Delta Modulation is _______________than PCM.
20. The bit error Probability of BPSK system is __________________that of QPSK.

III B.Tech. II Sem., II Mid-Term Examinations, April – 2015

Objective Exam
Answer All Questions. All Questions Carry Equal Marks.Time: 20 Min. Marks: 10.

1. Information rate of a source is [ ]
A) maximum when the source is continuous B) the entropy of the source measured in
bits/message
C) a measure of the uncertainity of the communication system
D) the entropy of the source measured in bits/sec.
2. The Hamming Weight of the (6,3) Linear Block coded word 101011 [ ]
A) 5 B) 4 C) 2 D) 3
3. Which of the following can be the generating polynomial for a (7,4) systematic Cyclic
code? [ ]
A) x3+x+1 B) x5+x2+1 C) x4+x3+1 D) x7+x4+x3+1
4. In a Linear Block code [ ]
A) the received power varies linearly with that of the transmitted power
B) parity bits of the code word are the linear combination of the message bits
C) the communication channel is a linear system
D) the encoder satisfies super position principle
5. The fundamental limit on the average number of bits/source symbol is [ ]
A) Mutual Information B) Channel capacity
C) Information content of the message D) Entropy of the source
6. A system has a band width of 3KHz and an S/N ratio of 29dB at the input of the receiver. If
the band width of the channel gets doubled, then [ ]
A) its capacity gets halved B) the corresponding S/N ratio gets doubled
C) the corresponding S/N ratio gets halved D) its capacity gets doubled
7. The Channel Matrix of a Noiseless channel [ ]
A) consists of a single nonzero number in each column
B) consists of a single nonzero number in each row
C) is a square Matrix
D) is an Identity Matrix
8. A source emits messages A and B with probability 0.8 and 0.2 respectively. The
redundancy provided by the optimum source-coding scheme for the above Source is
[]
A) 27% B) 72% C) 55% D) 45%
9. A source X and the receiver Y are connected by a noise free channel. Its capacity is [ ]
A) Max H(Y/X) B) Max H(X) C) Max H(X/Y) D) Max H(X,Y)
10. Exchange between Band width and Signal noise ratio can be justified based on [ ]
A) Hartley - Shanon‘s Law B) Shanon‘s source coding Theorem
C) Shanon‘s limit D) Shanon‘s channel coding Theorem
Code No: 56026 :2: Set No. 1

11. The Parity check matrix of a linear block code is
101100

011010
110001
Its Hamming distance is ___________________
12. The significance of PN sequence in CDMA is ________________
13. The cascade of two Binary Symmetric Channels is a __________________________
14. The source coding efficiency can be increased by using _______________________
15. The advantage of Spread Spectrum Modulation schemes over other modulations is
_________________
16. Entropy coding is a _____________________
17. A convolutional encoder of code rate 1/2 is a 3 stage shift register with a message word
length of 6.The code word length obtained from the encoder ( in bits) is
_____________
18. Relative to Hard decision decoding, soft decision decoding results in _____________
19. If T is the code vector and H is the Parity check Matrix of a Linear Block code, then the
code is defined by the set of all code vectors for which HT .T = ______________
20. The advantage of CDMA over Frequency hopping is ____________

KEYS
1. D
2. B
3. A
4. B
5. D
6. C
7. D
8. A
9. B
10. A
11. 3
12. to transmit the information signal using orthogonal codes
13. symmetric Binary channel
14. Source extension
15. It has soft capacity limit
16. Variable length coding scheme
17. 18
18. Better bit error probability
19. ZERO
20. It has soft capacity limit
Code No: 07A5EC09 Set No. 1

III B.Tech. I Sem., II Mid-Term Examinations, November – 2015
Objective Exam

Name: ______________________________ Hall Ticket No.
Answer All Questions. All Questions Carry Equal Marks.Time: 20 Min. Marks: 20.
1) The channel matrix of a noiseless channel [ ]

a)consists of a single nonzero number in each column.
b) consists of a single nonzero number in each row.
c) is an Identity Matrix. d) is a square matrix.
2 ) Information content of a message [ ]
a) increase with its certainty of occurrence. b) independent of the certainty of
occurrence.
c) increases with its uncertainty of occurrence. d) is the logarithm of its uncertainty of
occurrence.
3) The channel capacity of a BSC with the transition probability ½ is [ ]
a) 0 bits b) 1 bit c) 2 bits d) infinity
4) For the data word 1110 in a (7, 4) non-systematic cyclic code with the generator
polynomial 1+x2+x3, the code polynomial is [ ]
a) 1+x+x3+x5 b) 1+x2+x3+x5 c) 1+x2+x3+x4 d) 1+x+x5
5) A source transmitting ‘m’ number of messages is connected to a noise free channel. The
capacity of the channel is [ ]
a) m bits/symbol b) m2 bits/symbol c) log m bits/symbol d) 2m bits/symbol
6) Which of the following is a p(Y/X) matrix for a binary symmetric channel [ ]
a) b) c) d) None
7) Exchange between channel bandwidth and (S/N) ratio can be adjusted based on [ ]
a) Shannon’s limit b) Shannons source coding
c) Shannon’s channel coading d) Shannon Hartley theorem
8) For the data word 1110 in a (7, 4) non-systematic cyclic code with the generator
polynomial 1+x+x3, the code polynomial is [ ]
a) 1+x+x3+x5 b) 1+x2+x3+x5 c) 1+x2+x3+x4 d) 1+x4+x5
9) A source X with entropy 2 bits/message is connected to the receiver Y through a noise free
channel. The conditional probability of the source is H(X/Y) and the joint entropy of
the source and the receiver H(X, Y). Then [ ]
a) H(X,Y)= 2 bits/message b) H(X/Y)= 2 bits/message
c) H(X, Y)= 0 bits/message d) H(X/Y)= 1 bit/message
10) Which of the following is a p(Y/X) matrix for a binary Erasure channel [ ]
a) 11ppqq−⎡⎤⎢⎥−⎣⎦ b) c) d) None
Cont…..2

Code No: 07A5EC09 :2: Set No.1
II Fill in the blanks:
11) The information rate of a source is also referred to as entropy measured in
______________
12) H(X,Y)=______________ or __________________
13) Capacity of a noise free channel is _________________
14) The Shannon’s limit is ______________
15) The channel capacity with infinite bandwidth is not because ____________
16) Assuming 26 characters are equally likely , the average of the information content of
English language in bits/character is________________
17) The distance between two vector c1 and c2 is defined as the no.of components in which
they are differ is called as____________________
18) The minimum distance of a linear block code is equal to____________________of any
non-zero code word in the code.
19) A linear block code with a minimum distance dmin can detect upto ___________________
20) For a Linear Block code Code rate =_________________

24. References, Journals, websites and E-links:
TEXT BOOKS
1. Principles Of Communication Systems-Herberet Taub, Donald L Schiling, Goutham

saha,3rd edition, Mc Graw Hill 2008.
2. Digital and anolog communiation systems- Sam Shanmugam, John Wiley,2005.
3. Digital communications- John g. Prokaris, Masoud salehi-5th edition Mc Graw-Hill,
2008.
4. Digital communications- Simon Haykin, Jon Wiley, 2005
Websites:-
1. http://en.wikipedia.org/wiki/digital_communications
2. http://www.tmworld.com/archive/2011/20110801.php
3. www.pemuk.com
4. www.site.uottawa.com
5. www.tews.elektronik.com
Journals:-
1. Communicaions Journal
2. Omega online technical reference
3. Review of Scientific techniques
REFERNCES:
1. Digital communications- John g. Prokaris, Masoud salehi-5th edition Mc Graw-Hill,
2008.
2. Digital communication- Simon Haykin, Jon Wiley, 2005.
3. Digital communications-Lan A.Glover, Peter M.Grant.2nd edition, pearson edu., 2008.
4. Communication systems-B.P.Lathi, BS Publication, 2006.

25. Quality Control Sheets. a)course end survey b)Teaching Evaluation
c) CO- attainment

26. Student List:
GEETHANJALI COLLEGE OF ENGINEERING & ENGINEERING

Cheeryal (V), Keesara (M). Medchal Dist, TS 501 301
AUTONOMOUS
Student Nominal Roll
NO:Admin/B.Tech/SR/23 Rev No:
Academic Year 2018-19 Date: 28.11.2018
Class & Branch : B.Tech (ECE) III Year II sem Section: A Batch : 2016
SlN
o AdmnNo Student Name SlNo AdmnNo Student Name
1 15R11A0424 KONDAPALLI RAVI TEJA 29 16R11A0431 SISTA B S S SAI PHANEENDRA
2 15R11A04L9 M VISHAL 30 16R11A0432 SUBBAGARI SHREEDHA
3 16R11A0401 ANOOHYA ANAMANDALA 31 16R11A0434 SUMANTH PAL
4 16R11A0402 B AKSHITHA 32 16R11A0435 VANGARI DEEPTHY
5 16R11A0403 BANGARE VISHAL 33 16R11A0436 V ANIL KUMAR
6 16R11A0404 BAPATIA KOMALA SHIVANI 34 16R11A0437 VADAYALA NITHYA
7 16R11A0406 CHEPURI LOKNATH 35 16R11A0438 VAISHNAVI VARMA ALLURI
8 16R11A0407 CHILAKA SAITEJA 36 16R11A0439 VIRUPAKA SOWMYA
9 16R11A0408 CHILAMKURTHI ANUHYA 37 16R11A0441 VORUGANTI PRANEETH
10 16R11A0409 CHINTHALA SUDEEKSHA 38 16R11A0442 V A S SAI ANISH KUMAR
11 16R11A0410 DEVOJU VIJAYALAKSHMI 39 16R11A0443 RAMIDI SARVINDH REDDY
12 16R11A0411 DIBBA SHARON LILLY 40 16R11A0444 NARA ANJALI
13 16R11A0413 DOMMATA VAISHNAVI 41 16R11A0445 P PRANEETH SAI KUMAR REDDY
14 16R11A0414 G S S R L PRASUNA 42 16R11A0446 KHALED ALI AHMED
15 16R11A0415 GUDAVALLI NAGA SATYA SAI 43 16R11A0447 KOTAKONDA VINAY KUMAR
16 16R11A0416 GUNDABATHINA MOUNIKA 44 16U51A0476 GADI SAI POOJA
17 16R11A0417 JAMPANA SREE SAI PRASANNA 45 17R15A0401 KATTA PAVAN KALYAN
18 16R11A0420 KALSHAM GREESHMA 46 17R15A0402 ERELLI POOJA
19 16R11A0421 KATAKAM SAI DIKSHITH 47 17R15A0404 POTTA PAVANI
20 16R11A0422 V SAI SHUBHANG 48 17R15A0405 VADLAKONDA BHANU PRAKASH
21 16R11A0423 KORADA MAHATHI 49 17R15A0406 PENTABOINA SAIKUMAR
22 16R11A0424 KOTHURI SAVITHA VAISHNAVI 50 17R15A0407 BANOTH GANGA
23 16R11A0425 M PRATYUSHA 51 17R15A0408 MAROJU VAMSHI KRISHNA
24 16R11A0426 M VINEETH RAJ 52 17R15A0409 K RAJASHEKAR
25 16R11A0427 MAJETI ESWAR HEMANT 53 17R15A0410 BADHINENI VENKAT RAVALI
26 16R11A0428 PAKA SNEHITHA 54 17R15A0411 CHINDAM PRASHANTH
27 16R11A0429 POTUGANTI ANIL KUMAR 55 17R15A0412 PASULA ANUSHA
28 16R11A0430 RAMESH SUTHAR 56 17UP5A0401 INDRAVATH SANGEETHA
Total: 56 Males: 28 Females: 28
15R11A04L4 KARANAM ARAVIND KUMAR

DUE TO SHORTAGE OF ATTENDANCE
16R11A0440 VISLAVATH BHASKER
DEAN-ADMIN

AUTONOMOUS
Class & Branch : B.Tech (ECE) III Year II sem Section: B Batch : 2016
SlN
1 15R11A04C0 YELIGETI NAVYA SREE 32 16R11A0478 RAAGI SHRAVAN
2 15R11A04J4 BHUKYA RAJESH 33 16R11A0479 RAMASAGARAM ANURADHA
3 15R11A04M1 MADDIKATLA RAVI PRAKASH 34 16R11A0480 MANNE HARINI
4 15R11A04M4 M SAI KUMAR 35 16R11A0481 S SAI HARSHITHA
5 16R11A0449 A SAI SHASHANK 36 16R11A0482 SHIVA VENKAT SAI RAM
6 16R11A0450 ALESHIA A 37 16R11A0483 SAIKAM MOKSHA TEJA
7 16R11A0451 BANDARU TULASI 38 16R11A0484 SHAIK ABDUL REHMAN
8 16R11A0452 BACHU SUHAS 39 16R11A0485 SUBRAT MOHANTY
9 16R11A0453 BONAGIRI BINDUSAI 40 16R11A0486 T LOHIT KUMAR
10 16R11A0455 DEME AISHWARYA 41 16R11A0487 THOTA SAI SUNNY
11 16R11A0456 DONTULA VINEETH 42 16R11A0488 VADLA VISHWA PRAKASH CHARY
12 16R11A0457 GANKIDI SHIVA REDDY 43 16R11A0489 VALMIKI ASHWINI
13 16R11A0458 GUNDLA PAVANI 44 16R11A0490 VATTIGUNTA VAMSI
14 16R11A0459 JITENDRA SINGH TOMAR 45 16R11A0491 VELDHURTHI BHARATH CHARY
15 16R11A0460 K ATHUL TEJA 46 16R11A0492 VEMULA SNIGDHA
16 16R11A0461 K NAVEEN KUMAR 47 16R11A0493 VEMULA UDAY SAI
YELAMANCHI S V G MANIDWEEP
17 16R11A0462 KACHIRAJU HIMESH BHARADWAJ 48 16R11A0494 SHARMA
18 16R11A0463 KANNE TARANI 49 16R11A0495 VANKUDOTH RAHUL NAIK
19 16R11A0464 KANUKANTI SAI DHANRAJ 50 16R11A0496 MEDISETTI BHARGAV RISHI
20 16R11A0465 KONDURU PRANAV BHASKAR 51 16R15A0413 KASUKURTHI SAIKUMAR
21 16R11A0466 KUSULURI SWAPNA 52 16R15A0417 GATPA RAJU
22 16R11A0467 M SREEJA REDDY 53 17R15A0413 VEMULAPALLY RAKESH
23 16R11A0468 M SWATHI 54 17R15A0415 RAMIDI PRASANNA LAKSHMI
24 16R11A0469 MINAL HIRA 55 17R15A0416 KULKERNI VASANTHA
25 16R11A0470 MOHAMMED KHALED PASHA 56 17R15A0417 VULLENGALA NIHARIKA
26 16R11A0471 MUNAGALA VENKATA SRAVYA SREE 57 17R15A0418 D MEGHNA
27 16R11A0472 NEMALI BODHENDRA REDDY 58 17R15A0419 KANCHARLA KUSUMA SRI
28 16R11A0473 ADARSH REDDY PASULA 59 17R15A0420 GUDURU DURGAPRASAD REDDY
29 16R11A0474 P SURYA TEJA 60 17R15A0421 MANTHRI SWAMY
30 16R11A0475 PALAPARTHI RAJEEV 61 17R15A0422 BOLLA ROHITH
31 16R11A0476 PALUVAYI NIHAR SRITEJA

DEAN-ADMIN

AUTONOMOUS
Class & Branch : B.Tech (ECE) III Year II sem Section: C Batch : 2016
SlN
1 15R11A0416 JALDA VIGNESH KUMAR 30 16R11A04C1 L SHRAVAN
2 15R11A0438 NALLURI SURYA VENKATA AVINASH 31 16R11A04C2 MANCHIKATLA HARIPRASAD
3 15R11A0443 OGIRALA AJAY KUMAR 32 16R11A04C3 MANDA TEJASWI REDDY
4 15R11A0469 CH NEERAJ PAUL 33 16R11A04C4 MANDULA SANJAY
5 15R11A0478 JAGANNADHAM BHARGAV ROSHAN 34 16R11A04C5 NALLA AKHIL
6 15R11A04A2 POTHURI SUMANTH 35 16R11A04C6 NALLOLA ABHINAV
7 16R11A0497 ADUSUMILLI KAUSHIK 36 16R11A04C7 NAVEEN SINGH
8 16R11A0498 ALLADURGAM BADHRINADH 37 16R11A04C8 P S SAI TEJA REDDY
9 16R11A0499 SAI RAHUL ANAMALLI 38 16R11A04C9 MANGIPUDI RAJA VAMSHI KRISHNA
10 16R11A04A0 ANUMULA SUCHITHA 39 16R11A04D0 SIKHARAM SAI GURU KIRAN
11 16R11A04A1 ANUMULA VARSHA 40 16R11A04D1 SRI NITHYA ANNE
12 16R11A04A2 AREPATY VAMSHI KRISHNA 41 16R11A04D2 ADAPALA SRINIJA
13 16R11A04A3 B VENKAT PRATEEK 42 16R11A04D3 TANNERU HARISH
14 16R11A04A5 BURUGU SHANMUKA VAMSHI 43 16R11A04D4 TUMMEDA AJAY NAVEEN
15 16R11A04A6 C RAJESH 44 16R11A04D5 VALLEPU VENU KUMAR
16 16R11A04A7 CHINTAPALLI SATYAKIRAN 45 16R11A04D6 VAMSHI KRISHNA PATHI
VEMURI HANUMANTHA RAGHU
17 16R11A04A8 DEVALLA VIJAY KUMAR 46 16R11A04D7 SRIKAR
18 16R11A04A9 E NAVEEN KUMAR 47 16R11A04D8 YANAKANDLA MANASA
19 16R11A04B0 ERRA PEDDA NAGAIAH 48 16R11A04D9 RONDLA SOWMYA
20 16R11A04B1 GAJAVELLY GOWTHAM 49 16R11A04E0 VALLETI CHANDANA
21 16R11A04B2 GANAM VAMSHIKRISHNA 50 16R11A04E1 ASHISH B CHAKRAVARTHI KATTA
22 16R11A04B3 H VARUN 51 16R11A04E2 KONDURI HIMASRI
23 16R11A04B4 J DIVYA 52 16R11A04E3 A V S RAGHUVEER
24 16R11A04B5 JAYANTI KARTHIK 53 16R11A04E4 MULEM PRANEETHA
25 16R11A04B6 KALAKUNTLA RAJU 54 17R15A0423 VITTA SANDEEP REDDY
26 16R11A04B7 KALAM VENKAT REDDY 55 17R15A0424 REGU BHASKAR
27 16R11A04B8 KODAVATIKANTI PRASAD 56 17R15A0425 MARABOINA RAJASRI
28 16R11A04B9 KORADA HARIKA 57 17R15A0429 NARRA MOULIKA
29 16R11A04C0 KULAKARNI KARTHIK 58 17R15A0430 NANDIGAMA VINAY KUMAR
DEAN-ADMIN

AUTONOMOUS
Class & Branch : B.Tech (ECE) III Year II sem Section: D Batch : 2016
SlN
1 15R11A0423 KOLLI SAI KAMAL TEJ 29 16R11A04G9 NALLABELLI SAHITHI
2 15R11A0436 MANISH GOWDE 30 16R11A04H0 NANDURI SAI SWETHA
3 15R11A04B5 UPPALURI VENKATA KRISHNA SHRAVAN KU 31 16R11A04H1 POORNIMA SHASTRY
4 15R11A04C6 BIJJA KIRTHI SAI 32 16R11A04H2 PRODUTURI BHAGYA RAJ
5 166M1A0430 B SAI BHAVANI 33 16R11A04H3 S PRANAY
6 16R11A04E5 ANDABATLA DHEERAJ 34 16R11A04H4 SAIRI MANASA
7 16R11A04E6 ANNAM BHAVANA 35 16R11A04H5 SANKURI NAG NISHITHA
8 16R11A04E7 APPIKATLA DIVYA SRI 36 16R11A04H6 SHAIK NAZIMA BEGUM
9 16R11A04E8 ASHADAPU PAVAN 37 16R11A04H7 SINGIREDDY REVANTH REDDY
10 16R11A04E9 B DIVYA RANI 38 16R11A04H8 SOMMARAPU SAIKUMAR
11 16R11A04F0 B V KAUSHIK 39 16R11A04J0 SAI HRUSHIKESH REDDY T
12 16R11A04F1 BAJJURI SMRUTHI 40 16R11A04J1 U NARASIMHA
13 16R11A04F2 CHAVALI V P S P S RAVI SHASHANK 41 16R11A04J2 V AKHILA
14 16R11A04F3 CHAKKA MOUNIKA 42 16R11A04J3 V PRITHVI
VADDI VENKATA BHIMA SHANKAR
15 16R11A04F4 CHIRRA SRUTHI 43 16R11A04J4 RAO
16 16R11A04F5 D CHAKSHU SAI KUMAR 44 16R11A04J5 VADICHERLA RAVITEJA
17 16R11A04F6 DENDE RAGHUVARAN 45 16R11A04J7 VIKURTHI SANDHYA RANI
18 16R11A04F7 GOGULA PRUTHVI REDDY 46 16R11A04J8 YAGATI VAMSI
19 16R11A04F9 K MADHUSUDHAN REDDY 47 16R11A04J9 VANCHA PRUTHVIKA
20 16R11A04G0 KUKALA RAMYA SREE 48 16R11A04K0 DARSHANI PARALIKAR
21 16R11A04G1 KANDAVALLY SHASHI KUMAR 49 16R11A04K1 SHAIK SHABNAM
22 16R11A04G2 KOTHA NAVYA REDDY 50 16R11A04K2 YENUGULA YOCHANA
23 16R11A04G3 NAVEEN KUMAR YADAV 51 16R15A0403 KOLLURU SAI RAM
24 16R11A04G4 KUNDENA SHASHANK 52 17R15A0431 DASAMANTHULA JAI SAI KISHORE
25 16R11A04G5 KUPPIREDDY VARUN REDDY 53 17R15A0432 CHUNCHU SWATHI
26 16R11A04G6 M PRAVALLIKA 54 17R15A0433 DUBBAKA KISHORE
27 16R11A04G7 G S SRAVANI 55 17R15A0434 KAMBALA SUJATHA
28 16R11A04G8 MATHERA SWARNA LATHA 56 17R15A0435 BYAGARI NITISH
DEAN-ADMIN

AUTONOMOUS
Class & Branch : B.Tech (ECE) III Year II sem Section: E Batch : 2016
SlN
1 15R11A0464 BAIRABOINA REVANTH RAJ 30 16R11A04N1 MANTRALA AGASTYA VENKATA SRI HARSHA
2 16R11A04J6 VENKATAPURAM AVINASH 31 16R11A04N2 MARPADGA SHRAVYA
3 16R11A04K3 ANUGU MANIKANTA REDDY 32 16R11A04N3 MEDABALIMI CHANDRA SHEKHAR
4 16R11A04K4 BEEREDDY BALA SNEHITH REDDY 33 16R11A04N4 MOHAMMED FAZIL
5 16R11A04K5 BOJJA VARSHA 34 16R11A04N5 P SRIKAR REDDY
6 16R11A04K6 CHIPPA AKANKSHA 35 16R11A04N6 P V S DATTA ANIRUDH
7 16R11A04K7 CHITTALA DIVYA SREE 36 16R11A04N7 POTHULA ARAVIND KUMAR
8 16R11A04K8 DASARI VINAY REDDY 37 16R11A04N8 PRANEET KUMAR JAIN
9 16R11A04K9 DEVERSHETTY PRIYANKA 38 16R11A04N9 RASHMI TADAS
10 16R11A04L1 E ARUN REDDY 39 16R11A04P0 SALAGRAMA VISHNU TEJA
11 16R11A04L2 GADDAM SUMANJALI 40 16R11A04P1 TALLURI VENKAT KUMAR SWAMY
12 16R11A04L3 GURUMANCHI SAI KALYAN 41 16R11A04P2 TUMATI SONIYA
13 16R11A04L4 INDOORI LAKSHMI KANTH 42 16R11A04P3 VALLE BHUVANA KRUTHI
14 16R11A04L5 JATIN EKBOTE 43 16R11A04P4 KOMMA ARTHI
15 16R11A04L6 JILLABOINA PRAVEEN 44 16R11A04P6 KONDURU DISHHIRAA
16 16R11A04L7 KONDURI ANIRUDH 45 16R11A04P7 VATTEM LALITA YAMINI
17 16R11A04L8 K AVINASH MANOHAR 46 16R11A04P8 RAYALA NAVYA
18 16R11A04L9 KANAPARTHI HARIDRASHWA 47 16R11A04P9 TAMADA SRIDEVI
19 16R11A04M0 KETHOJI SAMRITA 48 17R15A0437 BHOGA ARUNA KUMARI
20 16R11A04M1 KODIMALA PRADEEP REDDY 49 17R15A0438 CHINTHALA SANTHOSH KUMAR
KOTHAPETA VUMMARASETTY
21 16R11A04M2 MADHULIKA 50 17R15A0439 DONTHA PRASHANTH
22 16R11A04M3 M V NARESH 51 17R15A0440 DADIGE SOUJANYA
23 16R11A04M4 B ANJANEYULU 52 17R15A0442 PEGADAPALLI SRIKANTH
24 16R11A04M5 BOBBA SUMANTH REDDY 53 17R15A0443 R SAI SUMANTH
25 16R11A04M6 CHAKATI PRAVEEN KUMAR 54 17R15A0444 S KIRAN KUMAR
26 16R11A04M7 MADDERLA SAI PRAKASH 55 17R15A0445 THOOM HARSHAVARDHAN
27 16R11A04M8 MADDI HARSHA 56 17R15A0446 SWARGAM MADHAN
28 16R11A04M9 MADHAVARAM YESHAWANTH 57 17R15A0447 BARLA SUNIL KUMAR
29 16R11A04N0 MALOTH SAGAR 58 17R15A0448 CHANDANA MAMIDALA

DEAN-ADMIN
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Digital

ECE DEPARTMENT DIGITAL COMMUNICATIONS

Courses file contents:

ECE DEPARTMENT DIGITAL COMMUNICATIONS

Name of the Subject : Digital Communications

Classification status (Unrestricted / Restricted ) : Unrestricted

Distribution List : Dept. Library, Concerned Faculty

1) Name : Ms.S. Krishna Priya 1) Name : D.Venkata Rami Reddy

3) Design : Assoc. Prof. 3) Design : Assoc. Prof.

4) Date : 02-12-2017 4) Date : 20 – 12 - 2018

Verified by : For Q.C Only.

Approved by : Head of the Department

1) Name : Prof. B. Hari Kumar

ECE DEPARTMENT DIGITAL COMMUNICATIONS

Geethanjali visualizes dissemination of knowledge and skills to students, who would

3. Mission of the Institution

4. Vision of the Department

5. Mission of the Department

i. To impart quality education in fundamentals of basic sciences, mathematics, electronics

ECE DEPARTMENT DIGITAL COMMUNICATIONS

6.1 Program Educational Objectives:

I. To prepare students with excellent comprehension of basic sciences, mathematics

III. To inculcate positive attitude, professional ethics, effective communication and

6.2 Program Outcomes of B.Tech ECE Program: (POs)

ECE DEPARTMENT DIGITAL COMMUNICATIONS

Program Specific Outcomes (PSOs):

1. An ability to design an Electronics and Communication Engineering system, component,

ECE DEPARTMENT DIGITAL COMMUNICATIONS

ECE DEPARTMENT DIGITAL COMMUNICATIONS

Develop ability to:

After the completion of the course, students will be able to:

9. Brief note on the course & how it fits in to the curriculum:

The move to digital modulation provides more information capacity, compatibility

ECE DEPARTMENT DIGITAL COMMUNICATIONS

11. Instructional learning outcomes:

Subject: Digital Communications

Elements of Digital Communication Systems

DC3. Advantages of digital communication systems over analog systems.

Base Band Transmission and Optimal Digital Signal

Pulse Coded Modulation

DC7: Explain the generation and reconstruction of PCM.

DC8: Analyze the effect of Quantization noise in Digital Communication.

DC 11: Illustrate the effect of Noise in PCM and DM systems.

ECE DEPARTMENT DIGITAL COMMUNICATIONS

Information Theory and Source coding techniques:

Linear Block Codes

DC23: Describe Encoding & decoding of Convolutional Codes.

DC25: Solve problems based on Viterbi algorithm.

ECE DEPARTMENT DIGITAL COMMUNICATIONS

Digital Modulation Techniques

DC33: Analyze the need and use of PLL in FSK Detection.

Spread Spectrum Modulation

DC38: Generate PN sequences and solve problems based on sequence generation.

DC39: Explain the need of synchronization in spread spectrum system.

DC40: Identify the Advancements in the digital communication.

12.Course mapping with PEOs and POs:

Course code course Semester PEO PEO PEO

ECE DEPARTMENT DIGITAL COMMUNICATIONS

CO1: explain the basic 1 2 3 2 2 2 1 2 2 3

CO2: analyze the concepts of 1 2 3 2 2 2 1 2 2 3

CO 3: develop linear block 1 2 3 2 2 2 1 2 2 3

CO 4: compare different pass 1 2 3 2 2 2 1 2 2 3

CO 5: describe the different 1 2 3 2 2 2 1 2 2 3

ECE DEPARTMENT DIGITAL COMMUNICATIONS