DSP PDF

COURSE FILE
Subject (Name) : Digital signal processing
Name (of the Faculty Member) : P.USHA
Designation : Asst. Prof.

Regulation /Course Code : R 16/ EE721PE
Year / Semester : IV/ I
Department : ECE
Academic Year : 2019-20
DEPARTMENT OF
ELECTRICAL AND ELECTRONICS
ENGINEERING
R-16
DIGITAL SIGNAL PROCESSING
IV B. Tech
2019-20
Course File Prepared by
P.USHA
Assistant Professor
DIGITAL SIGNAL PROCESSING(EE721PE)
3) Course Objectives, Course Outcomes and Topic Outcomes
a) Course Objectives
1. To provide background and fundamental material for the analysis and processing of digital
signals.
2. To familiarize the relationships between continuous time and discrete time signals and systems.
3. To study fundamentals of time, frequency and Z-plane analysis and to discuss the inter-
relationships of these analytic method.
4. To study the designs and structures of digital (IIR and FIR) filters from analysis to synthesis for a
given specifications.
5. The impetus is to introduce a few real-world signal processing applications.
6. To acquaint in FFT algorithms, Multi-rate signal processing technique and finite word length
effects.
b) Course Outcomes
1. Construct time, frequency and Z -transform analysis on signals and systems.

2. Compare the inter-relationship between DFT and various transforms.
3. Describe the significance of various filter structures.
4. Design a digital filter for a given specification.
5. Identify the tradeoffs between normal and multi rate DSP techniques and finite length word effects.
P.USHA, Assistant. Professor,dept of ECE,KGRCET

c) Topic outcomes
S.No Topic Topic outcome At the end of the topic the
student will be able to
1 UNIT-1 Introduction to DSP Explain a digital processing system.
2 Discrete time signals and sequences Compare discrete time signal and
sequence.
3 Linear shift invariant systems Describe a LTI system.
4 problems
5 Stability and Casuality Describe stability and causality of
system.
6 Linear constant coefficient difference Solve the difference equations.
equation
7 Frequency domain representation of DTS Demonstrate the frequency domain
representation of DTS.
8 Application of Z transform List the applications of Z transform.
9 Solution of difference equation of digital Solve the difference equation.
filters
10 System function, Explain stability criterion and its
frequency response.
11 Stability criterion Explain stability criterion and its
frequency response.
12 Frequency response of stable systems Explain stability criterion and its
frequency response.
13 Realization of digital filters-direct and
canonic form Design a digital filter using different
methods.
14 Cascade and parallel form Design a digital filter using different
methods.
15 problems Summarize the important concepts of
16 Revision, Discussion of previous question this unit.
papers
17 UNIT II-Discrete Fourier Series: DFS Describe the representation of DFS.
representation
18 Properties of DFS list the properties of DFS
19 DFT and Properties of DFT list the properties of DFT
20 Linear convolution using DFT Solve linear convolution using
different methods.
21 Overlap Add method Solve linear convolution using different
methods.
22 Overlap Save method Solve linear convolution using different
methods.
23 Relation between DTFT and DFS Analyze the relation between different
transforms.

24 Relation between DFT and Z transform Analyze the relation between different
transforms.
25 FFT Explain FFT.
26 Radix-2 decimation in time FFT algorithm Select an appropriate algorithm to
perform FFT.
27 Radix-2 decimation in frequency Select an appropriate algorithm to
perform FFT.
28 Inverse FFT describe Inverse FFT
29 FFT with general radix N describe FFT with general radix N.
30 Problems on overlap add method and Solve the problems on overla add and
save method. save method
31 Problems on DIT algorithm Solve the DIT algorithm
32 Problems on DIF algorithm Solve the DIF algorithm
33 Revision, Discussion of previous question revision
papers
34 UNIT III- IIR Digital Filters Explain IIR digital filters.
35 Analog filter approximations List the steps to design analog filter.
36 Butterworth filter Compare Butterworth and Chebyshev
37 Chebyshev filters filters.
38 Design of IIR filters from Analog filters Design IIR filters using different
techniques.
39 Step Technique Design IIR filters using different
techniques.
40 Impulse invariant technique Design IIR filters using different
techniques.
41 Bilinear transformation method Design IIR filters using different
techniques.
42 Spectral transformation Design IIR filters using different
techniques.
43 Conversion of low pass to other filters Convert IIR filters from low pass
designing.
44 Problems on approximations of different Solve the problems and summarize the
filters. important concepts of this unit.
45 Revision
46 Discussion of previous question papers
47 UNIT IV-FIR Digital filters Explain characteristics of FIR filters
48 Frequency response and its frequency response.
49 Design of FIR filters using fourier method Design FIR filters using various
techniques.
50 Digital filters using window technique: Deisgn FIR filter using various
Frequency sampling technique techniques.
51 Comparison of FIR and IIR filters Compare FIR and IIR filters.
52 Problems on Design of FIR filter Summarize the important concepts of
53 Revision this unit.
54 Discussion of previous question papers
55 UNIT V-Multirate digital signal processing Explain multirate DSP and its
applications.

56 Downsampling Compare up sampling and down
sampling.
57 Decimation Compare up sampling and down
sampling.
58 Upsampling Compare up sampling and down
sampling.
59 Interpolation Compare up sampling and down
sampling.
60 Sampling rate conversion Explain sampling rate conversion.
61 Finite word length effects Discuss different finite word length
effects.
62 Limit cycle Discuss different finite word length
effects.
63 Overflow oscillations Discuss different finite word length
effects.
64 Round off noise Describe round off noise.
65 Computational output round off noise
66 Methods to prevent overflow List different methods to prevent
overflow.
67 Trade off between round off and overflow Discuss different finite word length
noise effects.
68 Dead band effects Explain dead band effects.
69 Revision Summarize the important concepts of
70 Discussion of previous question papers this unit.

4) COURSE PRE–REQUISITES
1. Basics of mathematics
2. Different types of signals and systems
3. Difference between continuous and discrete signals

5) CO’s, PO’s mapping
CO PO1 PO2 PO3 PO4 PO5 PO6 PO7 PO8 PO9 PO10 PO11 PO12
CO1 3 3 1 3 3
CO2 3 3 1 3 3
CO3 3 3 3
CO4 3 3 3
CO5 2 2 1 2 1
Legends: 1 – Low
2 – Medium
3 – High

5.COURSE INFORMATION SHEET

5.a). COURSE DESCRIPTION:
PROGRAMME: B. Tech. (Electrical and Electronics DEGREE: BTECH

Engineering.)
COURSE: DIGITAL SIGNAL PROCESSING YEAR: IV SEM: I CREDITS: 4
COURSE CODE: EE721PE COURSE TYPE: ELECTIVE
REGULATION: R16
COURSE AREA/DOMAIN: Signal processing CONTACT HOURS: 4+1 (L+T)) hours/Week.
CORRESPONDING LAB COURSE CODE (IF ANY):NA LAB COURSE NAME: NA
5.b). SYLLABUS:
Unit Details Hours

Introduction: Introduction to Digital Signal Processing: Discrete Time signals &
Sequences, Linear Shift Invariant Systems, Stability, and Causality, Linear Constant
Coefficient Difference Equations, Frequency Domain Representation of Discrete Time Signals
and Systems
I 13
Realization of Digital Filters: Applications of Z -Transforms, Solution Difference Equations
of Digital Filters, System Function, Stability Criterion, frequency Response of Stable Systems,
Realisation of Digital Filters-Direct,Canonic,Cascade and Parallel Forms.
Discrete Fourier series: DFS Representation of Periodic Sequences, properties of Discrete Fourier
Series, Discrete Fourier Transforms: Properties of DFT, Linear Convolution of Sequences using
DFT, Computation of DFT: Over-Lap Add Method, Over-Lap Save Method, Relation
II between DTFT, DFS, DFT and Z-Transform. 17
Fast Fourier Transforms: Fast Fourier Transforms (FFT) – Radix-2 Decimation-in-Time and
Decimation-in-Frequency FFT Algorithms, Inverse FFT, and FFT with General Radix-N.
IIR Digital Filters: Analog filter approximations – Butter worth and Chebyshev, Design of IIR
III Digital Filters from Analog Filters, Step and Impulse Invariant Techniques, Bilinear 10
Transformation Method, Spectral Transformations.
FIR Digital Filters: Characteristics of FIR Digital Filters, Frequency Response, Design of FIR
IV Filters: Fourier Method, Digital Filters using Window Techniques, Frequency Sampling Technique, 08
Comparison of IIR & FIR filters
Multi rate Digital Signal Processing: Introduction, Down Sampling Decimation, Up sampling,
Interpolation, Sampling Rate Conversion.Finite Word Length Effects: Limit cycles, Overflow
V Oscillations, Round-oft Noise in IIR Digital Filters, Computational Output Round Off Noise, 16
Methods to Prevent Overflow, Trade Off Between Round Off and Overflow Noise, Dead Band
Effects.
Contact classes for syllabus coverage 64
Lectures beyond syllabus 02
Tutorial classes 00
Classes for gaps&Add-on classes 02
Total No. of classes 68
5.c). GAPS IN THE SYLLABUS - TO MEET INDUSTRY/PROFESSION REQUIREMENTS:
S.NO. DESCRIPTION No. Of Classes
1 Discrete time fourier transform 1
2 Applications of digital filters 1
5.d). TOPICS BEYOND SYLLABUS/ADVANCED TOPICS:
S.NO. DESCRIPTION No. Of Classes
1 Architecture of TMS320C5X processor 1
2 Wavelet Transforms 1
5. e). WEB SOURCE REFERENCES:

Sl. No. Name of book/ website
a. http://nptel.ac.in/courses/117104070/
b. http://nptel.ac.in/courses/117104070/9
c. http://nptel.ac.in/courses/117104070/6
d. http://nptel.ac.in/courses/117102060/39
5. f). DELIVERY/INSTRUCTIONAL METHODOLOGIES:
 CHALK & TALK  STUD. ASSIGNMENT  WEB RESOURCES
 LCD/SMART BOARDS  STUD. SEMINARS ☐ ADD-ON COURSES
5.g). ASSESSMENT METHODOLOGIES-DIRECT

 ASSIGNMENTS  STUD.  TESTS/MODEL  UNIV.
SEMINARS EXAMS EXAMINATION
☐ STUD. LAB ☐ STUD. VIVA ☐ MINI/MAJOR ☐

PRACTICES PROJECTS CERTIFICATIONS
☐ ADD-ON ☐ OTHERS
COURSES
5.h). ASSESSMENT METHODOLOGIES-INDIRECT
 ASSESSMENT OF COURSE OUTCOMES  STUDENT FEEDBACK ON

(BY FEEDBACK, ONCE) FACULTY (TWICE)
☐ASSESSMENT OF MINI/MAJOR PROJECTS ☐ OTHERS
BY EXT. EXPERTS
5.i). TEXT/REFERENCE BOOKS:
T/R BOOK TITLE/AUTHORS/PUBLICATION

Text Book Digital Signal Processing, Principles, Algorithms, and Applications John G.
Proakis, Dimitris G. Manolakis, Pearson Education / PHI, 2007.
Text Book Discrete Time Signal Processing — A. V. Oppenheim and R.W Schaffer,
PHI, 2009 Fundamentals of Digital Signal Processing — Loney Ludeman,
John Wiley, 2009
ReferenceBo Digital Signal Processing — Fundamentals and Applications — Li

ok Tan, Elsevier, 2008
Reference Digital Signal Processing — S.Salivahanan, A.Vallavaraj and C.Gnanapriya,

Book TMH, 2009
Reference Discrete Systems and Digital Signal Processing with MATLAB —

Book Taan EIAIi. CRC press, 2009.

Reference Digital Signal Processing – Nagoor Khani, TMG, 2012

Book
6) Micro Lesson Plan
Topic wise Coverage [Micro Lesson Plan]

S.No. Topic Scheduled date Planned date
Unit-I
1 UNIT-1 Introduction to DSP
2 Discrete time signals and sequences
3 Linear shift invariant systems
4 Stability and Causality
5 Linear constant coefficient difference

equation
6 Frequency domain representation of DTS
7 Application of Z transform
8 Solution of difference equation of digital

filters
System function,
9 Stability criterion
10 Frequency response of stable systems
11 Realization of digital filters-direct and

canonic form
12 Cascade and parallel form P.USHA, Assistant. Professor,dept of ECE,KGRCET
13 Revision of 1st unit
14 Discrete time fourier transform

15 UNIT II- Discrete Fourier Series: DFS
representation
16 Properties of DFS
17 DFT and Properties of DFT
18 Linear convolution using DFT
19 Overlap save method
20 Overlap add method
21 Relation between DTFT and DFS
22 Relation between DFT and Z transform
23 FFT
24 Radix-2 decimation in time FFT algorithm
25 Radix-2 decimation in frequency
26 Inverse FFT
27 FFT with general radix N
28 Problems on overlap add method and save

method.
29 Problems on DIT algorithm
30 Problems on DIF algorithm
31 Revision
32 UNIT III- IIR Digital Filters
33 Analog filter approximations

Butterworth filter
34 Chebyshev filters
35 Design of IIR filters from Analog filters
36 Impulse invariant technique

37 Bilinear transformation method
38 Spectral transformation
39 Relation between analog and digital

frequencies
40 Problems on approximations of different
filters.
41 Revision
42 UNIT IV-FIR Digital filters
43 Frequency response
44 Design of FIR filters using Fourier method
45 Design of FIR filters using window

technique
46 Window technique problems
47 Problems on windowing techniques
48 Problems on Digital filters
49 Frequency sampling technique
50 Comparison of FIR and IIR filters
51 Applications of digital filters
52 Revision
53 UNIT V-Multi rate digital signal processing
54 Decimation
55 Interpolation
56 Sampling rate conversion
57 Finite word length effects

58 Limit cycle
59 Overflow oscillations
60 Round off noise
61 Computational output round off noise
62 Methods to prevent overflow
63 Trade off between round off and overflow

noise
64 Dead band effects
65 Problems
66 Revision
67 Architecture of TMS320C5X processor
68 Wavelet Transforms

7. Teaching Schedule
Subject Digital Signal Processing
Text Books (to be purchased by the Students)
Book 1 Digital Signal Processing – Principles, Algorithms, and Applications: John G. Proakis, Dimitris
G.Manolakis, Pearson Education / PHI, 2007.
Book 2 Discrete Time Signal Processing – A. V. Oppenheim and R.W.Schaffer, PHI, 2009.
Reference Books
Book 3 Digital Signal Processing – Fundamentals and Applications – Li Tan, Elsevier, 2008.
Book 4 Digital Signal Processing – S.Salivahanam, A.Vallavaraj and C.Gnanapriya, TMH, 2009.
Chapters Nos No of
classes
Unit Topic Book 1 Book 2 Book 3 Book 4
1 4
I Introduction to DSP
Realization of Digital Filters 1 1 1 9
Discrete Fourier series 2 2 2 3 2
Discrete Fourier Transforms 3 3 4 6
II Fast Fourier Transforms 6 2 4 9
IIR Digital Filters 7 6 4 2
Analog Filter approximations 7 6 1 4
III Design of IIR Digital Filters from

7 6 1 4
Analog Filters.
FIR Digital Filters 8 7 9 4
IV Design of FIR Filters 8 7 9 6
Multirate Digital Signal Processing 9 10 10 4

Finite Word Length effects 10 8
V Dead Band Effects 9 11 11 2
Contact classes for syllabus coverage 64
Lectures beyond syllabus 02
Tutorial classes 00
Classes for gaps&Add-on classes 02
Total No. of classes 68

11. MID EXAM DESCRIPTIVE QUESTION PAPERS
College Code
K. G. Reddy College of Engineering &Technology
(Approved by AICTE, Affiliated to JNTUH)

QM
Chilkur (Vil), Moinabad (Mdl), RR District
Name of the Exam: I Mid Examinations FEB-2018 Marks: 10
Year-Sem & Branch: IV Year I Sem & EEE Duration: 60 Min
Date &
Subject: DSP
Session
Answer ANY TWO of the following Questions 2X5=10
1. Find the total response of the system described by difference equation y(n)-4y(n- 1) +4y(n-2)
= x(n)-x(n- 1) when the input is x(n)= (-1)n u(n) with the initial condition y (-1) =y (-2) =1.
2. A) Find x(k) of the given sequence x(n) = {1,2,3,4,4,3,2,1} using DIT-FFT algorithm.
B) Determine the number of additions & multiplications required to implement 16-point DFT&
16 point FFT
3.A) Find the IDFT of the given sequence X (k) = {2, 2-3j, 2+3j,-2}
B) Obtain the relation between DFT & Z-transform
4.Derive the transfer function and order for the analog low pass butterworth filter

K. G. Reddy College of Engineering &Technology
(Approved by AICTE, Affiliated to JNTUH)
Chilkur (Vil), Moinabad (Mdl), RR District
_____________________________________________________________________________
Name of the Exam: II mid Examinations April– 2018
Year-Sem & Branch: IV-I EEE Duration: 60 Min
Subject: DIGITAL SIGNAL PROCESSING Date & Session:
Answer ANY TWO of the following Questions 2X5=10
Bloom’s Course
Q.NO QUESTION
level outcome
Design a Chebyshev filter for the following
specifications using impulse invariance method
0.8 ≤ |H(ejw) | ≤ 1 , 0 ≤ w ≤ 0.2π
1 L6,L3 CO5
|H(ejw) | ≤ 0.2 , 0.6π≤ w ≤ π
For the desired frequency response given by

Hd (ejw) = e-j3w , |w| ≤ 3π/4
2 =0 , 3π/4 < |w| <π L3 CO6
Find the filter coefficients for N=7 using hanning
window. Plot the magnitude response.
a) Discuss in detail down sampling process with a
neat diagram
3 b) Consider a signal x(n)=u(n) L2,L3 CO7
i)obtain a signal with a decimation factor 3
ii)obtain a signal with a interpolation factor 3
a) What are limit cycles and discuss various types
4 of limit cycles in brief L1,L4 CO8,CO6
b) Compare IIR and FIR digital filters

12. MID EXAM OBJECTIVE QUESTION PAPERS
Code No: 126VK Set No. 1

JAWAHARLAL NEHRU TECHNOLOGICAL UNIVERSITY HYDERABAD
III B.Tech. II Sem., I Mid-Term Examinations, February-2018
DIGITAL SIGNAL PROCESSENIG
Objective Exam
Name: ______________________________ Hall Ticket No. A
Answer All Questions. All Questions Carry Equal Marks. Time: 20 Min. Marks: 10.
I. Choose the correct alternative:

1. A discrete time signal has [ ]
a) Continuous time continuous amplitude b) Continuous time discrete amplitude
c) Discrete time continuous amplitude d) Discrete time discrete amplitude
2. The necessary and sufficient condition for causality of an LTI system is, [ ]
its unit sample response
a) h(n) = 0 for negative values of n b) h(n) = 0 for positive values of n
c) h(n) = 0 for integer values of n d) h(n) = 0 for complex values of n
3. FFT algorithm calculates [ ]

a) DTFT b) DCT c) DFT d) DST
4. The no. Of complex multiplications for computing DIF-FFT algorithm are [ ]
a) (N/2)log2N. a) (N/4)log4N. a) (N/8)log8N. d) (N/16)log16N
5. The advantage of cascade realization is [ ]

a) Quantization error can be minimized b) Random error can be minimized
c) System error can be minimized d) None of the above
6. Linear convolution of two real sequences with P and Q points, result for zero appended real sequence
after executing circular convolution using FFT will go wrong if FFT of length N is used such that
[ ]
a) N < P+Q-1 b) N=P+Q-1 c) N>P+Q-1 d) N<P+Q
7 he fundamental period of
𝜋𝑛 2𝑓0
x(n)=cos( ) is
8

a) N=8 b) N=4 c) N=16 d)none
8. Z-transform reduces to fourier transform when it is evaluated on [ ]
a) A half circle b) Z-circle c) Unit circle d) Imaginary circle

9. An LTI system having system function H(z) stable if and only if all the poles of H(z) are ____________
the unit circle [ ]
a) Outside b) On c) Inside d) None of the above
10. The zero padding of sequence x(n) to find DFT results in [ ]

a) Better display of the frequency spectrum b) Can be used for non-linear filtering
c) Better display of the phase spectrum d) None of the above
Code No: 126VK :2: Set No. 1
II Fill in the Blanks
11. The fourier transform of a discrete and a periodic sequence is ______________
12. A ___________ signal exhibits no un certainity of value at any given instant of time
13. A linear system is one which satisfies the principle of ___________
14. The _____________ theorem states that the fourier transform of autocorrelation sequence is the
energy density spectrum
15. Average power of energy signals is__________
16. When a sequence is circularly shifted in time by 5 units, the magnitude response __________
17. DTFT can be obtained by performing _________ in time domain
18. In DIF the complex multiplication takes place after the ___________ operation
19. The transition band is ________ in butter worth filter compared to Chebyshev filter
20. The ___________ is discrete time counterpart of the laplace transform

Code No:126VK Set No. 1

KEY
I Choose the correct alternative:

1. c
2. a
3. c
4. a
5. a
6. b
7. b
8. c
9. c
10. a
11. Continuous and periodic
12. Deterministic
13. Superposition
14. Wiener Khintchine
15. Zero

16. Remains unchanged
17. Sampling
18. Add-subtract
19. Less
20. Z-transform
Code No: 126VK Set No. 1
JAWAHARLAL NEHRU TECHNOLOGICAL UNIVERSITY HYDERABAD
B.Tech. III Year, II Sem., II Mid-Term Examinations, April-2018
Objective Exam
Name: ______________________________ Hall Ticket No. A
Answer All Questions. All Questions Carry Equal Marks. Time: 20 Min. Marks: 10.
JJJ Choose the correct alternative:
1. Bilinear transformation provides [ ]

a) One-to-many mapping b) Many-to-one mapping
c) Many-to-many mapping d) One-to-one mapping
2. For recursive realization the present output y(n) is a function of [ ]

a) Past outputs, past and present inputs b) Future outputs, present and future inputs
c) Past outputs, past and future inputs d) None of the above
3. The direct form I realization of IIR digital filters require [ ]

a) M+N+1 multiplications b) M+N-1 multiplications c) M+N multiplications d) None of the above
4.The impulse response, which is symmetric having odd number of samples can be used to design
[ ]
a) low pass b) High pass c) Band pass d) All of the above
5. The advantage of cascade realization is [ ]

a) Quantization error can be minimized b) Random error can be minimized
c) System error can be minimizedd) None of the above
6. The magnitude of the side lobe level in Hanning window is [ ]

a) 31dB b) -31 dB c) 30dB d) -30dB
7. Frequency sampling realization of FIR filter introduces [ ]

a) More number of poles on unit circle b) More number of zeros on unit circle
c) Equal number of poles and zeros on the unit circle d) None of the above

8. Limit cycle is [ ]
a) Zero input limit cycle b) Overflow limit cycle
c) Both zero input and overflow limit cycle d) None of the above
9. Decimation is a process in which the sampling rate is [ ]

Reducedb) Enhanced c) Stable d) Unpredictable
Code No: 126VK :2: Set No. 1

21. The anti- symmetrical impulse response can be used to design ______________
22. For ___________ realization the current output y(n) is a function of only past and present inputs
23. A ____ filter can have perfectly linear phase
14 The direct form FIR filter needs ................ between the adders to reduce the delay of the
adder tree and to achieve high throughput.
15. The minimum stop band attenuation of rectangular window is ____________
16. Kaiser window is based on __________functions
17. __________ filter is used specifically after upsampling process for removal of unwanted images
18. A __________is formed by an interconnection of the up-sampler, the down-sampler, and the
components of an LTI digital filter
19. The process of quantization introduces ___________
20. The frequency response of the system with input h(n) and window length M is given by________

Code No:126VK Set No. 1
KEY
I Choose the correct alternative:
1. D
2. A
3. A
4. D
5. A
6. B
7. C
8. C
9. A
10. C
11. Hilbert transformers and differentiators
12. Non-recursive
13. FIR
14. Extra pipeline registers
15. -21 dB
16. Prolate spheroidal
17. Anti –imaging
18. Complex multi rate system
19. Error
20. H(w)=∑ h(n) e-jwn

13. ASSIGNMENT TOPICS WITH MATERIALS
UNIT-I
1. Classification of systems
A discrete-time system can be thought of as a transformation or operator that maps an input sequence
{x[n]} to an output sequence
By placing various conditions on T(·) we can define different classes of systems. We give some properties
of systems. Basic System Properties are as given below
1) Systems with or without memory: A system is said to be memoryless if the out put for each value of the
independent variable at a given time n depends only on the input value at time n. For example system specified
by the relationship y[n] = cos(x[n]) + z is memoryless. A particularly simple memoryless system is the identity
system defined by y[n] = x[n]
In general we can write input-output relationship for memoryless system as
y[n] = g(x[n])
Not all systems are memoryless. A simple example of system with memory is a delay defined by
y[n] = x[n − 1]
A system with memory retains or stores information about input values at times other than the current input
value.
2) Invertibility: A system is said to be invertible if the input signal {x[n]} can be recovered from the output
signal {y[n]}. For this to be true two different input signals should produce two different outputs. If some
different input signal produce same output signal then by processing output we cannot say which input
produced the output.
Example of an invertible system is
then x[n] = y[n] − y[n − 1] Example if a non-invertible system is y[n]=0. That is the system produces an all zero
sequence for any input sequence. Since every input sequence gives all zero sequence, we can not find out which
input produced the output. The system which produces the sequence {x[n]} from sequence {y[n]} is called the
inverse system. In communication system, decoder is an inverse of the encoder.
3) Causality: A system is causal if the output at any time depends only on values of the input at the present
time and in the past.
All memoryless systems are causal. An accumulator system defined by

is also causal. The system defined by

is noncausal.
2. Linear constant coefficient difference equations

An important subclass of difference equations is the set of linear constant coefficient difference
equations. These equations are of the form
Cy(n)=f(n)
where C is a difference operator of the form given

C=cNDN+cN−1DN−1+...+c1D+c0
in which D is the first difference operator

D(y(n))=y(n)−y(n−1).
Note that operators of this type satisfy the linearity conditions, and c0,...,cn are real constants.
However, Equation 2 can easily be written as a linear constant coefficient recurrence equation without
difference operators. Conversely, linear constant coefficient recurrence equations can also be written in
the form of a difference equation, so the two types of equations are different representations of the same
relationship. Although we will still call them linear constant coefficient difference equations in this
course, we typically will not write them using difference operators. Instead, we will write them in the
simpler recurrence relation form
∑k=0Naky(n−k)=∑k=0Mbkx(n−k)
where x is the input to the system and y is the output. This can be rearranged to find y(n) as
y(n)=1a0(−∑k=1Naky(n−k)+∑k=0Mbkx(n−k))
The forms provided by equations will be used in the remainder of this course.
A similar concept for continuous time setting, differential equations, is discussed in the chapter on time
domain analysis of continuous time systems. There are many parallels between the discussion of linear
constant coefficient ordinary differential equations and linear constant coefficient differece equations.
3. Frequency domain representation of discrete time signals
Let us assume we have an LTI system
x[n] h[.] y[n]
If x[n]  e j 2f nd
then
 
y[n]   h[k ]x[n  k ]   h[k ]e j 2f d ( nk )
k   k  

 e j 2f n  h[k ]e  j 2f k
d d
k  
e j 2f d n
H (e jw )
Eigenfunction eigenvalue
Example:

Let x[ n ]  A cos 2fn     A
2

e
j   2fn   e  j   2fn 

y[ n ]   h[ k ] x[ n  k ]
k  
A  
j   2f n  k      
 j   2f n  k  
  
2 k
 h[ k ]e  h[ k ]e 
  k   

A
2
  
H e
jw
e
j   2fn
 
  H e  jw e  j   2fn 
A special case of this problem exist when h[n] is real
 
H e jw  H e jw  
In this case
 
y[n]  A H e jw cos2fn      where   angH e jw  
In other words a sinusoidal input to a discrete time LTI system provides a sinusoidal output.
Frequency Domain Representation of Discrete-Time Signals and Systems

Discrete time Fourier Transform is a tool by which a time-domain sequence is mapped into a continuous
function of a frequency variable. Because the DTFT is periodic the parent discrete-time sequence can be simply
obtained by computing its Fourier Series representation.
Definition of the Forward Transform
 
Discrete-time Fourier transform X e jw of a sequence x[n] is defined as:
X e jw  

 x[n]e
n  
 jwn
(1)
 
In general X e jw is a complex function of the real variable w and can be written as:
   
X e jw  X re e jw  jX im e jw  
 
X e jw can alternatively be expressed in polar form as:
   
X e jw  X e jw e j ( w) where,   
( w)  arg X e jw (2)
In many applications the Fourier transform is called the Fourier Spectrum and likewise X e jw  and (w) are
referred to as the “magnitude spectrum” and “phase spectrum” respectively.
 
Note from eq.(2) that if we replace (w) with (w) + 2k , where k is an integer, X e jw remains unchanged
implying that the phase function cannot be uniquely specified for any Fourier Transform.
4. Solution of difference equations of digital filters

The difference equation is a formula for computing an output sample at time based on past and present
6.1
input samples and past output samples in the time domain. We may write the
general, causal, LTI difference equation as follows:
(6.1)

where is the input signal, is the output signal, and the constants
, are called the coefficients

As a specific example, the difference equation
specifies a digital filtering operation, and the coefficient sets and fully characterize
the filter. In this example, we have .
When the coefficients are real numbers, as in the above example, the filter is said to be real. Otherwise, it
may be complex.
Notice that a filter of the form of Eq. (5.1) can use ``past'' output samples (such as ) in the
calculation of the ``present'' output . This use of past output samples is calledfeedback. Any filter
having one or more feedback paths ( ) is called recursive. (By the way, the minus signs for the
feedback in Eq. (5.1) will be explained when we get to transfer functions in §6.1.)
More specifically, the coefficients are called the feedforward coefficients and the coefficients are
called the feedback coefficients.
A filter is said to be recursive if and only if for some . Recursive filters are also called infinite-
impulse-response (IIR) filters. When there is no feedback ( ), the filter is said to be

a nonrecursive or finite-impulse-response (FIR) digital filter.
When used for discrete-time physical modeling, the difference equation may be referred to as
an explicit finite difference scheme.6.2
Showing that a recursive filter is LTI (Chapter 4) is easy by considering its impulse-response
representation (discussed in §5.6). For example, the recursive filter
has impulse response , . It is now straightforward to apply the analysis of the

previous chapter to find that time-invariance, superposition, and the scaling property hold.
5. Realization of digital filters using direct and cascade form
UNIT-II
1. Properties of DFT
As a special case of general Fourier transform, the discrete time transform shares all properties (and their
proofs) of the Fourier transform discussed above, except now some of these properties may take different
forms. In the following, we always assume and .

 Linearity
 Time Shifting

Proof:
If we let , the above becomes
 Time Reversal
 Frequency Shifting
 Differencing
Differencing is the discrete-time counterpart of differentiation.
Proof:
 Differentiation in frequency
proof: Differentiating the definition of discrete Fourier transform with respect to , we get

 Convolution Theorems
The convolution theorem states that convolution in time domain corresponds to multiplication in frequency
domain and vice versa:
Recall that the convolution of periodic signals and is
Here the convolution of periodic spectra and is similarly defined as
Proof of (a):
Proof of (b):

 Parseval's Relation
2. Computation of DFT using Over-Lap Add Method and Over-Lap Save Method
There are many DSP applications where a long signal must be filtered insegments. For instance,
high fidelity digital audio requires a data rate of about 5 Mbytes/min, while digital video requires
about 500 Mbytes/min. With data rates this high, it is common for computers to have insufficient
memory to simultaneously hold the entire signal to be processed. There are also systems that process
segment-by-segment because they operate in real time. For example, telephone signals cannot be
delayed by more than a few hundred milliseconds, limiting the amount of data that are available for
processing at any one instant. In still other applications, the processing may require that the signal be
segmented. An example is FFT convolution, the main topic of this chapter.
The overlap-add method is based on the fundamental technique in DSP:
(1) Decompose the signal into simple components,
(2) Process each of the components in some useful way, and
(3) Recombine the processed components into the final signal.
When an N sample signal is convolved with an M sample filter kernel, the output signal is N + M - 1
sample long. For instance, the input signal,
(a), is 300 samples (running from 0 to 299), the filter kernel,
(b), is 101 samples (running from 0 to 100), and the output signal, (i), is 400 samples.
When an N sample signal is filtered, it will be expanded by M - 1 point to the right. (This is
assuming that the filter kernel runs from index 0 to M. If negative indexes are used in the filter kernel,
the expansion will also be to the left). In (a), zeros have been added to the signal between sample 300
and 399 to illustrate where this expansion will occur. Don't be confused by the small values at the ends
of the output signal, (i). This is simply a result of the windowed-sinc filter kernel having small values
near its ends. All 400 samples in (i) are nonzero, even though some of them are too small to be seen in
the graph.
Below Figures show the decomposition used in the overlap-add method. The signal is broken into
segments, with each segment having 100 samples from the original signal. In addition, 100 zeros are
added to the right of each segment. In the next step, each segment is individually filtered by convolving it
with the filter kernel. This produces the output segments shown in figures. Since each input segment is
100 samples long, and the filter kernel is 101 samples long, each output segment will be 200 samples long.
The important point to understand is that the 100 zeros were added to each input segment to allow for the
expansion during the convolution.
Notice that the expansion results in the output segments overlapping each other. These overlapping output
segments are added to give the output signal, (i). For instance, samples 200 to 299 in (i) are found by
adding the corresponding samples in (g) and (h). The overlap-add method produces exactly the same
output signal as direct convolution. The disadvantage is a much greater program complexity to keep track
of the overlapping samples.
3. Relation between DTFT, DFS, DFT and Z-Transform
4. Decimation-in-Time and Decimation-in-Frequency FFT Algorithms
The FFT is a complicated algorithm, and its details are usually left to those that specialize in such things.
This section describes the general operation of the FFT, but skirts a key issue: the use of complex numbers.
In complex notation, the time and frequency domains each contain one signal made up of N complex
points. Each of these complex points is composed of two numbers, the real part and the imaginary part.
For example, when we talk about complex sample X[42], it refers to the combination of ReX[42]
and ImX[42]. In other words, each complex variable holds two numbers. When two complex variables are
multiplied, the four individual components must be combined to form the two components of the product
The following discussion on "How the FFT works" uses this jargon of complex notation. That is, the
singular terms: signal, point, sample, and value, refer to the combination of the real part and the imaginary
part.
The FFT operates by decomposing an N point time domain signal into N time domain signals each
composed of a single point. The second step is to calculate the N frequency spectra corresponding to
these N time domain signals. Lastly, the N spectra are synthesized into a single frequency spectrum.
Below figure shows an example of the time domain decomposition used in the FFT. In this example, a 16
point signal is decomposed through four
Separate stages. The first stage breaks the 16 point signal into two signals each consisting of 8 points. The
second stage decomposes the data into four signals of 4 points. This pattern continues until there
are N signals composed of a single point. An interlaced decomposition is used each time a signal is broken
in two, that is, the signal is separated into its even and odd numbered samples.
There are Log2N stages required in this decomposition, i.e., a 16 point signal (2 4) requires 4 stages, a 512
point signal (27) requires 7 stages, a 4096 point signal (212) requires 12 stages, etc. Remember this
value, Log2N.
This simple flow diagram is called a butterfly due to its winged

appearance. The butterfly is the basic computational element of the FFT,
transforming two complex points into two other complex points.
Figure shows the structure of the entire FFT. The time domain
decomposition is accomplished with a bit reversal sorting algorithm.
Transforming the decomposed data into the frequency domain
involves nothing and therefore does not appear in the figure.
The frequency domain synthesis requires three loops. The outer loop runs
through the Log2N stages (i.e., each level in Fig., starting from the bottom
and moving to the top). The middle loop moves through each of the
individual frequency spectra in the stage being worked on (i.e., each of the
boxes on any one level in Fig.). The innermost loop uses the butterfly to
calculate the points in each frequency spectra (i.e., looping through the
samples inside any one box in Fig.). The overhead boxes in Fig. determine the beginning and ending
indexes for the loops, as well as calculating the sinusoids needed in the butterflies. Now we come to the
heart of this chapter, the actual FFT programs.
5. Inverse FFT
If you need to compute inverse fast Fourier transforms (inverse FFTs) but you only have forward FFT
software (or forward FFT FPGA cores) available to you, below are four ways to solve your problem.
Preliminaries
To define what we're thinking about here, an N-point forward FFT and an N-point inverse FFT are
described by:
Forward FFT→X(m)=N−1∑n=0x(n)e−j2πnm/N(1)(1)Forward FFT→X(m)=∑n=0N−1x(n)e−j2πnm/NInv
erse FFT→x(n)=1NN−1∑m=0X(m)ej2πmn/NInverse FFT→x(n)=1N∑m=0N−1X(m)ej2πmn/N
=1NN−1∑m=0[Xreal(m)+jXimag(m)]ej2πmn/N(2)(2)=1N∑m=0N−1[Xreal(m)+jXimag(m)]ej2πmn/N
Inverse FFT Method# 1
The first method of computing inverse FFTs using the forward FFT was proposed as a "novel" technique
in 1988 [1]. That method is shown in Figure 1.
Figure 1: Method# 1 for computing the inverse FFT

using forward FFT software.
The second method of computing inverse FFTs using the forward FFT, similar to Method#1, is shown in
Figure 2(a). This Method# 2 has an advantage over Method# 1 when the input X(m)X(m) spectral samples
are conjugate symmetric. In that case, shown in Figure 2(b), only one data flipping operation is needed
because the output of the forward FFT will be real-only.
UNIT-III
1. Analog filter approximations -Butter worth
The classical method of analog filters design is Butterworth approximation. The Butterworth filters are also
known as maximally flat filters. Squared magnitude response of a Butterworth low-pass filter is defined as
follows
where - radian frequency, - constant scaling frequency, - order of the filter.
Some properties of the Butterworth filters are:
 gain at DC is equal to 1;
 has a maximum at
 The first derivatives of (3.1) are equal to zero at . This is why Butterworth filters are
known as maximally flat filters.
2. Impulse Invariant Techniques

the impulse-invariant method converts analog filter transfer functions to digital filter transfer functions in
such a way that the impulse response is the same (invariant) at the sampling instants [346], [365, pp. 216-
219]. Thus, if denotes the impulse-response of an analog (continuous-time) filter, then the digital
(discrete-time) filter given by the impulse-invariant method will have impulse response ,
where denotes the sampling interval in seconds. Moreover, the order of the filter is preserved, and

IIR analog filters map to IIR digital filters. However, the digital filter's frequency response is
an aliasedversion of the analog filter's frequency response. 9.3
To derive the impulse-invariant method, we begin with the analog transfer function
(9.1)
3. and perform a partial fraction expansion (PFE) down to first-order terms [452]:9.4
where is the th pole of the analog system, and is its residue [452]. Assume that the system is at
least marginally stable [452] so that there are no poles in the right-half plane ( ). Such a PFE is
always possible when is a strictly proper transfer function (more poles than zeros
[452]).9.5 Performing the inverse Laplace transform on the partial fraction expansion we obtain the impulse
response in terms of the system poles and residues:
We now sample at intervals of seconds to obtain the digital impulse response
Taking the z transform gives the digital filter transfer function designed by the impulse-invariant method:
We see that the -plane poles have mapped to the -plane poles
(9.2)

and the residues have remained unchanged. Clearly we must have , i.e., the analog
poles must lie within the bandwidth spanned by the digital sampling rate . Otherwise, the pole
angle will be aliased into the interval . Stability is preserved
since
3.Analog filter approximations - Chebyshev
The Chebyshev polynomial of degree n is denoted Tn(x), and is given by the explicit formula Tn(x) = cos(n arccos x) This
may look trigonometric at first glance (and there is in fact a close relation between the Chebyshev polynomials and the
discrete Fourier transform); however ( can be combined with trigonometric identities to yield explicit expressions for
Tn(x) ,
T0(x)=1
T1(x) = x
T2(x)=2x2 − 1
T3(x)=4x3 − 3x
T4(x)=8x4 − 8x2 + 1 ···
Tn+1(x)=2xTn(x) − Tn−1(x) n ≥ 1.
4. Bilinear Transformation Method
The bilinear transform is a special case of a conformal mapping (namely, a Möbius transformation), often
used to convert a transfer function of a linear, time-invariant (LTI) filter in the continuous-time
domain (often called an analog filter) to a transfer function of a linear, shift-invariant filter in
the discrete-time domain (often called a digital filteralthough there are analog filters constructed

with switched capacitors that are discrete-time filters). It maps positions on the axis, , in the s-
plane to the unit circle, , in the z-plane. Other bilinear transforms can be used to warp the frequency
response of any discrete-time linear system (for example to approximate the non-linear frequency
resolution of the human auditory system) and are implementable in the discrete domain by replacing a
system's unit delays with first order all-pass filters.

The transform preserves stability and maps every point of the frequency response of the continuous-time
filter, to a corresponding point in the frequency response of the discrete-time filter, although
to a somewhat different frequency, as shown in the Frequency warping section below. This means that
for every feature that one sees in the frequency response of the analog filter, there is a corresponding
feature, with identical gain and phase shift, in the frequency response of the digital filter but, perhaps, at a
somewhat different frequency. This is barely noticeable at low frequencies but is quite evident at
frequencies close to the Nyquist frequency.
5. Spectral Transformations.
When designing a particular filter, it is common practice to first design a Low-pass filter, and then using
a spectral transform to convert that lowpass filter equation into the equation of a different type of filter.
This is done because many common values for butterworth, cheybyshev and elliptical low-pass filters are
already extensively tabulated, and designing a filter in this manner rarely requires more effort then
looking values up in a table, and applying the correct transformations. This page will enumerate some of
the common transformations for filter design.
It is important to note that spectral transformations are not exclusively used for analog filters. There are
digital variants of these transformations that can be applied directly to digital filters to transform them into a
different type. This page will only talk about the analog filter transforms, and the digital filter transforms will
be discussed elsewhere.
UNIT-IV
1. Characteristics of FIR Digital Filters
Linear phase is a property of a filter, where the phase response of the filter is a linear
function of frequency. The result is that all frequency components of the input signal are shifted in time
(usually delayed) by the same constant amount, which is referred to as the phase delay. And
consequently, there is no phase distortion due to the time delay of frequencies relative to one another.

For discrete-time signals, perfect linear phase is easily achieved with a finite impulse response (FIR)
filter. Approximations can be achieved with infinite impulse response (IIR) designs, which are more
computationally efficient.
Linear-phase FIR filter can be divided into four basic types.
Type impulse response
I symmetric length is odd
II symmetric length is even
III anti-symmetric length is odd
IV anti-symmetric length is even
When h(n) is nonzero for 0 ≤ n ≤ N −1 (the length of the impulse response h(n) is N), then the symmetry of
the impulse response can be written as
h(n) = h(N − 1 − n)
and the anti-symmetry can be written as
h(n) = −h(N − 1 − n).
2. Digital Filters using Window Techniques
(a) Rectangular
(b) Bartlett (or triangle)
(c) Hanning

(d) Harming
(e) Blackman
(f) Kaiser
where is modified zero-order Bessel function of the first kind given by
The main lobe width and first side lobe attenuation increase as we proceed down the window listed above.
An ideal lowpass filter with linear phase and cut off is characterized by
The corresponding impulse response is

Since this is symmetric about , if we change and use one of the windows listed above the will get linear
phase FIR filter. Transition width and minimum stopped attenuation are listed in the Table 9.3.
Window Transition Width Minimum stopband attenuation

Rectangular -21db
Bartlett -25dB
Hanning -44dB
Hamming -53dB
Blackman -74dB
Kaiser variable variable
3. Design of FIR Filters using Fourier Method
In designing FIR filter using Fourier series method the infinite duration impulse response
is truncated at n= ± (N-1/2).Direct truncation of the series will lead to fixed percentage
overshoots and undershoots before and after an approximated discontinuity in the
frequency response .
4. Comparison of IIR & FIR filters
FIR filter IIR filter

These filters can be easily designed to have These filters do not have linear phase.
perfectly linear phase.
FIR filters can be realized recursively and non- IIR filters are easily realized recursively
recursively.
Greater flexibility to control the shape of their Less flexibility, usually limited to specific kind
magnitude response. of filters.
Errors due to round off noise are less severe in The round off noise in IIR filters is more.
FIR filters, mainly because feedback is not used.
5. Frequency Response of FIR digital filters

The frequency response of an LTI filter may be defined as the spectrum of the output signaldivided by
the spectrum of the input signal. In this section, we show that the frequency response of any LTI filter is
given by its transfer function evaluated on the unit circle, i.e., . We then show that this is the same result we
got using sine-wave analysis in Chapter 1.
Beginning with Eq. (6.4), we have
where X(z) is the z transform of the filter input signal , is the z transform of the output signal , and is the
filter transfer function.
A basic property of the z transform is that, over the unit circle , we find the spectrum[84].8.1To
show this, we set in the definition of the z transform, Eq. (6.1), to obtain
which may be recognized as the definition of the bilateral discrete time Fourier
transform(DTFT) when is normalized to 1 When is causal, this definition reduces to the usual
(unilateral) DTFT definition:
(8.1)
DTFT
Applying this relation to gives
UNIT-V
1. Decimation
Decimation can be regarded as the discrete-time counterpart of sampling. Whereas in sampling we start with a
continuous-time signal x(t) and convert it into a sequence of samples x[n], in decimation we start with a discrete-
time signal x[n] and convert it into another discrete-time signal y[n], which consists of sub-samples of x[n]. Thus,
the formal definition of M-fold decimation, or down-sampling, is defined by Equation. In decimation, the sampling
rate is reduced from Fs to Fs/M by discarding M – 1 samples for every M samples in the original sequence
∞
y[ n]= v[nM ]=∑ h[k] x[nM -k]
k=-∞

2. Interpolation
Interpolation is the exact opposite of decimation. It is an information preserving operation, in that all samples of
x[n] are present in the expanded signal y[n]. The mathematical definition of L-fold interpolation is defined by
Equation and the block diagram notation is depicted in below Figure Interpolation works by inserting (L–1) zero-
valued samples for each input sample. The sampling rate therefore increases from Fs to LFs. With reference to
Figure, the expansion process is followed by a unique digital low-pass filter called an anti-imaging filter. Although
the expansion process does not cause aliasing in the interpolated signal, it does however yield undesirable replicas
in the signal’s frequency spectrum.
y[n] =L h[k]w[n-k]
3. Round-off Noise in IIR Digital Filters

the roundoff noise for infinite-impulse-response (IIR) digital filters realized in some canonical forms.
The minimum roundoff noise realization of this class is achieved by letting the finite-impulse-response
(FIR) coefficients be identical to the corresponding partial impulse response of the desired filter. When
the order of the numerator of the transfer function of the desired filter is greater than or equal to that of
the denominator, there is an optimal decomposition which is shown to reduce the roundoff noise without
adding additional multipliers or adders. The proposed structure is applied to parallel and cascade forms,
with real as well as complex arithmetic and shown to result in lower roundoff noise. By increasing the
order of the FIR filter, the optimal synthesis becomes a noise reduction scheme at the cost of additional
computational complexity
4. Limit cycles
When a stable IIR filter digital filter is excited by a finite sequence, that is constant, the output will
ideally decay to zero. However, the non-linearity due to finite precision arithmetic operations often
causes periodic oscillations to occur in the output. Such oscillations occur in the recursive systems are
called Zero input Limit Cycle Oscillation. Normally oscillations in the absence of output u (k) =0 by
equation given below is called Limit cycle oscillations

Y[k] = -0.625.y [k-1] +u[k]
5. Dead Band Effects

A deadband (sometimes called a neutral zone or dead zone) is a band of input values in the domain of
a transfer function in a control system or signal processing system where the output is zero (the output is
'dead' - no action occurs). Deadband regions can be used in control systems such as servoamplifiers to
prevent oscillation or repeated activation-deactivation cycles (called 'hunting' in proportional
control systems). A form of deadband that occurs in mechanical systems, compound machines such
as gear trains is backlash.

15) UNIT WISE-QUESTION BANK
UNIT-I
2 MARKS QUESTIONS WITH ANSWERS
1) What is a continuous and discrete time signal?

Continuous time signal: A signal x(t) is said to be continuous if it is defined for all
time t. Continuous time signal arises naturally when a physical waveform such as acoustics wave or light
wave is converted into an electrical signal.
Discrete time signal: A discrete time signal is defined only at discrete instants of time.
The independent variable has discrete values only, which are uniformly spaced. A discrete time signal is
often derived from the continuous time signal by sampling it at a uniform rate.
2) Give the classification of signals?
There are 5 types of classifications of signals

1. Continuous-time and discrete time signals
2. Even and odd signals
3. Periodic signals and non-periodic signals
4. Deterministic signal and Random signal
5. Energy and Power signal
3) What are the types of systems?

Continuous time and discrete time systems
1. Linear and Non-linear systems
2. Causal and Non-causal systems
3. Static and Dynamic systems
4. Time varying and time in-varying systems
5. Distributive parameters and Lumped parameters systems
6. Stable and Un-stable systems.
4) What are even and odd signals?

Even signal: continuous time signal x(t) is said to be even if it satisfies the condition x(t)=x(-t) for
all values of t.
Odd signal: he signal x(t) is said to be odd if it satisfies the condition x(-t)=-x(t) for all t. In other
words even signal is symmetric about the time origin or the vertical axis, but odd signals are anti-
symmetric about the vertical axis.
5) What are deterministic and random signals?

Deterministic Signal: deterministic signal is a signal about which there is no certainty with respect
to its value at any time. Accordingly, we find that deterministic signals may be modeled as
completely specified functions of time.
Random signal: random signal is a signal about which there is uncertainty before its actual
occurrence. (eg.) The noise developed in a television or radio amplifier is an example for random
signal.
1) What are elementary signals and name them?

The elementary signals serve as a building block for the construction of more complex signals. They
are also important in their own right, in that they may be used to model many physical signals that
occur in nature.
There are five elementary signals. They are as follows
1) Unit step function
2) Unit impulse function
3) Ramp function
4) Exponential function
5) Sinusoidal function
2) What are time invariant systems?

A system is said to be time invariant system if a time delay or advance of the input signal leads to
an identical shift in the output signal. This implies that a time invariant system responds identically no matter
when the input signal is applied. It also satisfies the condition
R{x (n-k)} = y (n-k).
3) What do you mean by periodic and non-periodic signals?

A signal is said to be periodic if x (n + N) = x (n), Where N is the time period. A signal is said to be
non-periodic if x (n + N) = -x (n).
4) Define linear and non-linear system.

Linear system is one which satisfies superposition principle. Supeosition principle:
The response of a system to a weighted sum of signals be equal to the corresponding weighted sum of
responses of system to each of individual input signal.
i.e., T [a1x1=(n)+a2x2(n)]=a1T[x1(n)]+a2T[x2(n)]
A system, which does not satisfy superposition principle, is known as non-linear system.
5) Define causal and non-causal system.

The system is said to be causal if the output of the system at any time ‘n’ depends only on present
and past inputs but does not depend on the future inputs.
e.g.:- y (n) =x (n)-x (n-1)
A system is said to be non-causal if a system does not satisfy the above definition.
1) Explain in detail about discrete time systems

A discrete-time system can be thought of as a transformation or operator that maps an input sequence
{x[n]} to an output sequence
By placing various conditions on T(·) we can define different classes of systems. We give some properties
of systems. Basic System Properties are as given below
1) Systems with or without memory: A system is said to be memoryless if the out put for each value of the
independent variable at a given time n depends only on the input value at time n. For example system specified
by the relationship y[n] = cos(x[n]) + z is memoryless. A particularly simple memoryless system is the identity
system defined by y[n] = x[n]
In general we can write input-output relationship for memoryless system as
y[n] = g(x[n])
Not all systems are memoryless. A simple example of system with memory is a delay defined by
y[n] = x[n − 1]
A system with memory retains or stores information about input values at times other than the current input
value.
2) Invertibility: A system is said to be invertible if the input signal {x[n]} can be recovered from the output
signal {y[n]}. For this to be true two different input signals should produce two different outputs. If some
different input signal produce same output signal then by processing output we cannot say which input
produced the output.
Example of an invertible system is
then x[n] = y[n] − y[n − 1] Example if a non-invertible system is y[n]=0. That is the system produces an all zero
sequence for any input sequence. Since every input sequence gives all zero sequence, we can not find out which
input produced the output. The system which produces the sequence {x[n]} from sequence {y[n]} is called the
inverse system. In communication system, decoder is an inverse of the encoder.
3) Causality: A system is causal if the output at any time depends only on values of the input at the present
time and in the past.
All memoryless systems are causal. An accumulator system defined by

is also causal. The system defined by

is noncausal.
4) Stability
There are several definitions for stability. Here we will consider bounded input bonded output(BIBO) stability. A
system is said to be BIBO stable if every bounded input produces a bounded output. We say that a signal {x[n]}
is bounded if |x[n]|<M < ∞ for all n
The moving average system
is stable as y[n] is sum of finite numbers and so it is bounded. The accumulator system defined by
is unstable. If we take {x[n]} = {u[n]}, the unit step then y[0] = 1, y[1] =2, y[2] = 3, are y[n] = n +1, n ≥ 0 so y[n]
grows without bound.
5) Time invariance
A system is said to be time invariant if the behavior and characteristics of the system do not change with time.
Thus a system is said to be time invariant if a time delay or time advance in the input signal leads to identical
delay or advance in the output signal. Mathematically if {y[n]} = T ({x[n]}), Then {y[n – no]} = T({x[n – no]}) for
any no
Let us consider the accumulator system
If the input is now {x1[n]} = {x [n – no]} then the corresponding output is

The shifted output signal is given by
The two expression look different, but infact they are equal. Let us change the index of summation by l = k – no
in the first sum then we see that
Hence, {y[n]} = {y[n – no]} and the system is time-invariant. As a second example consider the system defined
by y[n] = nx[n]
if
{x1[n]} = {x[n – no]}
y1[n] = nx1[n] = n x[n − n0]
while y[n – no] = (n – no)x[n – no] and so the system is not time-invariant. It is time varying. We can also see this
by giving a counter example. Suppose input is {x[n]} = {δ[n]} then output is all zero sequence. If the input is {δ[n
−1]} then output is {δ[n−1]} which is definitely not a shifted version version of all zero sequence.
6) Linearity
This is an important property of the system. We will see later that if we have system which is linear and time
invariant then it has a very compact representation. A linear system possesses the important property of
superposition: If an input consists of weighted sum of several signals, the and the output is also weighted sum
of the responses of the system to each of those input signals. Mathematically let {y1[n]} be the response of the

system to the input {x1[n]} and let {y2[n]} be the response of the system to the input {x2[n]}. Then the system is
linear if:
2) Prove that the impulse response of an LTI system is absolutely summable for stability of the system
Due to its convolution property, the z-transform is a powerful tool to analyze LTI systems
When the input is the eigenfunction of all LTI system, i.e., , the operation on this input by
the system can be found by multiplying the system's eigenvalue H(z) to the input.
Causal LTI systems

An LTI system is causal if its output y[n] depends only on the current and past input x[n] (but not the future).
Assuming the system is initially at rest with zero output , then its response to an
impulse at is at rest for , i.e., . Its response to a general
input is:
Due to the properties of the ROC, we know that If an LTI system is causal (with a right sided impulse response
function for ), then the ROC of its transfer function is the exterior of a circle
including infinity. In particular, when is rational, then the system is causal if and only if its ROC is the
exterior of a circle outside the out-most pole, and the order of numerator is no greater than the order of the
denominator.
Note the requirement for the orders of the numerator and denominator guarantees the existence of even
when .
Stable LTI systems

An LTI system is stable if its response to any bounded input is also bounded for all :
As the output and input of an LTI is related by convolution, we have:
which obviously requires:
In other words, if the impulse response function of an LTI system is absolutely integrable, then the
system is stable. We can show that this condition is also necessary, i.e., all stable LTI systems' impulse
response functions are absolutely integrable. Now we have:
An LTI system is stable if and only if its impulse response is absolutely summable, i.e., the frequency response
function exits, i.e. the ROC of its transfer function includes the unit circle
2) Explain the principle of operation of analog to digital conversion with a neat diagram
Digital signal processing (DSP) is the use of digital processing, such as by computers, to perform a wide variety
of signal processing operations. The signals processed in this manner are a sequence of numbers that
represent samples of a continuous variable in a domain such as time, space, or frequency. Digital signal
processing and analog signal processing are subfields of signal processing. DSP applications
include audio and speech signal processing, sonar, radar and other sensor array processing, spectral
estimation, statistical signal processing, digital image processing, signal processing
for telecommunications, control of systems, biomedical engineering, seismic data processing, among others.
Digital signal processing can involve linear or nonlinear operations. Nonlinear signal processing is closely
related to nonlinear system identification and can be implemented in the time, frequency, and spatial-temporal
domains.
Signal sampling
The increasing use of computers has resulted in the increased use of, and need for, digital signal processing. To
digitally analyze and manipulate an analog signal, it must be digitized with an analog-to-digital converter.
Sampling is usually carried out in two stages, discretization and quantization.
Discretization means that the signal is divided into equal intervals of time, and each interval is represented by a

single measurement of amplitude.
Quantization means each amplitude measurement is approximated by a value from a finite set. Rounding real
numbers to integers is an example.
The Nyquist–Shannon sampling theorem states that a signal can be exactly reconstructed from its samples if the
sampling frequency is greater than twice the highest frequency of the signal. In practice, the sampling
frequency is often significantly higher than twice that required by the signal's limited bandwidth.
Theoretical DSP analyses and derivations are typically performed on discrete-time signal models with no
amplitude inaccuracies (quantization error), "created" by the abstract process of sampling. Numerical methods
require a quantized signal, such as those produced by an analog-to-digital converter (ADC). The processed
result might be a frequency spectrum or a set of statistics. But often it is another quantized signal that is
converted back to analog form by a digital-to-analog converter (DAC).
Domains:
In DSP, engineers usually study digital signals in one of the following domains: time domain (one-dimensional
signals), spatial domain (multidimensional signals), frequency domain, and wavelet domains. They choose the
domain in which to process a signal by making an informed assumption (or by trying different possibilities) as
to which domain best represents the essential characteristics of the signal. A sequence of samples from a
measuring device produces a temporal or spatial domain representation, whereas a discrete Fourier
transform produces the frequency domain information, that is, the frequency spectrum.
Time and space domains
The most common processing approach in the time or space domain is enhancement of the input signal through
a method called filtering. Digital filtering generally consists of some linear transformation of a number of
surrounding samples around the current sample of the input or output signal. There are various ways to
characterize filters; for example:
 A "linear" filter is a linear transformation of input samples; other filters are "non-linear". Linear filters
satisfy the superposition condition, i.e. if an input is a weighted linear combination of different signals,
the output is a similarly weighted linear combination of the corresponding output signals.
 A "causal" filter uses only previous samples of the input or output signals; while a "non-causal" filter
uses future input samples. A non-causal filter can usually be changed into a causal filter by adding a
delay to it.
 A "time-invariant" filter has constant properties over time; other filters such as adaptive filters change in
time.
 A "stable" filter produces an output that converges to a constant value with time, or remains bounded
within a finite interval. An "unstable" filter can produce an output that grows without bounds, with
bounded or even zero input.
 A "finite impulse response" (FIR) filter uses only the input signals, while an "infinite impulse response"
filter (IIR) uses both the input signal and previous samples of the output signal. FIR filters are always
stable, while IIR filters may be unstable.
A filter can be represented by a block diagram, which can then be used to derive a sample
processing algorithm to implement the filter with hardware instructions. A filter may also be
described as a difference equation, a collection of zeroes and poles or, if it is an FIR filter,
an impulse response or step response.
The output of a linear digital filter to any given input may be calculated by convolving the
input signal with the impulse response.
Block Diagram:
3) What are the advantages of DSP over ASP?
 The main advantage of digital signals over analog signals is that the precise signal level of the digital
signal is not vital. This means that digital signals are fairly immune to the imperfections of real
electronic systems which tend to spoil analog signals. As a result, digital CD's are much more robust
than analog LP's.
 Codes are often used in the transmission of information. These codes can be used either as a means of
keeping the information secret or as a means of breaking the information into pieces that are
manageable by the technology used to transmit the code, e.g. The letters and numbers to be sent by a
Morse code are coded into dots and dashes.
 Digital signals can convey information with greater noise immunity, because each information
component (byte etc) is determined by the presence or absence of a data bit (0 or one). Analog signals
vary continuously and their value is affected by all levels of noise.
 Digital signals can be processed by digital circuit components, which are cheap and easily produced in
many components on a single chip. Again, noise propagation through the demodulation system is
minimized with digital techniques.
 Digital signals do not get corrupted by noise etc. You are sending a series of numbers that represent the
signal of interest (i.e. audio, video etc.)
 Digital signals typically use less bandwidth. This is just another way to say you can cram more
information (audio, video) into the same space.
 Digital can be encrypted so that only the intended receiver can decode it (like pay per view video,
secure telephone etc.)
 Enables transmission of signals over a long distance.
 Transmission is at a higher rate and with a wider broadband width.
 It is more secure.
 It is also easier to translate human audio and video signals and other messages into machine language.
 There is minimal electromagnetic interference in digital technology.
 It enables multi-directional transmission simultaneously.
4) Discuss the steps involved in calculating convolution sum by taking simple practical example
The steps involved in calculating sum are
1) Folding
2) Shifting
3) Multiplication
4) Summation
.Let's say a chef decides to offer private cooking lessons. A starting pupil receives a 1-hour lesson to explore
interests and background. Next day, and all the days following, the private lessons are 3 hours long. If the
function h is the hours per day for one pupil, the graph would look like:
Hours per day in training one student.

We have 1 hour of instruction for the start day d = 0. Afterwards, for d = 1, 2, …, we have 3 hours.On the first
day, there are 2 pupils. On the second day, 4 new pupils arrive and on the third day, there are 6. No additional
pupils enroll after the third day. If p is the number of pupils who start on a given day, then a graph of the
function p is:
Number of new pupils who enroll on each day.

How many hours of instruction are given on the first day? Two pupils times 1 hour each gives 1*2 = 2. If T is
the total hours per day, then on the zeroth day, T(0) = 2. Now it starts to get interesting.On the following day,
the 2 pupils from the previous day are each getting 3 hours of instruction, and the 4 new pupils receive their 1
hour of orientation. This means T(1) = 3*2 + 1*4 = 10.On the following day, T(2) = 3*2 + 3*4 + 1*6 = 24.
Seems like everybody wants to learn to make croissants! Time to hire more help.But what is this type of
computation? What we have here is called a convolution. It's a type of sum of multiplications that keeps track
of what happened before. Here's how to set up the convolution. First, we fold one of the functions. Here is
the p function folded:

Folding the p function.

Then we line up the folded p with h:
Lining up the functions.

Multiplying point-by-point gives 1*2 = 2. This is the T(0) result.Next day, the pupils shift over to the right by
1.
Shifting p by one day.

We multiply and add to get T(1) = 3*2 + 1*4 = 10. Let's use a new symbol λ to keep track of the day. Our
function h is written as h(λ). To fold h we write h(-λ). To shift to the right, h(-λ) becomes h(d - λ). We're almost
there! The Σ means to add. Now, we can write the convolution as:
The λ = 0 at the bottom of the Σ is our start value for λ. We will have d multiplies to add: one for λ = 0, another
for λ = 1, and we keep going until λ = d. See the 2 above the Σ in the next equation? The number 2 tells us the
end value for λ. To clarify, let's work out the number of hours of instruction for d = 2.

5) What are the Characteristics of a DSP?

 A unit that can handle floating numbers is present directly in the data flow path.
 The accumulators or multipliers that are present are highly parallel in nature.
 Special hardware is included in order to carry looping at a very low cost.
 Their architecture is specially designed so that fetching multiple data at the same time is possible.
 The calculations are usually carried out by fixed point arithmetic process in order to speed them up.
 Most of the registers present in computers today move the data to the lower-most bit if an overflow
occurs. However, in case of digital signal processors, the overflow is retained at the maximum point
itself.
 Specialised instructions are present for modulo and reversed bit addressing.
FILL IN THE BLANKS
1) _______ is a System which carries out mathematical operations on a sequences of samples related to a
signal.
2) _______ is defined as any physical quantity that varies with time, space, or any other independent
variable
3) A discrete time signal having a set of discrete values is ____________
4) The Process of converting a continuous-valued signal into discrete-valued signal is called
__________________.
5) Any signal that can be uniquely described by an explicit mathematical expression, a table of data, or a
well-defined rule is ______________
6) A system is said to be _________ if the output of the system at any time n depends only on present and
past inputs
7) ____________ is a constant function that describes the magnitude and phase shift of a filter over a range
of frequencies
8) ___________ is the property that states the Z-transforms of a sum of signals is the sum of individual Z-
transforms.
9) __________ of time signals corresponds to the multiplication of Z-transform.
10) If the ROC extends outward from the outermost pole, then the system is _________.
1) DSP
2) Signal
3) Digital Signal
4) Quantization
5) Deterministic
6) Causal
7) Frequency Response
8) Linearity
9) Convolution
10) Causal
MULTIPLE CHOICE QUESTIONS
1) The interface between an analog signal and a digital processor is

a. D/A converter
b. A/D converter
c. Modulator
d. Demodulator
ANSWER: (b) A/D converter
2) The speech signal is obtained after
a. Analog to digital conversion
b. Digital to analog conversion
c. Modulation
d. Quantization
ANSWER: (b) Digital to analog conversion
3) Telegraph signals are examples of
a. Digital signals
b. Analog signals
c. Impulse signals
d. Pulse train
ANSWER: (a) Digital signals
4) As compared to the analog systems, the digital processing of signals allow
1) Programmable operations
2) Flexibility in the system design
3) Cheaper systems
4) More reliability
a. 1, 2 and 3 are correct
b. 1 and 2 are correct
c. 1, 2 and 4 are correct
d. All the four are correct
ANSWER: (d) All the four are correct
5) The Nyquist theorem for sampling
1) Relates the conditions in time domain and frequency domain
2) Helps in quantization
3) Limits the bandwidth requirement
4) Gives the spectrum of the signal
c. 1 and 3 are correct
ANSWER: (c) 1 and 3 are correct
6) Roll-off factor is
a. The bandwidth occupied beyond the Nyquist Bandwidth of the filter
b. The performance of the filter or device
c. Aliasing effect
d. None of the above
ANSWER: (a) The bandwidth occupied beyond the Nyquist Bandwidth of the filter
7) A discrete time signal may be
1) Samples of a continuous signal
2) A time series which is a domain of integers
3) Time series of sequence of quantities
4) Amplitude modulated wave
ANSWER: (a) 1, 2 and 3 are correct
8) The discrete impulse function is defined by
a. δ(n) = 1, n ≥ 0
= 0, n ≠ 1
b. δ(n) = 1, n = 0
= 0, n ≠ 1
c. δ(n) = 1, n ≤ 0
= 0, n ≠ 1
d. δ(n) = 1, n ≤ 0
= 0, n ≥ 1
ANSWER: (b) δ(n) = 1, n = 0
= 0, n ≠ 1
9) DTFT is the representation of
a. Periodic Discrete time signals

b. Aperiodic Discrete time signals
c. Aperiodic continuous signals
d. Periodic continuous signals
ANSWER :(b) Aperiodic Discrete time signals
10) The transforming relations performed by DTFT are
1) Linearity
2) Modulation
3) Shifting
4) Convolution
ANSWER: (d) All the four are correct
UNIT 2
1. State the properties of DFT
 Periodicity
 Linearity and symmetry Multiplication of two DFTs
 Circular convolution
 Time reversal
 Circular time shift and frequency shift Complex conjugate
 Circular correlation
2. Define circular convolution
Let x1(n) and x2(n) are finite duration sequences both of length N with DFTs X1 (k) and X2 (k). If
X3(k) = X1(k) X2(k) then the sequence x3(n) can be obtained by circular.
3. Define sectional convolution
If the data sequence x (n) is of long duration it is very difficult to obtain the output sequence y(n) due
to limited memory of a digital computer. Therefore, the data sequence is divided up into smaller sections.

These sections are processed separately one at a time and controlled later to get the output.
4. What is FFT?
The Fast Fourier Transform is an algorithm used to compute the DFT. It makes use of the symmetry and
periodicity properties of twiddle factor to effectively reduce the DFT computation time. It is based on the
fundamental principle of decomposing the computation of DFT of a sequence of length N into successively
smaller DFTs.
5. How many multiplications and additions are required to compute N point DFT using radix- 2FFT?
The number of multiplications and additions required to compute N point DFT using radix-2 FFT are N log2 N
and N/2 log2 N respectively
1. What is overlap-add method?

In this method the size of the input data block xi (n) is L. To each data block we append M-1 zeros
and perform N point cicular convolution of xi (n) and h(n). Since each data block is terminated with M-1
zeros the last M-1 points from each output block must be overlapped and added to first M- 1 points of the
succeeding blocks. This method is called overlap-add method.
2. Why FFT is needed?

The direct evaluation DFT requires N2 complex multiplications and N2 –N complex additions. Thus
for large values of N direct evaluation of the DFT is difficult. By using FFT algorithm, the number of
complex computations can be reduced. Therefore, we use FFT.
3. What are the applications of FFTalgorithm?

The applications of FFT algorithm includes
Linear filtering
Correlation Spectrum
analysis
4. Distinguish between linear convolution and circular convolution of two sequences.
Linear convolution Circular convolution

If x(n) is a sequence of L number of samples If x(n) is a sequence of L number of samples
and h(n) with M number of samples, after and h(n) with M samples, after convolution
convolution y(n) will have N=L+M-1 samples. y(n) will have N=max(L,M)samples.
It can be used to find the response of a linear It cannot be used to find the response of a filter
filter
Zero padding is not necessary to find the Zero padding is necessary to find the response
response of a linear filter.
5. What are the differences and similarities between DIF and DITalgorithms?
Differences:
1) The input is bit reversed while the output is in natural order for DIT, whereas for DIF the output is
bit reversed while the input is in naturalorder.
2) The DIF butterfly is slightly different from the DIT butterfly, the difference being that the complex
multiplication takes place after the add-subtract operation inDIF.
Similarities:
Both algorithms require same number of operations to compute the DFT. Both algorithms can be done in
place and both need to perform bit reversal at some place during the computation.
1. Explain properties of Discrete Fourier Transform.

As a special case of general Fourier transform, the discrete time transform shares all properties (and their
proofs) of the Fourier transform discussed above, except now some of these properties may take different
forms. In the following, we always assume and .

 Linearity
 Time Shifting
Proof:
If we let , the above becomes

 Time Reversal
 Frequency Shifting
 Differencing
Differencing is the discrete-time counterpart of differentiation.
Proof:
 Differentiation in frequency
proof: Differentiating the definition of discrete Fourier transform with respect to , we get
 Convolution Theorems
The convolution theorem states that convolution in time domain corresponds to multiplication in frequency
domain and vice versa:

Recall that the convolution of periodic signals and is
Here the convolution of periodic spectra and is similarly defined as
Proof of (a):
Proof of (b):

 Parseval's Relation
2) Describe the decimation in time [DIT] radix-2 FFT algorithm to determineN-point
DFT.
This algorithm is very similar in concept to the Decimation in Frequency (DIF) Algorithm discussed earlier,
so the presentation will be a little less detailed. Have a look at the section describing the DIF algorithm first.
We defined the FFT as:
If N is even, the above sum can be split into 'even' (n=2n') and 'odd' (n=2n'+1) halves, where n'=0..N/2-1,
and re-arranged as follows:
This process of splitting the 'time domain' sequence into even an odd samples is what gives the algorithm its
name, 'Decimation In Time'. As with the DIF algorithm, we have succeeded in expressing an N point
transform as 2 (N/2) point sub-transforms. The principal difference here is that the order we do things has
changed. In the DIF algorithm the time domain data was 'twiddled' before the two sub-transforms were
performed. Here the two sub-transforms are performed first. The final result is obtained by 'twiddling' the
resulting frequency domain data. There is a slight problem here, because the two sub-transforms only give
values for k=0..N/2-1. We also need values for k=N/2..N-1. But from the periodicity of the DFT we know:

Also..
So, for k=0..N/2-1:
and..
where:
This all we need to produce a simple recursive DIT FFT routine for any N which is a regular power of 2
(N=2p).
3) Explain overlap add method

There are many DSP applications where a long signal must be filtered insegments. For instance, high
fidelity digital audio requires a data rate of about 5 Mbytes/min, while digital video requires about 500
Mbytes/min. With data rates this high, it is common for computers to have insufficient memory to
simultaneously hold the entire signal to be processed. There are also systems that process segment-by-
segment because they operate in real time. For example, telephone signals cannot be delayed by more than a
few hundred milliseconds, limiting the amount of data that are available for processing at any one instant. In
still other applications, the processing may require that the signal be segmented. An example is FFT
convolution, the main topic of this chapter.
The overlap-add method is based on the fundamental technique in DSP:
(1) Decompose the signal into simple components,
(2) Process each of the components in some useful way, and
(3) Recombine the processed components into the final signal.
When an N sample signal is convolved with an M sample filter kernel, the output signal is N + M - 1 sample
long. For instance, the input signal,
(a), is 300 samples (running from 0 to 299), the filter kernel,
(b), is 101 samples (running from 0 to 100), and the output signal, (i), is 400 samples.
When an N sample signal is filtered, it will be expanded by M - 1 point to the right. (This is assuming
that the filter kernel runs from index 0 to M. If negative indexes are used in the filter kernel, the expansion
will also be to the left). In (a), zeros have been added to the signal between sample 300 and 399 to illustrate
where this expansion will occur. Don't be confused by the small values at the ends of the output signal, (i).
This is simply a result of the windowed-sinc filter kernel having small values near its ends. All 400 samples
in (i) are nonzero, even though some of them are too small to be seen in the graph.
Below Figures show the decomposition used in the overlap-add method. The signal is broken into
segments, with each segment having 100 samples from the original signal. In addition, 100 zeros are added
to the right of each segment. In the next step, each segment is individually filtered by convolving it with the
filter kernel. This produces the output segments shown in figures. Since each input segment is 100 samples
long, and the filter kernel is 101 samples long, each output segment will be 200 samples long. The important
point to understand is that the 100 zeros were added to each input segment to allow for the expansion during
the convolution.
Notice that the expansion results in the output segments overlapping each other. These overlapping output
segments are added to give the output signal, (i). For instance, samples 200 to 299 in (i) are found by adding
the corresponding samples in (g) and (h). The overlap-add method produces exactly the same output signal
as direct convolution. The disadvantage is a much greater program complexity to keep track of the
overlapping samples.
4) Explain FFT
The FFT is a complicated algorithm, and its details are usually left to those that specialize in such things.
This section describes the general operation of the FFT, but skirts a key issue: the use of complex numbers.
In complex notation, the time and frequency domains each contain one signal made up of N complex points.
Each of these complex points is composed of two numbers, the real part and the imaginary part. For
example, when we talk about complex sample X[42], it refers to the combination of ReX[42] and ImX[42]. In
other words, each complex variable holds two numbers. When two complex variables are multiplied, the
four individual components must be combined to form the two components of the product
The following discussion on "How the FFT works" uses this jargon of complex notation. That is, the
singular terms: signal, point, sample, and value, refer to the combination of the real part and the imaginary
part.
The FFT operates by decomposing an N point time domain signal into N time domain signals each
composed of a single point. The second step is to calculate the N frequency spectra corresponding to
these N time domain signals. Lastly, the N spectra are synthesized into a single frequency spectrum.
Below figure shows an example of the time domain decomposition used in the FFT. In this example, a 16
point signal is decomposed through four

Separate stages. The first stage breaks the 16 point signal into two signals each consisting of 8 points. The
second stage decomposes the data into four signals of 4 points. This pattern continues until there
are N signals composed of a single point. An interlaced decomposition is used each time a signal is broken in
two, that is, the signal is separated into its even and odd numbered samples.
There are Log2N stages required in this decomposition, i.e., a 16 point signal (2 4) requires 4 stages, a 512
point signal (27) requires 7 stages, a 4096 point signal (212) requires 12 stages, etc. Remember this
value, Log2N.
This simple flow diagram is called a butterfly due to its winged

appearance. The butterfly is the basic computational element of the FFT,
transforming two complex points into two other complex points.
Figure shows the structure of the entire FFT. The time domain
decomposition is accomplished with a bit reversal sorting algorithm.
Transforming the decomposed data into the frequency domain
involves nothing and therefore does not appear in the figure.
The frequency domain synthesis requires three loops. The outer loop runs
through the Log2N stages (i.e., each level in Fig., starting from the bottom
and moving to the top). The middle loop moves through each of the
individual frequency spectra in the stage being worked on (i.e., each of the
boxes on any one level in Fig.). The innermost loop uses the butterfly to
calculate the points in each frequency spectra (i.e., looping through the
samples inside any one box in Fig.). The overhead boxes in Fig. determine the beginning and ending indexes
for the loops, as well as calculating the sinusoids needed in the butterflies. Now we come to the heart of this
chapter, the actual FFT programs.
5) Explain about inverse FFT

If you need to compute inverse fast Fourier transforms (inverse FFTs) but you only have forward FFT
software (or forward FFT FPGA cores) available to you, below are four ways to solve your problem.
Preliminaries
To define what we're thinking about here, an N-point forward FFT and an N-point inverse FFT are described
by:
Forward FFT→X(m)=N−1∑n=0x(n)e−j2πnm/N(1)(1)Forward FFT→X(m)=∑n=0N−1x(n)e−j2πnm/NInvers
e FFT→x(n)=1NN−1∑m=0X(m)ej2πmn/NInverse FFT→x(n)=1N∑m=0N−1X(m)ej2πmn/N
=1NN−1∑m=0[Xreal(m)+jXimag(m)]ej2πmn/N(2)(2)=1N∑m=0N−1[Xreal(m)+jXimag(m)]ej2πmn/N
The first method of computing inverse FFTs using the forward FFT was proposed as a "novel" technique in
1988 [1]. That method is shown in Figure 1.

The second method of computing inverse FFTs using the forward FFT, similar to Method#1, is shown in
Figure 2(a). This Method# 2 has an advantage over Method# 1 when the input X(m)X(m) spectral samples
are conjugate symmetric. In that case, shown in Figure 2(b), only one data flipping operation is needed
because the output of the forward FFT will be real-only.

Figure 2: Method# 2 Processing flow: (a) standard Method# 2;
(b) Method# 2 when X(m) samples are conjugate symmetric.
The next two inverse FFT methods are of interest because they avoid the data reversals necessary in
Method# 1 and Method# 2.
The third method of computing inverse FFTs using the forward FFT, by way of data swapping, is shown in
Figure 3.

The fourth method of computing inverse FFTs using the forward FFT, by way of complex conjugation, is
shown in Figure 4.

FILL IN THE BLANKS
1) The important tools used in frequency analysis of signals are _____________and __________ ___
2) __________ can be used to find the response of a linear filter.
3) _________ is not necessary to find the response of a linear filter in linear convolution.
4) The linear convolution of 2 finite duration sequences x(n) and h(n) of lengths L samples and M samples
will result in a output sequence of duration ___________ samples
5) By using with Zero padding, DFT can be used in ____________

6) The two methods to compute DFT are ____________ and __________________
7) In overlap save method the size of the input data block is __________
8) In overlap add method the size of the input data block is ____________
9) ___________ is defined as an algorithm that efficiently computes the DFT.
10) ___________multiplications are required to compute DFT
1 DFT , IDFT
2 Linear
Convolution
3 Zero Padding
4 L+M-1
5 Linear Filtering
6 Overlap Add and
Overlap Save
7 N=L+M-1
8 L
9 FFT
10 N2
Multiple Choice Questions
2𝜋
1. Given that , W = 𝑒 −𝑖( 𝑁 ) ,where N = 3. Then F = 𝑊 𝑁 can be computed as F =
(A) 0 (B) 1
(C)- 1 (D) 5
Solution
The correct answer is (B).

2𝜋
−𝑖( )
2) Given that W = 𝑒 𝑁 , where N = 3 and F = 𝑊𝑁/2 can be computed as F =
(A) 0 (B) 1
(C) -1 (D) e
Solution
The correct answer is (C).
4 − 6𝑖
3.Given that N=2, {f} = { }. The valuesfor vector shown in
−2 + 4𝑖
Can be computed as
−2 −2
(A) { } (B){ }
−6 6
2 2
(C) { } (D) { }
−6 6
Solution
The correct answer is(D).

4 − 6𝑖
4. Given that N=2, {f} = { }. The values for vector shown in below equation
−2 + 4𝑖
can be computed as
−2
(A) { }
−10
−1
(B) { }
−10
−2
(C) { }
−5
−1
(D) { }
−5
Solution
The correct answer is(A).
5. If the forcing function F(t) is given as F(t) = ∑7𝑛=0 10 sin(2𝜋𝑛𝑡).Then, to avoid aliasing phenomenon, the
minimum number of sample data points Nminshould be
(A)8 (B)16
(C)24 (D) 32
Solution
The correct answer is (B).
6. Based on the figure below, aliasing phenomena will not occur because there were

(A) 2 sample data points per cycle.
(B) 4 sample data points per cycle.
(C) 4 sample data points per 2 cycles.
(D) 6 sample data points per 2 cycles.
Solution
The correct answer is (D)
7. Using the definition E= 𝑒 −𝑖2𝜋/𝑁 , and the Euler identity𝑒 ±𝑖𝜃 = 𝑐𝑜𝑠𝜃 ± 𝑠𝑖𝑛𝜃, the value E N/6 can be computed
as
(A) 0.866 − 0.5i
(B) − 0.866 + 0.5i
(C) − 0.5 − 0.866i
(D) 0.5 − 0.866i
Solution

The correct answer is (D),
8. Using the definition E = 𝑒 −𝑖2𝜋/𝑁 , and the Euler identity 𝑒 ±𝑖𝜃 = 𝑐𝑜𝑠𝜃 ± 𝑠𝑖𝑛𝜃, the value E 6N can be computed
as
(A) i+1
(B) i−1
(C) 1
(D) -1
Solution
The correct answer is (C).

9)
Solution
The correct answer is(A).
(10) For N=24=16, levelL=2and referring to the figure shown below, the only terms of vector f2(−)which only
need to compute are:

(A) f2(4−7,12−15) (B )f2(0−3,8−11)
(C) f2(0−7) (D) f2(8−15)
Solution
The correct answer is(B).
UNIT III
1. What are the different types of filters based on impulseresponse?

Based on impulse response the filters are of two types
1. IIRfilter
2. FIRfilter
The IIR filters are of recursive type, whereby the present output sample depends on the present input,
past input samples and output samples.
The FIR filters are of non-recursive type, whereby the present output sample depends on the present
input sample and previous input samples.
2. What are the different types of filters based on frequencyresponse?

Based on frequency response the filters can be classified as
1. Low passfilter
2. High passfilter
3. Bandpassfilter
4. Band rejectfilter
3. What are the advantages and disadvantages of FIRfilters?

Advantages:
1. FIR filters have exact linearphase.
2. FIR filters are alwaysstable.
3. FIR filters can be realized in both recursive and non-recursivestructure.
4. Filters with any arbitrary magnitude response can be tackled using FIRsequence.
Disadvantages:
1. For the same filter specifications the order of FIR filter design can be as high as 5 to 10 times that
in an IIRdesign.
2. Large storage requirement isrequirement
3. Powerful computational facilities required for theimplementation.
4. How one can design digital filters from analogfilters?

· Map the desired digital filter specifications into those for an equivalent analogfilter.
· Derive the analog transfer function for the analogprototype.
· Transform the transfer function of the analog prototype into an equivalent digital filter
transferfunction.
5. Mention the procedures for digitizing the transfer function of an analogfilter.

The two important procedures for digitizing the transfer function of an analog filter are
· Impulse invariancemethod.
· Bilinear transformationmethod.
. Approximation of derivatives
1. Distinguish between FIR filters and IIRfilters.
FIR filter IIR filter

These filters can be easily designed to have These filters do not have linear phase.
perfectly linear phase.
FIR filters can be realized recursively and non- IIR filters are easily realized recursively
recursively.
Greater flexibility to control the shape of their Less flexibility, usually limited to specific kind
magnitude response. of filters.
Errors due to round off noise are less severe in The round off noise in IIR filters is more.
FIR filters, mainly because feedback is not used.
2. What is the mapping procedure between S-plane & Z-plane in the method of mapping differentials? What
are its characteristics?
The mapping procedure between S-plane & Z-plane in the method of mapping of differentials is
given by
H(Z) =H(S)|S=(1-Z-1)/T
The above mapping has the following characteristics

a. The left half of S-plane maps inside a circle of radius ½ centered at Z= ½ in the Zplane.
b. The right half of S-plane maps into the region outside the circle of radius ½ in
the Z-plane.
c. The j .-axis maps onto the perimeter of the circle of radius ½ in theZ-plane.
3. Write a short note on pre-warping.
The effect of the non-linear compression at high frequencies can be compensated. When the desired
magnitude response is piece-wise constant over frequency, this compression can be compensated by
introducing a suitable pre-scaling, or pre-warping the critical frequencies by using the formula.
4. What are the advantages & disadvantages of bilineartransformation?

Advantages:
· The bilinear transformation provides one-to-onemapping.
· Stable continuous systems can be mapped into realizable, stable digitalsystems.
· There is noaliasing.
Disadvantage:
· The mapping is highly non-linear producing frequency, compression at highfrequencies.
· Neither the impulse response nor the phase response of the analog filter is preserved in a digital
filter obtained by bilineartransformation.
5. Distinguish analog and digitalfilters
Analog Filter Digital Filter

Constructed using active or passive components Consists of elements like adder, subtractor
and it is described by a differential equation and delay units and it is described by a
difference equation
Frequency response can be changed by Frequency response can be changed by
changing the components changing the filter coefficients
It processes and generates analog output Processes and generates digital output
Output varies due to external conditions Not influenced by external conditions
1) Explain about the Butterworth filter approximations
The classical method of analog filters design is Butterworth approximation. The Butterworth filters are also
known as maximally flat filters. Squared magnitude response of a Butterworth low-pass filter is defined as
follows

where - radian frequency, - constant scaling frequency, - order of the filter.
Some properties of the Butterworth filters are:
 gain at DC is equal to 1;
 has a maximum at
 The first derivatives of (3.1) are equal to zero at . This is why Butterworth filters are
known as maximally flat filters.
Using above properties the expression can be rearranged to the form
Given this expression can be written as follows
Function has poles and doesn't have any finite zeros. It is easy to see that if is a pole of
(3.2), then is also a pole of (3.2). In order to find the poles of transfer function that satisfy (3.2), we
have to select one pole from each pair of the poles of expression (3.2). As it was mentioned
before, the poles of a valid filter have to have negative real parts.
The poles of (3.2) can be found as roots of equation
Observing that , where stands for the odd number, roots of can be obtained as
solutions to the equation
The solution of (3.4) can be presented in the form
So, roots (3.5) are the poles of .
All poles lie on a circle of radius in the complex s-plane. Since the difference has the
same value for all roots, it can be concluded that the poles are equally spaced on the circumference.
Fig 3.1 Pole locations of the squared

magnitude response for the Butterworth
low-pass filters with orders N=6 and N=5.
Scaling frequency .
It is easy to see that for

the . Therefore, N roots of (3.5)
have the negative real parts; these are the
poles of . The remaining roots are
the poles of . Real parts in (3.5) are also never zero, so poles never fall to the imaginary axis. For the
Butterworth low-pass filters with odd orders, two of the poles have zero imaginary parts, so they fall on the
real axis in the s-plane. For the filters with even orders, all imaginary and real parts are nonzero.
The poles of Butterworth filter lie on left half of the s-plane, and they can be given as follows
2) Write down the advantages and disadvantages of Digital filters
Advantages
 It has linear phase response.

 Thermal and environmental variation cannot change the performance.
 It is so-called adaptive filter because the frequency response can be possible to adjust automatically
with implementation of programmable processor.
 It is possible to filter several input sequences without any hardware replication.
 All data can be stored
 It has repeatable performance unit to unit.
 Due to its operating level is at low frequency, it is used where the use of analog system is impractical.
Disadvantages
 In reality, the signal bandwidth of the digital sequence is much lower than the analog sequence. Signal
processing speed is one of the key factors of calculating the total device performance. Actually the
speed operation totally depends on the number of the arithmetic operation in the processor.
 Finite word-length effect, which results quantizing noise and round-off noise, is another major
drawback during computation.
 It needs much longer time to design and develop the digital sequences though it can be used on other
tasks or applications once developed. Ordinarily good support of computer aided design can convert
them into a enjoyable tasks.
3) Discuss magnitude characteristics of an analog Butterworth filter and give its pole locations
Ideal Frequency Response for a Butterworth Filter

Note that the higher the Butterworth filter order, the higher the number of cascaded stages there are within
the filter design, and the closer the filter becomes to the ideal “brick wall” response.
In practice however, Butterworth’s ideal frequency response is unattainable as it produces excessive passband
ripple.
Where the generalised equation representing a “nth” Order Butterworth filter, the frequency response is given
as:
Where: n represents the filter order, Omega ω is equal to 2πƒ and Epsilon ε is the maximum pass band gain,
(Amax). If Amax is defined at a frequency equal to the cut-off -3dB corner point (ƒc), ε will then be equal to one
and therefore ε2 will also be one. However, if you now wish to define Amax at a different voltage gain value, for
example 1dB, or 1.1220 (1dB = 20logAmax) then the new value of epsilon, ε is found by:
 Where:
 H0 = the Maximum Pass band Gain, Amax.
 H1 = the Minimum Pass band Gain.

Transpose the equation to give:
The Frequency Response of a filter can be defined mathematically by its Transfer Function with the standard
Voltage Transfer Function H(jω) written as:
 Where:
 Vout = the output signal voltage.
 Vin = the input signal voltage.
 j = to the square root of -1 (√-1)
 ω = the radian frequency (2πƒ)
Note: ( jω ) can also be written as ( s ) to denote the S-domain. and the resultant transfer function for a second-
order low pass filter is given as:
Normalized Low Pass Butterworth Filter Polynomials
To help in the design of his low pass filters, Butterworth produced standard tables of normalized second-order
low pass polynomials given the values of coefficient that correspond to a cut-off corner frequency of 1
radian/sec.
1 (1+s)
2 (1+1.414s+s2)

3 (1+s)(1+s+s2)
4 (1+0.765s+s2)(1+1.848s+s2)
5 (1+s)(1+0.618s+s2)(1+1.618s+s2)
6 (1+0.518s+s2)(1+1.414s+s2)(1+1.932s+s2)
7 (1+s)(1+0.445s+s2)(1+1.247s+s2)(1+1.802s+s2)
8 (1+0.390s+s2)(1+1.111s+s2)(1+1.663s+s2)(1+1.962s+s2)
9 (1+s)(1+0.347s+s2)(1+s+s2)(1+1.532s+s2)(1+1.879s+s2)
10 (1+0.313s+s2)(1+0.908s+s2)(1+1.414s+s2)(1+1.782s+s2)(1+1.975s+s2)
Filter Design – Butterworth Low Pass
Find the order of an active low pass Butterworth filter whose specifications are given as: Amax = 0.5dB at a pass
band frequency (ωp) of 200 radian/sec (31.8Hz), and Amin = 20dBat a stop band frequency (ωs) of 800
radian/sec. Also design a suitable Butterworth filter circuit to match these requirements.
Firstly, the maximum pass band gain Amax = 0.5dB which is equal to a gain of 1.0593(0.5dB = 20log A) at a
frequency (ωp) of 200 rads/s, so the value of epsilon ε is found by:
Secondly, the minimum stop band gain Amin = 20dB which is equal to a gain of 10 (20dB = 20log A) at a stop
band frequency (ωs) of 800 rads/s or 127.3Hz.
Substituting the values into the general equation for a Butterworth filters frequency response gives us the
following:
Since n must always be an integer ( whole number ) then the next highest value to 2.42 is n = 3, therefore a “a
third-order filter is required” and to produce a third-order Butterworth filter, a second-order filter stage
cascaded together with a first-order filter stage is required.
From the normalised low pass Butterworth Polynomials table above, the coefficient for a third-order filter is
given as (1+s)(1+s+s2) and this gives us a gain of 3-A = 1, or A = 2. As A = 1 + (Rf/R1), choosing a value for both
the feedback resistor Rf and resistor R1 gives us values of 1kΩ and 1kΩ respectively, ( 1kΩ/1kΩ + 1 = 2 ).
We know that the cut-off corner frequency, the -3dB point (ωo) can be found using the formula 1/CR, but we
need to find ωo from the pass band frequency ωp then,

So, the cut-off corner frequency is given as 284 rads/s or 45.2Hz, (284/2π) and using the familiar
formula 1/CR we can find the values of the resistors and capacitors for our third-order circuit.
Note that the nearest preferred value to 0.352uF would be 0.36uF, or 360nF.
Third-order Butterworth Low Pass Filter
and finally our circuit of the third-order low pass Butterworth Filter with a cut-off corner frequency of 284
rads/s or 45.2Hz, a maximum pass band gain of 0.5dB and a minimum stop band gain of 20dB is constructed as
follows.

4.
5.

1. IIR digital filters are of the following nature
a. Recursive
b. Non Recursive
c. Reversive
d. Non Reversive
ans-a
2. In IIR digital filter the present output depends on
a. Present and previous Inputs only
b. Present input and previous outputs only
c. Present input only
d. Present Input, Previous input and output
ans-d

3. The most common technique for the design of I I R Digital filter is
a. Direct Method
b. In direct method
c. Recursive method
d. non recursive method
ans-a
4. In the design a IIR Digital filter for the conversion of analog filter in to Digital domain the desirable property
is
a. The axis in the s - plane should map outside the unit circle in the z - Plane
b. The Left Half Plane(LHP) of the s - plane should map in to the unit circle in the Z -
Plane
c. The Left Half Plane(LHP) of the s-plane should map outside the unit circle in the z-
Plane
d. The Right Half Plane(RHP) of the s-plane should map in to the unit circle in the Z -
Plane
Ans-d
5.The s plane and z plane are related as

a. z = esT
b. z = e2sT
c. z = 2esT
d. z = esT/2
ANSWER: (a) z = esT
6. The IIR filter designing involves

a. Designing of analog filter in analog domain and transforming into digital domain
b. Designing of digital filter in analog domain and transforming into digital domain
c. Designing of analog filter in digital domain and transforming into analog domain
d. Designing of digital filter in digital domain and transforming into analog domain
ANSWER: (b) Designing of digital filter in analog domain and transforming into digital domain
7. For a system function H(s) to be stable

a. The zeros lie in left half of the s plane
b. The zeros lie in right half of the s plane
c. The poles lie in left half of the s plane
d. The poles lie in right half of the s plane
ANSWER: (c) The poles lie in left half of the s plane
8.IIR filter design by approximation of derivatives has the limitations

1) Used only for transforming analog high pass filters
2) Used for band pass filters having smaller resonant frequencies
3) Used only for transforming analog low pass filters
4) Used for band pass filters having high resonant frequencies

ANSWER: (c) 2 and 3 are correct
9.The filter that may not be realized by approximation of derivatives techniques are
1) Band pass filters
2) High pass filters
3) Low pass filters
4) Band reject filters

ANSWER: (b) 2 and 4 are correct
10. Which among the following represent/s the characteristic/s of an ideal filter?
a. Constant gain in pass band
b. Zero gain in stop band
c. Linear Phase Response
d. All of the above
ANSWER: (d) All of the above
FILL IN THE BLANKS
1. Filters exhibit their dependency upon the system design for the stability purpose
2. The impulse invariant method is obtained by
3. The transformation technique in which there is one to one mapping from s-domain to z-domain is
4. IIR filters have both

5. The poles of chebyshev filter lies on the
6. LTI IIR systems are stable if all the poles of the system function lie the unit circle
7. Bilinear transformation avoids in digitizing IIR filter
8. Impulse response of IIR filter have number of samples
9. IIR filter can have linear phase , if and only if ,it is both and
10. The advantage of direct form-II over direct form-I realization is needs
ANSWERS
1. IIR
2. Sampling the impulse response of an equivalent analog filter
3. Bilinear transformation method
4. poles and zeros
5. ellipse
6. inside

7. aliasing
8. infinite
9. stable ,physically realizable
10. less memory locations
UNIT IV
1. 2 MARKS QUESTIONS WITH ANSWERS
2. How phase distortion and delay distortion are introduced?
The phase distortion is introduced when the phase characteristics of a filter is nonlinear within the
desired frequency band. The delay distortion is introduced when the delay is not constant within the desired
frequency band.
3. What is mean by FIR filter?

The filter designed by selecting finite number of samples of impulse response h (n) obtained from
inverse Fourier transform of desired frequency response H(w) are called FIR filters
4. Write the steps involved in FIR filter design

Choose the desired frequency response Hd(w)
Take the inverse Fourier transform and obtain Hd(n) Convert the
infinite duration sequence Hd(n) to h(n) Take Z transform of h(n)
to get H(Z)
5. List the disadvantages of FIR FILTER

The duration of impulse response should be large to realize sharp cutoff filters. The non integral
delay can lead to problems in some signal processing applications.
6. Define necessary and sufficient condition for the linear phase characteristic of a FIR filter?
The phase function should be a linear function of w, which in turn requires constant group delay
and phase delay.
1. List the well-known design technique for linear phase FIR filter design?
Fourier series method and window method Frequency
sampling method
Optimal filter design method
2.What are the properties of FIR filter?

1. FIR filter is always stable.
2. A realizable filter can always be obtained.
3. FIR filter has a linear phase response.
3.What are the disadvantages of Fourier series method?

In designing FIR filter using Fourier series method the infinite duration impulse
response is truncated at n= ± (N-1/2).Direct truncation of the series will lead to fixed
percentage overshoots and undershoots before and after an approximated discontinuity in the
frequency response .
4. Give the desirable characteristics of the windows?
The desirable characteristics of the window are

1. The central lobe of the frequency response of the window should contain most of the
energy and should be narrow.
2. The highest side lobe level of the frequency response should be small.
3. The side lobes of the frequency response should decrease in energy rapidly as w tends to p.
5.State Frequency Warping
Because of the non-linear mapping: the amplitude response of digital IIR filter is expand
at lower frequencies and compressed at higher frequencies in comparison to the
analog filter.
1. Compare FIR and IIR filter
S.No FIR Filter IIR Filter
1. The impulse response of this filter is restricted The impulse response extends to infinite
to finite number of samples duration
2. FIR Filters have linear phase IIR filter don’t have linear phase
3. Always stable Not always stable
4. Greater flexibility Less flexibility

5. Errors due to roundoff noise are less severe IIR filters are more susceptible to errors due to
roundoff noise

2. What do you understand by Linear Phase Response in filters?
In linear phase filter ( ) α, the linear phase filter does not alter the shape of the
original signal. If phase response of the filter is nonlinear the output signal is
distorted one. In many cases Linear Phase filter is required throughout the pass
band of the filter to preserve the shape of the given signal within the pass band.
An IIR filter cannot produce a linear phase. The FIR filter can give linear phase,
when the impulse response of the filter is symmetric about its midpoint.
3. Explain various window techniques
(a) Rectangular
(b) Bartlett (or triangle)
(c) Hanning
(d) Harming
(e) Blackman
(f) Kaiser
where is modified zero-order Bessel function of the first kind given by
The main lobe width and first side lobe attenuation increase as we proceed down the window listed above.
An ideal lowpass filter with linear phase and cut off is characterized by
The corresponding impulse response is
Since this is symmetric about , if we change and use one of the windows listed above the will get linear
phase FIR filter. Transition width and minimum stopped attenuation are listed in the Table 9.3.
Window Transition Width Minimum stopband attenuation

Rectangular -21db
Bartlett -25dB
Hanning -44dB
Hamming -53dB
Blackman -74dB
Kaiser variable variable

4. Explain Linear Phase characteristics of FIR filters
Linear phase is a property of a filter, where the phase response of the filter is a linear
function of frequency. The result is that all frequency components of the input signal are shifted in
time (usually delayed) by the same constant amount, which is referred to as the phase delay. And
consequently, there is no phase distortion due to the time delay of frequencies relative to one another.
For discrete-time signals, perfect linear phase is easily achieved with a finite impulse response (FIR)
filter. Approximations can be achieved with infinite impulse response (IIR) designs, which are more
computationally efficient.
Linear-phase FIR filter can be divided into four basic types.
Type impulse response
I symmetric length is odd
II symmetric length is even
III anti-symmetric length is odd
IV anti-symmetric length is even
When h(n) is nonzero for 0 ≤ n ≤ N −1 (the length of the impulse response h(n) is N), then the
symmetry of the impulse response can be written as
h(n) = h(N − 1 − n)
and the anti-symmetry can be written as
h(n) = −h(N − 1 − n).
5.Why are FIR filters generally preferred over IIR filters in multi rate (decimating and interpolating)
systems?
Because only a fraction of the calculations that would be required to implement a decimating or interpolating
FIR in a literal way actually needs to be done.
Since FIR filters do not use feedback, only those outputs which are actually going to be used have to be
calculated. Therefore, in the case of decimating FIRs (in which only 1 of N outputs will be used), the other
N-1 outputs don’t have to be calculated. Similarly, for interpolating filters (in which zeroes are inserted
between the input samples to raise the sampling rate) you don’t actually have to multiply the inserted zeroes
with their corresponding FIR coefficients and sum the result; you just omit the multiplication-additions that
are associated with the zeroes (because they don’t change the result anyway.)
In contrast, since IIR filters use feedback, every input must be used, and every input must be calculated
because all inputs and outputs contribute to the feedback in the filter.
1. FIR filters ________

A. are non-recursive
B. do not adopt any feedback
C. are recursive
D. use feedback
a. A & B
b. C & D
c. A & D
d. B & C
ANSWER:(a) A & B
2. How is the sensitivity of filter coefficient quantization for FIR filters?

a. Low
b. Moderate
c. High
d. Unpredictable
ANSWER: (a) Low
3.The Chebyshev filters have
1) Flat pass band

2) Flat stop band
3) Equiripple pass band
4) Tapering stop band
a. 1 and 2 are correct

ANSWER: 2 and 3 are correct
4. Consider the assertions given below. Which among them is an advantage of FIR Filter?
a. Necessity of computational techniques for filter implementation

b. Requirement of large storage
c. Incapability of simulating prototype analog filters
d. Presence of linear phase response
ANSWER: Presence of linear phase response
5. In FIR filter design, which among the following parameters is/are separately controlled by using Kaiser
window?
a. Order of filter (M)

b. Transition width of main lobe
c. Both a and b
ANSWER: Both a and b
6. Which window function is also regarded as 'Raised-cosine window'?
a. Hamming window
b. Hanning window
c. Barlett window
d. Blackman window
ANSWER: Hanning window

7. In Barlett window, the triangular function resembles the tapering of rectangular window sequence
_______ from the middle to the ends.
a. linearly
b. elliptically
c. hyperbolically
d. parabolically
ANSWER: linearly
8. In FIR filters, which among the following parameters remains unaffected by the quantization effect?
a. Magnitude Response
b. Phase Characteristics
c. Both a and b
ANSWER: Phase Characteristics
9. In the frequency response characteristics of FIR filter, the number of bits per coefficient should be
_________in order to maintain the same error.
a. Increased
b. Constant
c. Decreased
ANSWER: Increased
10. A filter is said to be linear phase filter if the phase delay and group delay are _______
a. High
b. Moderate
c. Low
d. Constant
ANSWER: Constant
FILL IN THE BLANKS
1. In linear phase FIR filters filter is most versatile

2. FIR filters are always stable because its poles lies at
3. FIR filters have exactly phase
4. Width of the main lobe is wider than the in window design method
5. FIR filter is of length M then the order of FIR filter N is given as
6. Filters also known as feed forward filters
7. Transfer function of the FIR filter is given as
8. Width of main lobe of rectangular window is
9. Width of the transition region between pass band and stop band in H(W) increases with the
of the main lobe
10. Width of main lobe of blackmann window is
Answers
1. Type-I
2. Origin
3. Linear
4. Transition bandwidth
5. N=M-1
6. FIR
m N
7. ∑bk z-k/1+∑ak z-k
K=0 k=1
8. 4π/M
9. Width
10. 12π/M
UNIT V
1.Give the different quantization errors occur due to finite word length
registers in digital filters?
 Input quantization errors

 Coefficient quantization errors
 Product quantization errors
2. Mention the different quantization methods available for Finite Word Length Effects?
1. Truncation
2. Rounding
3. State truncation?
Truncation is a process of discarding all bits less significant than LSB that is retained
4. Define Rounding?
Rounding a number to b bits is accomplished by choosing a rounded result as the b
bit number closest number being unrounded.
5. List the two types of limit cycle behavior of DSP?

1. Zero limit cycle behavior
2. Over flow limit cycle behavior
1. Mention the methods to prevent overflow?

 Saturation arithmetic and
 Scaling
2. Define overflow oscillations

The overflow caused by adder makes the filter output to oscillate between
maximum amplitude limits and such oscillations is referred as overflow
oscillations
3. What do you understand by input quantization error?

.In digital signal processing, the continuous time input signals are converted into
digital by using b bit ADC. The representation of continuous signal amplitude by
a fixed digit produces an error, which is known as input quantization error.
4. Determine Dead Band of the Filter.

The Limit cycle occurs as a result of quantization effect in multiplication. The
amplitude of output during a limit cycle are confined to a range of values called
the dead band of the filter
5. Why Rounding is preferred to truncation in realizing digital filter?

• The quantization error due to rounding is independent of type arithmetic
• The mean of rounding error is zero
• The variance of rounding error is low
1. Distinguish between fixed point and floating point arithmetic
S.No Fixed Point Arithmetic Floating Point Arithmetic

1. Fast Operation Slow Operation
2. Relatively Economical More expensive because of
costlier hardware
3. Overflow occurs in addition Overflow does not arise
4. Used in small computers Used in large general purpose
computers
5. Small Dynamic Range Increased dynamic range
MA.Sohana Parveen, Assistant. Professor

2. Draw the quantization noise model for a I order System
3. What do you understand by (Zero input) Limit cycle oscillations?
When a stable IIR filter digital filter is excited by a finite sequence, that is constant, the
output will ideally decay to zero. However, the non-linearity due to finite precision
arithmetic operations often causes periodic oscillations to occur in the output. Such
oscillations occur in the recursive systems are called Zero input Limit Cycle Oscillation.
Normally oscillations in the absence of output u (k) =0 by equation given below is called
Limit cycle oscillations
Y[k] = -0.625.y [k-1] +u[k]
4. Explain decimation process
Decimation can be regarded as the discrete-time counterpart of sampling. Whereas

in sampling we start with a continuous-time signal x(t) and convert it into a
sequence of samples x[n], in decimation we start with a discrete-time signal x[n]
and convert it into another discrete-time signal y[n], which consists of sub-samples
of x[n]. Thus, the formal definition of M-fold decimation, or down-sampling, is
defined by Equation. In decimation, the sampling rate is reduced from Fs to Fs/M
by discarding M – 1 samples for every M samples in the original sequence

∞
y[ n]= v[nM ]=∑ h[k] x[nM -k]
k=-∞
5. write short notes on interpolation
Interpolation is the exact opposite of decimation. It is an information preserving

operation, in that all samples of x[n] are present in the expanded signal y[n]. The
mathematical definition of L-fold interpolation is defined by Equation and the
block diagram notation is depicted in below Figure Interpolation works by
inserting (L–1) zero-valued samples for each input sample. The sampling rate
therefore increases from Fs to LFs. With reference to Figure, the expansion process
is followed by a unique digital low-pass filter called an anti-imaging filter.
Although the expansion process does not cause aliasing in the interpolated signal,
it does however yield undesirable replicas in the signal’s frequency spectrum.
y[n] =L h[k]w[n-k]

1. Decimation is a process in which the sampling rate is __________.

a. enhanced
b. stable
c. reduced
d. unpredictable
ANSWER:(c) reduced
2. Anti-imaging filter with cut-off frequency ωc = π/ I is specifically used _______ upsampling

process for the removal of unwanted images.
a. Before
b. At the time of
c. After
d. All of the above
ANSWER: (c) After

3. The effects caused due to finite word lengths are
1) Coefficient quantization error

2) Adder overflow limit cycle
3) Round off noise
4) Limit cycles

ANSWER: All the four are correct
4. The error in the filter output that results from rounding or truncating calculations within the filter
is called
a. Coefficient quantization error

b. Adder overflow limit cycle
c. Round off noise
d. Limit cycles
ANSWER: Round off noise
5. To change the sampling rate for better efficiency in two or multiple stages, The decimation and
interpolation factors must be _________unity.
a. Less than
b. Equal to
c. Greater than
ANSWER: Greater than
6. How is/are the roundoff errors reduced in the digital FIR filter?
a. By representation of all products with double-length registers

b. By rounding the results after acquiring the final sum
c. Both a and b

7. Consider the assertions (steps) given below. Which among the following is a correct sequence of
designing steps for the sampling rate converters?
A. Computation of decimation/interpolation factor for each stage.

B. Clarification of anti-aliasing / anti-imaging filter requirements.
C. Designing of filter at each stage.
D. Calculation of optimum stages of decimation/ interpolation yielding maximum efficient
implementation.
a. A, B, C, D
b. C, A, D, B
c. D, A, B, C
d. B, D, A, C
ANSWER: B, D, A, C
8. How is the sampling rate conversion achieved by factor I/D?
a. By increase in the sampling rate with (I)

b. By filtering the sequence to remove unwanted images of spectra of original
signal
c. By decimation of filtered signal with factor D
d. All of the above
ANSWER: All of the above
9. Which is/are the correct way/s for the result quantization of an arithmetic operation?
a. Result Truncation
b. Result Rounding
c. Both a and b
10.the quantization noise can be reduced

a. by increasing step size
b. by reducing step size
c. by reducing no.of bits used for quantization

d. none
ANSWER: b
FILL IN THE BLANKS
1. occurs as a result of the quantization effects in multiplication

2. The conversion of continuous time input signal into digital value produces an error, which is
known as
3. Does not arise in floating point arithmetic
4. Is a process used to increase the sampling rate
5. The amplitudes of output during a limit cycle are confined to the range of values that is
called the of the filter
6. Filter is used in decimation process
7. In fixed point arithmetic round off error occurs only for
8. Arise at the output of multiplier
9. Is a process of discarding all bits less significant than LSB that is retained
10. Decimation is a process which is used to the sampling rate
Answers
1. Limit cycle
2. Input quantization error
3. Overflow
4. Interpolation
5. Dead band
6. Anti-aliasing
7. Addition
8. Product quantization error
9. Truncation
10. decrease

COURSE FILE CONTENTS

S.No. Topics Page No.
1 Vision, Mission, PEO’s, PO’s & PSO’S
2 Syllabus (University Copy)
3 Course Objectives, Course Outcomes And Topic Outcomes
4 Course Prerequisites
5 Course Information Sheet (CIS)
a). Course Description
b). Syllabus
c). Gaps in Syllabus
d). Topics beyond syllabus
e). Web Sources-References
f). Delivery / Instructional Methodologies
g). Assessment Methodologies-Direct
h). Assessment Methodologies –Indirect
i). Text books & Reference books
6 Micro Lesson Plan
7 Teaching Schedule
8 Unit Wise Hand Written notes
9 OHP/LCD SHEETS /CDS/DVDS/PPT (Soft/Hard copies)
10 University Previous Question papers
11 MID exam Descriptive Question Papers
12 MID exam Objective Question papers
13 Assignment topics with materials
14 Tutorial topics and Questions
15 Unit wise-Question bank
1 Two marks question with answers 5 questions
2 Three marks question with answers 5 questions
3 Five marks question with answers 5 questions
4 Objective question with answers 10 questions
5 Fill in the blanks question with answers 10 questions
16 Course Attainment
17 CO-PO Mapping
18 Beyond syllabus Topics with material
19 Result Analysis-Remedial/Corrective Action
20 Record of Tutorial Classes
21 Record of Remedial Classes
22 Record of guest lecturers conducted

Part – 2
S.NO TOPICS
1 Attendance Register/Teacher Log Book
2 Time Table
3 Academic calendar
4 Continuous Evaluation – marks (Test, Assignments etc)
5 Status Report Internal Exams & Syllabus coverage
6 Teaching Dairy/Daily Delivery Record Micro lesson Plan
7 Continuous Evaluation – MID marks
8 Assignment Evaluation-marks/Grades
9 Special Descriptive Tests Marks
10 Sample students descriptive answer sheets
11 Sample students assignment sheets

1. Vision, Mission, Program Educational Objectives (PEOs),

Program Outcomes (POs), Program Specific Outcomes
(PSOs).
VISION
To be recognized as a full-fledged center for learning and research in various fields of

Electronics and Communication Engineering through industrial collaboration and provide
consultancy for solving the real time problems
MISSION
 To inculcate a spirit of research and teach the students about contemporary

technologies in Electronics and Communication to meet the growing needs of the
industry.
 To enhance the practical knowledge of students by implementing projects based
on real time problems through industrial collaboration

PROGRAMME EDUCATIONAL OBJECTIVES (PEO’s)
After 3-5 years from the year of graduation, our graduates will,
1. With an exposure to the areas of VLSI, Embedded Systems, Signal / Image

Processing, Communications, Wave theory, and Electronic circuits in modern
electronics and communications environment.
2. Demonstrate the impact of Electronics and Communications Engineering on the
society, ethical, social and professional responsibilities/implications of their work.
3. With strong foundations in mathematical, scientific and engineering fundamentals
necessary to formulate, solve and analyze Electronics and communications.
Engineering problems.
4. Have strong communication inter-personal skills, multicultural adoptability and to
work effectively in multidisciplinary teams.
5. Engage life-long learning to become successful in their professional work.
PEO 1: To inculcate the adaptability skills into the students for design and use of
analog and digital circuits and communications and any other allied fields of
Electronics.
PEO 2: Ability to understand and analyze engineering issues in a broader perspective

with ethical responsibility towards sustainable development.
PEO 3: To develop professional skills in students that prepares them for immediate
employment and for lifelong learning in advanced areas of Electronics and
communications and related fields.
PEO 4: To equip with skills for solving complex real-world problems related to VLSI,
Embedded Systems, Signal/Image processing, Communications, and Wave
theory.
PEO 5: Graduates will make valid judgment, synthesize information from a range of
sources and communicate them in sound ways in order to find an
economically viable solution.
PEO 6: To develop overall personality and character with team spirit, professionalism,
integrity, and moral values with the support of humanities, social sciences and
physical educational courses.

PROGRAMME OUTCOMES (PO’s)
A graduate of the Electronics and Communication Engineering Program will demonstrate:
PO1: Engineering knowledge: Apply the knowledge of mathematics, science, engineering

fundamentals, and an engineering specialization to the solution of complex engineering
problems.
PO2: Problem analysis: Identify, formulate, review research literature, and analyze complex
engineering problems reaching substantiated conclusions using first principles of
mathematics, natural sciences, and engineering sciences.
PO3: 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.
PO4: Conduct investigations of complex problems: Use research-based knowledge and

research methods including design of experiments, analysis and interpretation of data, and
synthesis of the information to provide valid conclusions.
PO5: Modern tool usage: Create, select, and apply appropriate techniques, resources, and
modern engineering and IT tools including prediction and modeling to complex
engineering activities with an understanding of the limitations.
PO6: The engineer and society: Apply reasoning informed by the contextual knowledge to
assess societal, health, safety, legal and cultural issues and the consequent responsibilities
relevant to the professional engineering practice.
PO7: 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.
PO8: Ethics: Apply ethical principles and commit to professional ethics and responsibilities and
norms of the engineering practice.
PO9: Individual and team work: Function effectively as an individual, and as a member or
leader in diverse teams, and in multidisciplinary settings.
PO10: Communication: Communicate effectively on complex engineering activities with the

engineering community and with society at large, such as, being able to comprehend and
write effective reports and design documentation, make effective presentations, and give
and receive clear instructions.
PO11: Project management and finance: Demonstrate knowledge and understanding of the
engineering and management principles and apply these to one’s own work, as a member
and leader in a team, to manage projects and in multidisciplinary environments.
PO12: Life-long learning: Recognize the need for, and have the preparation and ability to engage
in independent and life-long learning in the broadest context of technological change.

PROGRAMME SPECIFIC OUTCOMES (PSO’s)
PSO 1: Problem Solving Skills – Graduate will be able to apply latest electronics
techniques and communications principles for designing of communications
systems.
PSO 2: Professional Skills – Graduate will be able to develop efficient and effective
Communications systems using modern Electronics and Communications
engineering techniques.
PSO 3: Successful Career – To produce graduates with a solid foundation in

Electronics and Communications engineering who will pursue lifelong
learning and professional development including post graduation.
PSO 4: The Engineer and Society– Ability to apply the acquired knowledge for the
advancement of society and self.

(EE721PE) DIGITAL SIGNAL PROCESSING
IV year B.Tech.EEE-I sem L/T/P/D C

3/-/-/- 3
UNIT -I
Introduction: Introduction to Digital Signal Processing: Discrete Tim signals &
Sequences, Linear Shift Invariant Systems, Stability, and Causality, Linear Constant
Coefficient Difference Equations, Frequency Domain Representation of Discrete Time
Signals and Systems
Realization of Digital Filters: Applications of Z — Transforms, Solution Difference
Equations of Digital Filters, System Function, Stability Criterion frequency Response of
Stable Systems, Realization of Digital Filters — Direct, Canonic, Cascade and Parallel
Forms.
UNIT -II
Discrete Fourier series: DFS Representation of Periodic Sequence properties of Discrete
Fourier Series, Discrete Fourier Transforms: Properties of DFT, Linear Convolution of
Sequences using DFT, Computation of D Over-Lap Add Method, Over-Lap Save Method,
Relation between DTFT, DFS, DFT and Z-Transform.
Fast Fourier Transforms: Fast Fourier Transforms (FFT) – Radix-2 Decimation-in-Time and
Decimation-in-Frequency FFT Algorithms, Inverse FFT, and FFT with General Radix-N.
UNIT- III
IIR Digital Filters: Analog filter approximations – Butter worth and Chebyshev, Design of IIR
Digital Filters from Analog Filters, Step and Impulse Invariant Techniques, Bilinear
Transformation Method, Spectral Transformations.
UNIT-IV
FIR Digital Filters: Characteristics of FIR Digital Filters, Frequency Response, Design of FIR
Filters: Fourier Method, Digital Filters using Window Techniques, Frequency Sampling
Technique, Comparison of IIR & FIR filters
UNIT-V
Multi rate Digital Signal Processing: Introduction, Down Sampling Decimation, Up sampling,
Interpolation, Sampling Rate Conversion. Finite Word Length Effects: Limit cycles, Overflow
Oscillations, Round-off Noise in IIR Digital Filters, Computational Output Round Off Noise,
Methods to Prevent Overflow, Trade Off Between Round Off and Overflow Noise, Dead
Band Effects.
TEXT BOOKS
 Digital Signal Processing, Principles, Algorithms, and Applications John G. Proakis,
Dimitris G. Manolakis, Pearson Education / PHI, 2007.
 Discrete Time Signal Processing — A. V. Oppenheim and R.W Schaffer, PHI,
2009 Fundamentals of Digital Signal Processing — Loney Ludeman, John Wiley, 2009
REFERENCE BOOKS
 Digital Signal Processing — Fundamentals and Applications — Li Tan, Elsevier, 2008
 Fundamentals of Digital Signal Processing using MATLAB — Robert Schilling, Sandra L.

Harris, Thomson, 2007
 Digital Signal Processing — S.Salivahanan, A.Vallavaraj and C.Gnanapriya, TMH, 2009
 Discrete Systems and Digital Signal Processing with MATLAB — Taan EIAIi. CRC press,
2009.
 Digital Signal Processing – A Practical approach, Emmanuel C Ifeachor and Barrie W.
Jervis, 2nd Edition, Pearson Education, 2009
 Digital Signal Processing – Nagoor Khani, TMG, 2012
GAPS IN THE SYLLABUS
1.DISCRETE TIME FOURIER TRANSFORM
Discrete Time Fourier Transform of an aperiodic discrete time signal
Given a general aperiodic signal of finite duration, that is; for some integer N,
. From this aperiodic signal we can construct a periodic signal for which
is one period. As we chose period N to be larger than the duration of , is identical
to . As the period , for any finite value of n.
The Fourier series representation of is :
Since over a period that includes the interval , it is convenient to choose the
interval of summation to be this period, so that can be replaced by in the summation.
Therefore,
Another way of representing DTFT of a periodic discrete signal

In continuous time, the fourier transform of is an impulse at .However in discrete
time ,for signal the discrete time fourier transform is periodic in with period
. The DTFT of is a train of impulses at

i.e Fourier Transform can be written as :
Consider a periodic sequence x[n] with period N and with fourier series representation
Then discrete time Fourier Transform of a periodic signal x[n] with period N can be written as :
2.APPLICATIONS OF DIGITAL FILTERS

The applications of digital filters are
1. Noise suppression
(a) imaging devices (medical, etc)
(b) biosignals (heart, brain)
(c) signals stored on analog media (tapes)
2. Enhancement of selected frequency ranges
(a) equalizers for audio systems (increasing the bass)
(b) edge enhancement in images
3. Removal or attenuation of selected frequencies
(a) removing the DC component of a signal
(b) removing interferences at a specific frequency, for example those caused by power supplies
4. Bandwidth limiting
(a) anti-aliasing filters for sampling
(b) ensuring that a transmitted signal occupies only its alloted frequency band.
5. Special operations
(a) differentiation
(b) integration
(c) Hilbert transform
6. Simulation/Modeling
(a) simulating communication channels
(b) modeling human auditory system
TOPICS BEYOND SYLLABUS
1.ARCHITECTURE OF TMS320C5X PROCESSOR
2.WAVELET TRANSFORM
A wavelet is a wave-like oscillation with an amplitude that begins at zero, increases, and then
decreases back to zero. It can typically be visualized as a "brief oscillation" like one recorded by
a seismograph or heart monitor. Generally, wavelets are intentionally crafted to have specific
properties that make them useful for signal processing. Using a "reverse, shift, multiply and
integrate" technique called convolution, wavelets can be combined with known portions of a
damaged signal to extract information from the unknown portions.
All wavelet transforms may be considered forms of time-frequency representation for continuous-
time (analog) signals and so are related to harmonic analysis. Almost all practically useful discrete
wavelet transforms use discrete-time filterbanks. These filter banks are called the wavelet and
scaling coefficients in wavelets nomenclature. These filterbanks may contain either finite impulse
response (FIR) or infinite impulse response (IIR) filters. The wavelets forming a continuous
wavelet transform (CWT) are subject to the uncertainty principle of Fourier analysis respective
sampling theory: Given a signal with some event in it, one cannot assign simultaneously an exact
time and frequency response scale to that event. The product of the uncertainties of time and
frequency response scale has a lower bound. Thus, in the scaleogram of a continuous wavelet
transform of this signal, such an event marks an entire region in the time-scale plane, instead of
just one point. Also, discrete wavelet bases may be considered in the context of other forms of the
uncertainty principle.
Wavelet transforms are broadly divided into three classes: continuous, discrete and
multiresolution-based.
Continuous wavelet transforms (continuous shift and scale parameters)
In continuous wavelet transforms, a given signal of finite energy is projected on a continuous
family of frequency bands (or similar subspaces of the Lp function space L2(R)). For instance, the
signal may be represented on every frequency band of the form [f, 2f] for all positive frequencies
f > 0. Then, the original signal can be reconstructed by a suitable integration over all the resulting
frequency components.
The frequency bands or subspaces (sub-bands) are scaled versions of a subspace at scale 1. This
subspace in turn is in most situations generated by the shifts of one generating function ψ in
L2(R), the mother wavelet.
Discrete wavelet transforms (discrete shift and scale parameters)
It is computationally impossible to analyze a signal using all wavelet coefficients, so one may
wonder if it is sufficient to pick a discrete subset of the upper halfplane to be able to reconstruct
a signal from the corresponding wavelet coefficients. One such system is the affine system for
some real parameters a > 1, b > 0. The corresponding discrete subset of the halfplane consists of
all the points (am, namb) with m, n in Z.
Multiresolution based discrete wavelet transforms
Fig- D4 wavelet
In any discretised wavelet transform, there are only a finite number of wavelet coefficients for
each bounded rectangular region in the upper halfplane. Still, each coefficient requires the
evaluation of an integral. In special situations this numerical complexity can be avoided if the
scaled and shifted wavelets form a multiresolution analysis. This means that there has to exist an
auxiliary function, the father wavelet φ in L2(R), and that a is an integer. A typical choice is a = 2
and b = 1. The most famous pair of father and mother wavelets is the Daubechies 4-tap wavelet.
UNIT-I
Learning Objectives
• Why process signals digitally?
• Definition of a real-time application.
• Why use Digital Signal Processing
processors?
• What are the typical DSP algorithms?
• Parameters to consider when choosing
a DSP processor.
Why go digital?
• Digital signal processing techniques are now so
powerful that sometimes it is extremely
difficult, if not impossible, for analogue signal
processing to achieve similar performance.
• Examples:
– FIR filter with linear phase.
– Adaptive filters.
Why go digital?
• Analogue signal processing is achieved
by using analogue components such as:
– Resistors.
– Capacitors.
– Inductors.
• The inherent tolerances associated with
these components, temperature,
voltage changes and mechanical
vibrations can dramatically affect the
effectiveness of the analogue circuitry.
Why go digital?
• With DSP it is easy to:
– Change applications.
– Correct applications.
– Update applications.
• Additionally DSP reduces:
– Noise susceptibility.
– Chip count.
– Development time.
– Cost.
– Power consumption.
Why NOT go digital?
• High frequency signals cannot be
processed digitally because of two
reasons:
–Analog to Digital Converters, ADC
cannot work fast enough.
–The application can be too
complex to be performed in real-
time.
Real-time processing
• DSP processors have to perform tasks in
real-time, so how do we define real-time?
• The definition of real-time depends on the
application.
• Example: a 100-tap FIR filter is performed
in real-time if the DSP can perform and
complete the following operation between
two samples:
Real-time processing
Waiting Time
Processing Time
n n+1
Sample Time
• We can say that we have a real-time

application if:
– Waiting Time  0
Why do we need DSP processors?
• Why not use a General Purpose Processor
(GPP) such as a Pentium instead of a DSP
processor?
– What is the power consumption of a
Pentium and a DSP processor?
– What is the cost of a Pentium and a DSP
processor?
Why do we need DSP processors?
• Use a DSP processor when the following

are required:
– Cost saving.
– Smaller size.
– Low power consumption.
– Processing of many “high” frequency
signals in real-time.
Introduction to DSP
Analog system Discrete system
Block diagram of DSP system
Discrete Time Signals & Sequences
• Digital signals are discrete in both time (the
independent variable) and amplitude (the
dependent variable). Signals that are discrete
in time but continuous in amplitude are
referred to as discrete-time signals.
Linear-Time Invariant System
• Special importance for their mathematical
tractability
• Most signal processing applications involve LTI
systems
• LTI system can be completely characterized by
their impulse response
Stability
Causality
Linear Constant-Coefficient Difference
Equations
• A linear constant-coefficient difference
equation of order N looks like:
• All solutions y[n] can be expressed as a sum

yh[n]+yp[n]
Frequency Response
Applications of z-transform
• Inputs and outputs are related by difference
equations and z-transform techniques are
used to solve those difference equations
• Voice transmission
• Transmission systems
Cascade form
Parallel form
UNIT-II
23
Discrete Fourier Series
~
x[n]  ~
x[n  rN]
~
• Given a periodic sequence x[n] with period N so that
~ 1 ~
x[n]   Xk e j2  / Nkn
N k
• The Fourier series representation can be written as
e j2  / Nk  mNn  e j2  / Nkne j2 mn  e j2  / Nkn
• The Fourier series representation of continuous-time

periodic signals require infinite many complex exponentials
• Not that for discrete-time periodic signals we have
N 1
 X k e
~ 1 ~ j  2 / N kn
x [ n] 
N k 0
• Due to the periodicity of the complex exponential we only

need N exponentials for discrete time Fourier series
Discrete Fourier Series Pair
~ 1 N1 ~
x[n]   Xk e j2  / Nkn
N k 0
• A periodic sequence in terms of Fourier series coefficients
N 1
X k    ~
~
x [n]e  j 2 / N kn
n 0
• The Fourier series coefficients can be obtained via
W N  e  j  2 / N 
• For convenience we sometimes use
~ N 1
~
Xk    x[n]Wkn
N
• Analysis equation n0
~ 1 N 1 ~
x[n]   Xk WNkn
• Synthesis equation
N k 0
25
Example 1
~

1 n  rN
x[n]   n  rN  
r   0 else
• DFS of a periodic impulse train
~ N 1 N 1
Xk    ~x[n]e j2 / Nkn 
n0
 [n]e
n0
 j2  / N kn
 e  j2  / Nk 0  1
• Since the period the signal is N

~

1 N 1 j2  / Nkn
x[n]   n  rN   e
r   N k 0
• We can represent the signal with the DFS

coefficients as
Example 2
• DFS of an periodic rectangular
• pulse train
 j  4k / 10  sin k / 2

 j  2 / 10k 5
4

X k    e  j 2 / 10kn 
~ 1 e
 e
n 0 1  e  j 2 / 10k sin k / 10
• The DFS coefficients

Properties of DFS
~ ~
x1 n DFS
 X1 k 
• Linearity ~ ~
x n
2 DFS
 X2 k 
~ ~
a~
x1 n  b~
x2 n  aX1 k   bX2 k 
DFS

~ ~
x n DFS
 Xk 
~  j2 km / N~
• Shift of a Sequence x n  m DFS
  e Xk 
~
e j2 nm / N~
x n DFS
 Xk  m
~ ~
• Duality x n DFS
 Xk 
~
Xn DFS
 N~
x  k 
Symmetry Properties
Symmetry Properties Cont’d
Periodic Convolution
• Take two periodic sequences
~ ~
 X1 k 
~ ~
 X2 k 
• Let’s form the product
X 3 k   X 1 k X 2 k 
~ ~ ~
• The periodic sequence with given DFS can be written as

N 1
~
x3 n   ~x m~x n  m
m0
1 2
• Periodic convolution is commutative

N 1
x3 n   ~
~ x2 m~
x1 n  m
m 0
Periodic Convolution Cont’d
N 1
~
x3 n  ~ ~
x
m0
m
 1 x2 n  m
~ N 1
 N 1 ~ ~  kn
X3 k    
  x1[m]x2 [n  m]
WN
n0  m0 
• Substitute periodic convolution into the DFS equation

~ N 1
~  N1 ~ kn 
X3 k    x1[m] 2
 x [n  m]WN 
m0  n0 
• Interchange summations
N 1
N X 2 k 
km ~
 2
~
x [
n 0
n  m ]
W kn
N  W
• The inner sum is the DFS of shifted sequence

N 1
 N 1 ~ kn 
N 1
X 3 k    x1 [m]  x 2 [n  m]WN    ~ x1 [m]WNkm X 2 k  X 1 k X 2 k 
~ ~ ~ ~ ~
m 0  n 0  m 0
• Substituting
Graphical Periodic Convolution
The Fourier Transform of Periodic
Signals
• Periodic sequences are not absolute or square summable
– Hence they don’t have a Fourier Transform
• We can represent them as sums of complex exponentials: DFS

• We can combine DFS and Fourier transform
~ j 2 ~ 
  2k 

Xe   Xk    
k   N  N 
• Fourier transform of periodic sequences
– Periodic impulse train with values proportional to DFS coefficients
– This is periodic with 2 since DFS is periodic
• The inverse transform can be written as
1 2    ~ j jn 1 2     2 ~ 
2  X e e d 
 
2   Xk   
2k  jn
e d
k   N  N 
0   0  
2 k
1  ~ 2    2k  jn 1 N1 ~
Xk  e d   Xk e
j n

N k   0  
  
 N  N k 0
N
Relation between Finite-length and Periodic Signals
• Consider finite length signal x[n] spanningfrom 0 to N-1 

x[n]  x[n]  ~
~ p[n]  x[n]   n  rN   xn  rN
r   r  
• Convolve with periodic impulse train
• The Fourier transform of the periodic sequence is
2  2k 
       

~ ~
X e j  X e j  P e j   X e j     
k   N  N 
2  j 2Nk  
       2k 

~
X e j  X  e  
k   N   N 
• This implies that
~  j 2Nk 
Xk   X e   X e j
   
2 k
  N
• DFS coefficients of a periodic signal can be thought as equally spaced samples of

the Fourier transform of one period
• Consider the Example
following sequence
1 0  n  4
x[n]  
0 else
• The Fourier transform

sin5 / 2
Xe j   e  j2
sin / 2
~ sink / 2
Xk   e  j4 k / 1 0
sink / 10 
• The DFS coefficients

Discrete Fourier Transform (DFT)
• The DFT provides uniformly spaced samples

of the Discrete-Time Fourier Transform
(DTFT)
• DFT definition:
N 1 2nk 2nk
j 1 N 1
X [k ]   x[n]e x[n]   X [k ]e
j
N N
n 0 N n 0
• Requires N2 complex multiplies and N(N-1)

complex additions
Faster DFT computation?
• Take advantage of the symmetry and periodicity
of the complex exponential (let WN=e-j2/N)
– symmetry: WNk [ N n ]  WN kn  (WNkn )*
– periodicity: WNkn  WNk [ n  N ]  WN[ k  N ]n
• Note that two length N/2 DFTs take less
computation than one length N DFT: 2(N/2)2<N2
• Algorithms that exploit computational savings are
collectively called Fast Fourier Transforms
Decimation-in-Time Algorithm
• Consider expressing DFT with even and odd
input samples:
N 1
X [k ]   x[n]WNnk
n 0
  x[ n
n even
]W N 
nk
 x[
n odd
n ]W nk
N
N 1 N 1
  x[2r ](WN2 ) rk  WNk  x[2r  1](WN2 ) rk
2 2
r 0 r 0
N 1 N 1
  x[2r ]W 
2 2
rk
N /2 W k
N x[ 2 r  1]W rk
N /2
r 0 r 0
DIT Algorithm (cont.)
• Result is the sum of two N/2 length DFTs
X [k ]  G
[k ]  WNk  H
 [k ]
N/2 DFT N/2 DFT
of even samples of odd samples
• Then repeat decomposition of N/2 to N/4 DFTs, etc.
x[0,2,4,6] N/2
DFT
X[0…7]
x[1,3,5,7] N/2
DFT
WN0... 7
Detail of “Butterfly”
• Cross feed of G[k] and H[k] in flow diagram is
called a “butterfly”, due to shape
W Nr
or simplify:
WN( r  N 2 ) W Nr -1
( WNr )
8-point DIT-FFT Diagram
x[0,4,2,6,
X[0…7]
1,5,3,7]
W N0 1
W N0 1
W N0 1 WN2 1
W N0 1
W N0 1 W N1 1
W N0 1 WN2 1
W N0 1 WN2 W N3
1 1
8-point DIF-FFT Diagram
UNIT-III
Chebyshev 1 Approximation:
The Chebyshev 1 approximation for an ideal lowpass filter
has equal-valued ripples in the passband . It is known as
minimax approximation and also known as the equiripple
approximation.
The magnitude squared function of Chebyshev

approximation:
where Cn() is the Chebyshev polynomial of degree n. It is

defined by
For n = 2, 3, 4, 5, these polynomials are
Chebyshev II Approximation: (or inverse Chebyshev
filters)
Maximally flat at w=0; decreases monotonically as the
frequency increases and has an equiripple response in the
stopband.
The magnitude square function of the inverse Chebyshev

low pass filter: |H(j)|2 =
Chebyshev I
Chebyshev II
UNIT-IV
Frequency response of FIR digital filters
 we consider the FIR filters in which the impulse
response h[n] are assumed to be symmetric or
antisymmetric.
 Since the order of the polynomial in each of these two
types can be either odd or even, we have four types of

filters with different properties.
Linear phase FIR filters:

Type 1: The coefficients are symmetric, i.e., h[n] = h[Nn],
and the order N is even.
Type III: The coefficients are antisymmetric, i.e.,

h[n] =  h[Nn], and the order N is even.
Type IV: The coefficients are antisymmetric, i.e.,

h[n] =  h[Nn], and the order N is odd.
Frequency Sampling Method
In this approach we are given H(K) we need to
find H(Z)
This is an interpolation problem and the solution

is given in the DFT part of the course
1 N 1 1  zN
H ( z )   H (k ). 2
N k 0 j k
1  e N .z 1
Comparison of FIR & IIR digital filters
UNIT-IV
Frequency response of FIR digital filters
 we consider the FIR filters in which the impulse
response h[n] are assumed to be symmetric or
antisymmetric.
 Since the order of the polynomial in each of these two
types can be either odd or even, we have four types of

filters with different properties.

Type 1: The coefficients are symmetric, i.e., h[n] = h[Nn],
and the order N is even.
Type III: The coefficients are antisymmetric, i.e.,

h[n] =  h[Nn], and the order N is even.
Type IV: The coefficients are antisymmetric, i.e.,

h[n] =  h[Nn], and the order N is odd.
Frequency Sampling Method
In this approach we are given H(K) we need to
find H(Z)
This is an interpolation problem and the solution

is given in the DFT part of the course
1 N 1 1  zN
H ( z )   H (k ). 2
N k 0 j k
1  e N .z 1
Comparison of FIR & IIR digital filters
Thank you
SET-1
III B.TECH - II SEMESTER EXAMINATIONS, APRIL/MAY, 2011
(COMMON TO EEE, ECE, EIE, ETM, ICE)
Time: 3hours Max. Marks: 80
Answer any FIVE questions
All Questions Carry Equal Marks
1. a) Define an LTI System and show that the output of an LTI system is given by the convolution of
Input sequence and impulse response.
b) Prove that the system defined by the following difference equation is an LTI system y(n) =
x(n+1)- 3x(n)+x(n-1) ; n≥0. [8+8 ]
2. a) Define DFT and IDFT. State any Four properties of DFT.
b) Find 8-Point DFT of the given time domain sequence x(n) = {1, 2, 3, 4}. [8+8]
3. a) Derive the expressions for computing the FFT using DIT algorithm and hence draw the standard
butterfly structure.
b) Compare the computational complexity of FFT and DFT. [8+8]
4. Discuss and draw various IIR realization structures like Direct form – I, Direct form-II, Parallel and
cascade forms for the difference equation given y(n) = - 3/8 Y(n-1) + 3/32 y(n-2) + 1/64 y(n-3) + x(n)
+ 3 x(n-1) + 2 x(n-2).
5. a) Compare Butterworth and Chebyshev approximation techniques.
b) Design a Digital Butterworth LPF using Bilinear transformation technique for the following
specifications 0.707 ≤ | H(w) | ≤ 1 ; 0 ≤ w ≤ 0.2π | H(w) | ≤ 0.08 ; 0.4 π ≤ w ≤ [ 8+8]
6. a) Compare FIR and IIR filters
b) Design an FIR Digital High pass filter using Hamming window whose cut off freq is 1.2 rad/s
and length of window N=9. [8+8]
7. a) Define Multirate systems and Sampling rate conversion
b) Discuss the process of n Decimation by a factor D and explain how the aliasing effect can be
eliminated. [8+8]
8. Discuss various Modified Bus structures of Programmable DSP Processors.[16]

SET-2
1. a) Write short notes on classification of systems.
b) Derive BIBO stability criteria to achieve stability of a system. [8+8]
2 .a) Define DFS. State any Four properties of DFS.
b) Find the IDFT of the given sequence x(K) = {2, 2-3j, 2+3j, -2}. [8+8]
3.a) Find X(K) of the given sequence x(n) = { 1,2,3,4,4,3,2,1}using DIT- FFT algorithm
b) Compare the computational complexity of FFT and DFT. [8+8]
4. What are the various basic building blocks in realization of Digital Systems and hence discuss
transposed form realization structures.
5. a) Compare Impulse Invariant and Bilinear transformation techniques.
b) Compute the poles of an Analog Chebyshev filter TF that satisfies the Constraints 0.707 ≤ |
H(jΩ)| ≤ 1 ; 0 ≤ Ω≤ 2 | H(jΩ)| ≤ 0.1 ; Ω ≥ 4 and determine Ha(s) and hence obtain H(z) using Bilinear
transformation. [16]
6. a) Derive the conditions to achieve Linear Phase characteristics of FIR filters
b) Design an FIR Digital Low pass filter using Hanning window whose cut off freq is 2 rad/s and
length of window N=9. [8+8]
7. a) Discuss the implementation of Polyphase filters for Interpolators with an example
b) Discuss the sampling rate conversion by a factor I/D with the help of a Neat block Diagram.
[8+8]
8. Write short notes on:
a) VLIW Architecture of Programmable Digital Signal Processors
b) Multiplier and Multiplier Accumulator

SET-3
1. a) Discuss various discrete time sequences.
b) Give the Basic block diagram of Digital Signal Processor. [8+8]
2. a) Define DFS. State any Four properties of DFS.
b) Find the IDFT of the given sequence x(K) = {2, 2-3j, 2+3j, -2}. [8+8]
3.a) Find IFFT of the given X(K) = { 1,2,3,4,4,3,2,1}using DIF algorithm
b) Bring out the relationship between DFT and Z-transform. [8+8]
4. a) Define Z-Transform and List out its properties.
b) Discuss Direct form, Cascade and Linear phase realization structures of FIR filters. [8+8 ]
5. a) Discuss digital and analog frequency transformation techniques.
b) Discuss IIR filter design using Bilinear transformation and hence discuss frequency warping
effect. [8+8]
6. a) Compare various windowing functions.
b) Design an FIR Digital Low pass filter using rectangular window whose cut off freq is 2 rad/s and
length of window N=9. [8+8]
7. a) Define Interpolation and Decimation. List out the advantages of Sampling rate conversion.
b) Discuss the sampling rate conversion by a factor I with the help of a Neat block Diagram.[8+8]
8. a) Discuss Various Addressing modes of Programmable Digital Signal Processors.
b) Give the Internal Architecture of TMS320C5X 16 bit fixed point processor.[ 8+8]
SET-4
1. a) Define Linearity, Time Invariant, Stability and Causality.
b) The discrete time system is represented by the following difference equations in which x(n) is
input and y(n) is output. Y(n) = 3y 2 (n-1)- nx(n)+4x(n-1)-2x(n-1). [8+8]
2. a) Define Convolution. Compare Linear and Circular Convolution techniques.
b) Find the Linear convolution of the given two sequences x(n)={1,2} and h(n) ={1,2,3} using
DFT and IDFT. [8 +8]
3. a) Develop DIT-FFT algorithm and draw signal flow graphs for decomposing the DFT for N=6 by
considering the factors for N = 6 = 2.3.
b) Bring out the relationship between DFT and Z-transform. [8+8]
4. a) Discuss transposed form structures with an example.
b) Discuss Direct form, Cascade realization structures of FIR filters. [8+8]
5. a) Discuss digital and analog frequency transformation techniques.
b) Discuss IIR filter design using Impulse Invariant transformation and list out its advantages and
Limitations. [8+8]
6. a) Compare various windowing functions
b) Design an FIR Digital Band pass filter using rectangular window whose upper and lower cut off
freq.’s are 1 & 2 rad/s and length of window N = 9. [8+8]
7. a) Define Interpolation and Decimation.
b) Discuss the sampling rate conversion by a factor I/D with the help of a Neat block Diagram. [8
+8]
8. a) Write a short notes on On-Chip peripherals of Programmable DSP’s.
b)Give the Internal Architecture of TMS320C5X 16 bit fixed point processor. [8+8]
SET-1
1. (a) Discuss impulse invariance method of deriving IIR digital filter from corresponding analog
filter.
(b) Use the Bilinear transformation to convert the analog filter with system function H(S) = S +
0.1/(S + 0.1)2 + 9 into a digital IIR filters. Select T = 0.1 and compare the location of the zeros in
H(Z) with the locations of the zeros obtained by applying the impulse invariance method in the
conversion of H(S). [8+8]
2. (a) Design a high pass filter using hamming window with a cut-off frequency of 1.2 radians/second
and N=9
(b) Compare FIR and IIR filters. [10+6]
3. (a) For each of the following systems, determine whether or not the system is i. stable ii. causal iii.
linear iv. shift-invariant. A. T[x(n)] = x(n − n0 ) B. T [x(n)] = e x (n) C. T[x(n)] = a x(n) + b. Justify
your answer.
(b) A system is described by the difference equation y(n)-y(n-1)-y(n-2) = x(n1). Assuming that the
system is initially relaxed, determine its unit sample response h(n). [8+8]
4. (a) Implement the decimation in time FFT algorithm for N=16.
(b) In the above Question how many non - trivial multiplications are R e q u i r e d .
5. (a) Discuss the frequency-domain representation of discrete-time systems and signals by deriving
the necessary relation.
(b) Draw the frequency response of LSI system with impulse response h(n) = a n u(−n) (|a| < 1)
6. (a) Describe how targets can be decided using RADAR
(b) Give an expression for the following parameters relative to RADAR i. Beam width ii.
Maximum unambiguous range
(c) Discuss signal processing in a RADAR system. [6 +6+4]
7. (a) An LTI system is described by the equation y(n)=x(n)+0.81x(n-1)- 0.81x(n2)-0.45y(n-2).

Determine the transfer function of the system. Sketch the poles and zeroes on the Z-plane.
(b) Define stable and unstable systems. Test the condition for stability of the first-order IIR filter
governed by the equation y(n)=x(n)+bx(n1). [8+8]
8. (a) Compute Discrete Fourier transform of the following finite length sequence considered to be of
length N. i. x(n) = δ(n + n0 ) where 0 < n0 < N ii. x(n) = a n where 0 < a < 1.
(b) If x(n) denotes a finite length sequence of length N, show that x((−n))N =x((N − n))N . [8+8]
SET-2
1. (a) Design a high pass filter using hamming window with a cut-off frequency of 1.2 radians/second
and N=9
(b) Compare FIR and IIR filters. [10+6]
2. (a) Describe how targets can be decided using RADAR
(b) Give an expression for the following parameters relative to RADAR i. Beam width ii.
Maximum unambiguous range (c) Discuss signal processing in a RADAR system. [6 +6+4]
3. (a) An LTI system is described by the equation y(n)=x(n)+0.81x(n-1)- 0.81x(n2)-0.45y(n-2).

Determine the transfer function of the system. Sketch the poles and zeroes on the Z-plane.
(b) Define stable and unstable systems. Test the condition for stability of the first-order IIR filter
governed by the equation y(n)=x(n)+bx(n1). [8+8]
4. (a) Compute Discrete Fourier transform of the following finite length sequence considered to be of
length N. i. x(n) = δ(n + n0 ) where 0 < n0 < N ii. x(n) = a n where 0 < a < 1.
(b) If x(n) denotes a finite length sequence of length N, show that x((−n))N = x((N − n))N . [8+8]
5. (a) For each of the following systems, determine whether or not the system is i. stable ii. causal iii.
linear iv. shift-invariant. A. T[x(n)] = x(n − n0 ) B. T [x(n)] = e x (n) C. T[x(n)] = a x(n) + b. Justify
your answer
(b) A system is described by the difference equation y(n)-y(n-1)-y(n-2) = x(n1). Assuming that the
system is initially relaxed, determine its unit sample response h(n). [8+8]
6. (a) Discuss the frequency-domain representation of discrete-time systems and signals by deriving
the necessary relation.
(b) Draw the frequency response of LSI system with impulse response h(n) = a n u(−n) (|a| < 1)
[8+8]
7. (a) Implement the decimation in time FFT algorithm for N=16.
(b) In the above Question how many non - trivial multiplications are required.
8. (a) Discuss impulse invariance method of deriving IIR digital filter from corresponding analog
filter. (b) Use the Bilinear transformation to convert the analog filter with system function H(S) =
S + 0.1/(S + 0.1)2 + 9 into a digital IIR filters. Select T = 0.1 and compare the location of the zeros in
H(Z) with the locations of the zeros obtained by applying the impulse invariance method in the
conversion of H(S). [8+8]

DSP PDF

Uploaded by

Document Information

Original Title

Copyright

Available Formats

Share this document

Share or Embed Document

Sharing Options

Did you find this document useful?

Is this content inappropriate?

Copyright:

Available Formats

DSP PDF

Uploaded by

Copyright:

Available Formats

COURSE FILE

Subject (Name) : Digital signal processing

Name (of the Faculty Member) : P.USHA

Designation : Asst. Prof.

DIGITAL SIGNAL PROCESSING

3) Course Objectives, Course Outcomes and Topic Outcomes

1. Construct time, frequency and Z -transform analysis on signals and systems.

P.USHA, Assistant. Professor,dept of ECE,KGRCET

P.USHA, Assistant. Professor,dept of ECE,KGRCET

P.USHA, Assistant. Professor,dept of ECE,KGRCET

P.USHA, Assistant. Professor,dept of ECE,KGRCET

P.USHA, Assistant. Professor,dept of ECE,KGRCET

5) CO’s, PO’s mapping

P.USHA, Assistant. Professor,dept of ECE,KGRCET

5.COURSE INFORMATION SHEET

PROGRAMME: B. Tech. (Electrical and Electronics DEGREE: BTECH

Unit Details Hours

5.c). GAPS IN THE SYLLABUS - TO MEET INDUSTRY/PROFESSION REQUIREMENTS:

S.NO. DESCRIPTION No. Of Classes

1 Discrete time fourier transform 1

2 Applications of digital filters 1

5.d). TOPICS BEYOND SYLLABUS/ADVANCED TOPICS:

S.NO. DESCRIPTION No. Of Classes

1 Architecture of TMS320C5X processor 1

5. e). WEB SOURCE REFERENCES:

5. f). DELIVERY/INSTRUCTIONAL METHODOLOGIES:

 CHALK & TALK  STUD. ASSIGNMENT  WEB RESOURCES

 LCD/SMART BOARDS  STUD. SEMINARS ☐ ADD-ON COURSES

5.g). ASSESSMENT METHODOLOGIES-DIRECT

P.USHA, Assistant. Professor,dept of ECE,KGRCET

☐ STUD. LAB ☐ STUD. VIVA ☐ MINI/MAJOR ☐

5.h). ASSESSMENT METHODOLOGIES-INDIRECT

 ASSESSMENT OF COURSE OUTCOMES  STUDENT FEEDBACK ON

5.i). TEXT/REFERENCE BOOKS:

T/R BOOK TITLE/AUTHORS/PUBLICATION

ReferenceBo Digital Signal Processing — Fundamentals and Applications — Li

Reference Digital Signal Processing — S.Salivahanan, A.Vallavaraj and C.Gnanapriya,

Reference Discrete Systems and Digital Signal Processing with MATLAB —

P.USHA, Assistant. Professor,dept of ECE,KGRCET

Reference Digital Signal Processing – Nagoor Khani, TMG, 2012

6) Micro Lesson Plan

Topic wise Coverage [Micro Lesson Plan]

2 Discrete time signals and sequences

3 Linear shift invariant systems

4 Stability and Causality

5 Linear constant coefficient difference

8 Solution of difference equation of digital

10 Frequency response of stable systems

11 Realization of digital filters-direct and

13 Revision of 1st unit

14 Discrete time fourier transform

17 DFT and Properties of DFT

18 Linear convolution using DFT

19 Overlap save method

20 Overlap add method

21 Relation between DTFT and DFS

22 Relation between DFT and Z transform

24 Radix-2 decimation in time FFT algorithm

25 Radix-2 decimation in frequency

27 FFT with general radix N

28 Problems on overlap add method and save

30 Problems on DIF algorithm

32 UNIT III- IIR Digital Filters