DSP Couse File

DIGITAL SIGNAL PROCESSING
Contents :
1. Objectives
2. Outcomes
3. Prerequisites
4. Syllabus
5. Session plan
6. For each unit
a) Topic wise lecture notes
b) Assignment questions
c) Previous year questions
d) Objective Type questions
e) References
7. Bibiliography
Objective:
This course is an essential course that provides design technique for processing all types of signals in various fields.
The main objective of the course are:
To provide background and fundamental material for the analysis and development of signals.
To study the relationship between continuous and discrete time systems.
To discuss time, frequency and z plane analysis and the interrelationship between these methods.
To study design and structure of IIR and FIR filters for different specifications.
To develop various FFT algorithms and multirate signal processing techniques for finite word length.
Outcomes:
Able to obtain different Continuous and Discrete time signals.

Able to calculate Z-transforms for discrete time signals and system functions.
Ability to calculate discrete time domain and frequency domain of signals using discrete Fourier series and
Fourier transform.
Ability to develop Fast Fourier Transform (FFT) algorithms for faster realization of signals and
Able to design Digital IIR filters from Analog filters using various techniques (Butterworth and
Chebyshev).
Able to design Digital FIR filters using window techniques,Fouriour methods and frequency
sampling technique..
Ability to design different kinds of interpolator and decimator.
Ability to demonstrate the impacts of finite word length effects in filter design.
Prerequisites:
Laplace Transforms
Fourier Transforms
Signals and system
Syllabus:
UNIT-I:
Introduction and Realization of Digital Filters:
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.
Digital Filters:
Applications of z-transforms, solution of difference equations of digital filters. System function, stability criterion,
frequency response of stable systems. Realization of digital filters direct, canonic, cascade and parallel forms,
Lattice structures.
UNIT -II:
Discrete Fourier series and Fast Fourier Transform
Properties of discrete Fourier series, DFS representation of periodic sequences, Properties of DFS, Discrete Fourier
transforms: Properties of DFT, linear convolution of sequences using DFT. Computation of DFT ; over-lap add
method, Over-lap save method, Relation between DTFT, DFS, DFT and Z-Transform.
Fast Fourier Transforms (FFT)

Fast Fourier transforms (FFT) Radix-2 decimation in time and decimation in frequency, FFT Algorithms, Inverse
FFT, FFT with General Radix-N
UNIT-III:
IIR Digital Filters
Analog filter approximations Butter worth and Chebshev, Design of IIR Digital filters from analog filters, Bilinear
transformation method, step and impulse invariance techniques, Spectral transformations.
UNIT-IV:
FIR Digital Filters
Characteristics of FIR Digital Filters, frequency response. Design of FIR Digital Filters, Fourier method. Digital
filters using Window techniques, Frequency Sampling technique, Comparison of IIR & FIR filters.
UNIT-V:
Multirate Digital Signal Processing:
Introduction, Down Sampling, Decimation, up sampling, interpolation, sampling rate conversion, conversion of
band pass signals, concept of resampling, Application of multi rate signal processing.
Finite Word Length Effects
Limit cycles, Over flow oscillations, Round-off noise in IIR digital filters, Computational output round off noise,
Methods to prevent overflow, Tradeoff between round off and overflow noise, Measurement of coefficient
quantization effects through pole-zero movement, Dead band effects.
TEXT BOOKS:
T1: Proakis, J.Gard and D.G.ManolakisDigital Signal Processing: Principals, Algorithms and Applications,
3rdEdn., PHI, 2007.
T2: Discrete Time Signal Processing A.V. Oppenheim and RW schaffer, PHI, 2009
T3:LoneyLudermanFundamentals of Digital Signal Processing. John wiley, 2009
REFERENCE BOOKS:
R1: Li Tan, Fundamentals and Applications Digital Signal Processing Elsevier, 2008.
R2: T Robert J Schilling, Sandra L Harris, Fundamentals of Digital Signal Processing using Matlab, Thomson,
2007
R3: S. SalivahananDigital Signal Processing TMH, 2000
R4:Taan S. Elaali, Discrete systems and Digital Signal Processing with MATLAB CRC Press, 2009
R5: Emmanuel C Ifeachor and Barrie W.Jervis Digital Signal Processing- A Practical approach Pearson
Education 2nd edition, 2009
R6: "Digital Signal Processing" -RAMESH BABU Ed- 4thedition,scitech publications
Session Plan:
Topics in each unit as per Modules & sub modules for session Text Books/ Teaching Aids
JNTU syllabus Lecture plan for each topic used
Reference (BB/OHP/LCD)
No. Books
An overview of the course, application. T1: 1.1, 2.1;
Discussion of Syllabus and books. T2: 2.1
Overview, Introduction and BB
Requirement of DSP and L1 R2: 1.2.1; R3:
Application 1.2
UNIT -I: Introduction and Realization of Digital Filters
T1: 2.2 ; BB
Discrete time signals and Signal Definition , Classification and T2:2.2
L2
Sequences Representation R2:1.2.2 ; R3:
1.5
T1: 2.2; T2: BB

System Definition, Different types of
Linear Shift Invariant Systems L3 2.3 ;T1:2.3.5 ;
Systems
T2: 2.2
T1:2.2.6 ; T2: BB
L4
Causal and Non Causal Systems and 2.2; R1:
Stability and Causality
Stable and Unstable Equations 1.3 ;T2: 2.5; BB
T2:2.4.2 ;
Linear Constant Coefficient T1: 2.4.3 ; T2: BB

L5,L6 Solution of Difference Equations
Difference Equations 2.5
T1: 4.2.1 , BB
4.2.3 ;
T2: 2.6 ; R1:
L7 Impulse and Step Response
Response of Stable System 1.5
L8 System Function, Frequency Response BB
T1: 3.1,3.2 ; BB
Z Transform Definition and its T2: 4.1,4.2;
Applications of Z-Transform L9 R6: 2.1,2.2 ;
Applications
R3: 4.2
Explanation for the use of Difference BB

L10 Equation to Realize Digital Filters
Solution of Difference Equations
T1: 3.5.2 ; T2:
of Digital Filters L11 4.5 ; R6:
Solution for DE using Z-Transform and BB
inverse Z Transform 2.15; R3: 4.4
T1: 3.6.1 ; BB
System Functions L12 Transfer Function derivation
R6: 2.9 ;
T1: 3.6.4 ; BB
Stability Criterion L13 Derivation for Stability Condition
R6: 2.11;
Frequency Response of Stable To compute and Plot Amplitude and BB

L14 R6: 2.9
Systems Phase
Realization of Digital Filter using Form I T1: 7.3.1 ; OHP

Direct and Canonical Forms L15
and Form II R6: 5.14.1;
Realization of Digital FILTER Using T1: 7.3.3 ; OHP

Cascade Form L16
Cascade Form
R6: 5.14.2
T1: 7.3.4 ; OHP

Realization of Digital FILTER using
Parallel Form L17
Parallel Form
R6: 5.14.6
UNIT -II: Discrete Fourier Series and Fast Fourier Transform
DFS Representation of Periodic DFS Representation of T2: 8.2; R1:3.2 BB

L18
Sequences Periodic Sequences ; R6: 3.3
T2: 8.3 ; R6: BB

Properties f Discrete Fourier
L19 Linearity Shifting 3.2 ;
Series
R2: 3.3
T1: 7.1 ; T2: BB

DFT Definition, Discrete
8.6 ;
Discrete Fourier Transforms L20 Fourier Transforms of some
R3: 6.3: R6:
Functions
3.4
T1: 7.2; BB
T2:8.7;
Properties of DFT L21 Linearity , Shifting Property
R6: 3.6 ; R3:
6.3.2
Linear Convolution of Sequences T2: 8.9; R6: BB

L22 Linear Convolution Problems
using DFT 3.7
Computation of DFT, BB
Problems in DFT ,overlap Add
Relationship between L23 R6: 3.10
method
DTFT,DFS,DFT& Z-transform
R6: 3.10 BB
Introduction to FFT Overlap
Fast Fourier Transforms (FFT) L24 T1: 6.1; T2:
Save method 9.0 ; R6:
4.1- 4.3 ;
R1:6.0
Concept of Butterfly Diagram OHP

T1: 6.1.3 ; T2:
Radix-2 Decimation in frequency
L25 9.3; R6:4.4 ;
and FFT Algorithms Solution using Butterfly OHP
R1:6.2
Diagram to compute FFT
Solution using Butterfly T1: 6.1.3 ; T2: BB

Inverse FFT Diagram to Compute FFT, 9.4 ; R1:
L26
Decimation in Frequency 4.6 ; R1:6.3
T1: 6.1.5,6.1.6 BB
Overview of the Different ; T2: 9.5;
L27
Algorithms on FFT R6: 4.5,4.7 ;
R1: 6.2,6.3
T2: 9.3,9.4 ; BB
FFT with General Radix
T1: 6.1.3 ; BB
Formula for inverse FFT ,
L28 Problems and FFT with Radix R6: 4.9; R3:
N 3.6
T2: 9.6 ; R1:

6.4 ;
UNIT -III: IIR Digital Filters
T1: 3.1,3.2 ; BB
T2: 4.1,4.2;
Z Transform Definition and its
Applications of Z-Transform L29 R6: 2.1,2.2 ;
Applications
R3: 4.2
Explanation for the use of BB

Difference Equation to Realize
Digital Filters
Solution of Difference Equations T1: 3.5.2 ; T2:
L30
of Digital Filters
Solution for DE using Z- 4.5 ; R6: BB
Transform and inverse Z 2.15; R3: 4.4
Transform
T1: 3.6.1 ; BB
R6: 2.9 ;
Transfer Function derivation,
System Functions, Stability
L31 Derivation for Stability
Criterion
Condition
T1: 3.6.4 ;
R6: 2.11;
Frequency Response of Stable To compute and Plot BB
L32 R6: 2.9
Systems Amplitude and Phase
T1: 7.3.1 ; BB
Realization of Digital Filter
Direct and Canonical Forms L33
using Form I and Form II R6: 5.14.1;
T1: 7.3.3 ; BB
Realization of Digital FILTER
Using Cascade Form. R6: 5.14.2
Cascade Form and parallel form L34
Realization of Digital FILTER T1: 7.3.4 ;
using Parallel Form
R6: 5.14.6
R3: 8.1; BB
Introduction to Filter
Analog Filter Approximation L35
Classification FIR,IIR
R6:5.1
R3:8.5 ; BB
Design of IIR Filter using
Butter worth Filter L36
Butterworth Filter R6: 5.5
R3:8.6 ; BB
Design of IIR Filter using
Chebishev Filter L37
Chebishev Filter
R6: 5.7
Need to convert in Digital BB

Design of IIR Filter from Analog
L38 Domain , Overview of R6:5.12
Filters
different techniques
R6: 5.12.3 ; BB
Bilinear Transformation Method L39 Mathematical Representation
T1: 8.3.3
Step & Impulse Invariance Backward Derivative, Forward R6: 5.12.4 ; BB

L40
Techniques Derivative T1: 8.3.1
R6: 5.12.2 ; BB
LTI L41 Impulse Invariance Techniques
T1: 8.3.2
Low Pass to High Pass , Low R6: 5.13 ; BB

Spectral Transformations L42 Pass to Band Pass and Inter
Conversion T1: 8.4
UNIT -IV: FIR Digital Filters
Filter Definition , OHP

Characteristics of FIR Digital T1: 8.2.1 ;
L43 Classification, Characteristics
Filters R6: 6.1-6.2 ;
of FIR Digital Filters
Frequency Response L44 Frequency Response BB

R3: 7.3; R6:
6.3;
Rectangular, Triangular BB
L45
Design of FIR Digital Filters T1: 8.2.2 ; T2:
Hamming and Hanning BB
using Window Techniques L46 7.4 ; R6: 6.6
Window and overview of
Keiser Window
Design of FIR Filter using BB

T1: 8.2.3 ; R6:
Frequency Sampling Technique L47 Frequency Sampling
6.9 ; R3: 7.4
Technique
Comparison of IIT and FIR Comparison of IIR and FIR OHP

L48 R6: 6.9
Filters Filters
UNIT -V: Multirate Digital Signal Processing
T1: 10.1 ; BB
Introduction and application of
Introduction L49 R6:8.1; R3:
Multirate DSP
11.1
Concept and Formulation of T1:10.2 ; R6: BB

Down sampling: Decimation L50
Decimation 8.2
Concept and Formulation of T1: 10.3 ; R6: BB

Up sampling: Interpolation L51
Interpolation 8.4
T1: 10.4 ; R6: BB

Nyquist Rate, Up Sampling
Sampling Rate Conversion L52 8.8 ; R3:
and Down Sampling
11.3 ;
Sampling rate conversion of OHP

T1: 10.7, R6:
conversion of band pass signals L53 band pass signals by a rational
8.15
factor I/D
Design of phase splitter, BB

concept of resampling,
Interfacing of digital systems,
Application of multi rate signal L54 T1:10.9
sub band coding of speech
processing
signals
Introduction to Limit Cycles BB

Introduction to Limit Cycles, T1:7.11 ; R3:
Overflow Oscillations Round off L55 Overflow Oscillations 10.7 ; R6:
Noise in IIR Digital filters 7.6
Round off Noise
Computational output round off BB

L56 Rounding and truncation errors R3:10.3
noise
R3: 10.9 ; R6: OHP

Method to Prevent Overflow L57 Signal Scaling
7.12
IIR Digital Filters
Trade between Round off and Trade between Round off and BB
L58 R3: 10.5
Overflow Noise Overflow Noise
Measurement of Coefficient Measurement of Coefficient BB

Quantization effects through L59 Quantization R6: 7.9,7.15
Pole Zero Movement,
L60 Feature of Pole Zero R6: 7.10 BB
Dead band effects Movement, dead band effects
SUGGESTED BOOKS:
TEXT BOOKS:
T1: Proakis, J.Gard and D.G.ManolakisDigital Signal Processing: Principals, Algorithms and Applications,
3rdEdn., PHI, 2007.
T2: Discrete Time Signal Processing A.V. Oppenheim and RW schaffer, PHI, 2009
T3:LoneyLudermanFundamentals of Digital Signal Processing. John wiley, 2009
REFERENCE BOOKS:
R1: Li Tan, Fundamentals and Applications Digital Signal Processing Elsevier, 2008.
R2: T Robert J Schilling, Sandra L Harris, Fundamentals of Digital Signal Processing using Matlab, Thomson,
2007
R3: S. SalivahananDigital Signal Processing TMH, 2000
R4:Taan S. Elaali, Discrete systems and Digital Signal Processing with MATLAB CRC Press, 2009
R5: Emmanuel C Ifeachor and Barrie W.Jervis Digital Signal Processing- A Practical approach Pearson
Education 2nd edition, 2009
R6: "Digital Signal Processing" -RAMESH BABU Ed- 4thedition,scitech publications

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UNIT 1
Assignment questions
1) (a)What is a discrete time signal? Mention and define the mostly used discrete time signals. Classify
them?
(b)What is a discrete time system and classify them.
Explain about the linear time invariant systems.
(d)Explain the terms Stability and causality of LTI systems
2) Show that :
a) (n)=u(n)-u(n-1) b) u(n)= (k)(and k=-to n)=(n-k)(and k=0 to )
3) Consider the system y(n)=T[x(n)]=x(n) determine whether the system is invariant or time variant?
4) Compute the convolution y(n)=x(n)*h(n) of the following signals
a) x(n)={1,2,4},h(n)={1,1,1,1,1} b) x(n)={1,2,-1},h(n)={1}
c) x(n)={1,2,-1},h(n)=u(n) d) x(n)={1,1,0,1,1},h(n)={1,-2,-3,4}
e) x(n)=(1/2)nu(n),h(n)=(1/4)nu(n)
5) Determine the response of the system with impulse response h(n)=anu(n) to the input signal x(n)=u(n)-
u(n-10)
6) Determine the response of the relaxed system characterized by impulse response

n n
h(n)=(1/2) u(n)for input signals a)x(n)=2 u(n)
b)x(n)=u(-n)
7) Determine the direct form-II realization for each of the following LTI system
a) 2y(n)+y(n-1)-4y(n-3)=x(n)+3x(n+3)
b)y(n)=x(n)-x(n-1+2x(n-2)-3x(n-4)
8) find the convolution of the following sequences
x(n)=u(n) and h(n)=anu(-n-1),a>1
9) Compute and sketch the convolution y(n) of the sequences
1. x(n ) = {1,2,4} h(n ) = {1,1,1,1,1}
2. x(n ) = {1,2,3,4} h(n ) = {4,3,2,1}
10) a) Determine the impulse response of system described by second order difference equation
Y (n) 4y(n-1) = x(n ) - x(n - 1 )
b) The zero state response of a causal LTI system to input x(n) = {1,3,3,1} is y(n) = {1,4,6,4,1}.determine its
impulse response
11) a) Determine if the system is time invariant or not
i. y(n) = T[x(n)] = x(n) x(n-1)
ii. y(n) = T[x(n)] = nx(n)
b) Determine if the systems are linear, non linear, causal, and non causal
1. y(n) = nx(n) 2. y(n)= x(n) 3. y(n ) = x(2n)
12) Discuss and draw various IIR Realization Structures like Direct Form-I, Parallel and cascade forms for
thedifference equation given by y(n)=3/8y(n-1)+3/32y(n-2)+1/64y(n-3)+x(n)+3x(n-1)+2x(n-2)
13) Obtain the parallel and cascade realization structures for the system function given by
H(Z)=(1+1/4z- 1)/(1+1/2z-1+1/4z-2) .
14) Compare Direct form-I and Direct form-II structures w.r.t to hardware requirements.
15) Determine the frequency response of the system represented by the difference equation y(n)+3y(n- 1)+2y(n-
2)=2x(n)-
x(n-1) and comment upon the stability of the
16 (a) Test the following sytems for linearity, Time Invariance, Causality and Stability
y(n)=sin(2nf/F)x(n)
(b) A digital system is characterized by the following difference equation:
y(n)=x(n)+ay(n-1) assuming that the system is relaxed initially , determine its impulse response?
17) An LTI System is described by the equation y(n)=x(n)+0.8x(n-1)+0.7x(n2)

-0.45y(n-2). Determine the transfer function of the system. Sketch its poles and zeros on the Z-Plane?
18a ) Determine the impulse and unit step response of the systems described by the
followingdifference equation y(n)=0.6y(n-1)-0.08y(n-2)+x(n).
b) Find frequency response.
19Find inverse Z-transform of X ( Z ) log(1 0.5 Z 1 ); Z 0.5 using differentiation property.
(a) Find the inverse Z-transform of

1 3Z 1
X (Z ) ,Z 2
1 3Z 1 2Z 2
Z ( Z 2 4 Z 5)
(b) Find the inverse Z-transform of Z ( Z ) for ROC
( Z 3)( Z 1)( Z 2)
2 Z 3 Z 3 Z 1
(i) (ii) (iii)
20 a. Find the Z-transform and ROC of the ant causal sequence
x ( n ) { 3, 2, 1, 0,1}
b. Find the Z-transform of the sequence
x ( n ) {2, 1, 3, 2,1, 0, 2, 3, 1}
c. Find the stability of the system whose impulse response h(n) 2n.u(n)
Previous question papers(2011-2016)

1. Discuss and draw various IIR Realization Structures like Direct Form-I, Parallel and cascade forms for the
difference equation given by y(n)=3/8y(n-1)+3/32y(n-2)+1/64y(n-3)+x(n)+3x(n-1)+2x(n-2) (April-2011)
2. (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? (April-2011)
(b ) Prove that the system defined by the following equation is an LTI System
y(n)=x(n+1) -3x(n)+x(n-1); n0 (April-2011)
3. Write short notes on classification of systems (April-2011)
4. Discuss various discrete time sequences (April-2011)
5. Give the differences between anolog and digital systems ? (April-2012)
6. Determine the Frequency Response , Magnitude Response and Phase Response of the second order system
y(n)+1/y(n-1)=x(N)-x(n-1). (April-2012)
7. (a) What is Signal Processingand list the advantages, limitations of digital signal processing. List
out some applications of it?
(b) Discuss in brief about the classification of signals? (April-2012)
8. What is an LTI System? Show that an LTI System combined with time scaling property may result in an
Time- Variant System (April-2012)
9. An LTI System is described by the equation y(n)=x(n)+0.8x(n-1)+0.7x(n2)
-0.45y(n-2). Determine the transfer function of the system. Sketch its poles and zeros on the Z-Plane?
(April-2012)
10. Obtain all the possible realization structures of the following transfer function
11. (a )Determine the impulse and unit step response of the systems described by the
followingdifference equation y(n)=0.6y(n-1)-0.08y(n-2)+x(n).
b) Find frequency response. (May -2013)
12. Obtain the parallel and cascade realization structures for the system function given by
H(Z)=(1+1/4z- 1)/(1+1/2z-1+1/4z-2) . ( May -2013)
13. Compare Direct form-I and Direct form-II structures w.r.t to hardware requirements.
( May -2013)
14. Determine the frequency response of the system represented by the difference equation y(n)+3y(n-
1)+2y(n-2)=2x(n)-x(n-1) and comment upon the stability of the system.
(May -2013)
15. a)Write whether an LTI System with an impulse response represented below is causal or
not?justify h(n)=u(n+2)-u(n-2)
b )Write whether an LTI System with an impulse responserepresented below is stable or
not?justify?h(n)=3nu(n-1)
c) Determine the range of values of a for which an LTI System represented below is Stable,
h(n)=an for n0 (JUNE-2014)
16. A Causal System is represented by the following difference equation
Y(n)+1/4y(n-1)=x(n)+1/2x(n-1). Find the system transfer function H(Z), unit sample response,
magnitude and phase function of the ystem (May -2015)
17. .(a) Determine the Direct Forms I and II for the second order filter given by:
( )=2 ( 1) ( 2) + ( ) ( 1).
(b)Obtain the Cascade realization of the system function:
( ) = 1+ + 1+ + (May -2015)
18. (a) Test the following sytems for linearity, Time Invariance, Causality and Stability
y(n)=sin(2nf/F)x(n)
(b) A digital system is characterized by the following difference equation:
y(n)=x(n)+ay(n-1)
assuming that the system is relaxed initially , determine its impulse response? (May-2016)
19. Discuss the concept of stability and causality with examples? (Nov/Dec-2016)
20. Explain the canonical form of digital filter realization? (Nov/Dec-2016)
objective type questions
1. y(n)=x(2n) is a ____________ system [ ]
a) time invariant b) causal c) non causal d) none
2. y(n) = nx2(n) is a ____________system [ ]
a) Linear b) Non-linear c) time-invariant d) none
3. y(n)= x(n) +x(n-1) is a ____________ system [ ]
a) Dynamic b) Static c) time variant d) None
4. x(-n+2) is obtained by which of the following operatio n [ ]
a) x(-n) is shifted left by 2 samples

b) x(-n) is shifted right by 2 samples
c) x(n) is shifted left by 2 samples
d) none
5. The necessary and sufficient condition for causality of an LTI system is [ ]
a) h(n) =0 for n=0 b) h(n) =0 for n>0 c) h(n) =0 for n<0 d) none
6. Convolution is a measure of similarity beteen two signals.
7. If the input output relation of a system doesnot vary with time,the system is said to be [ ]
a.time variant b.Time Invariant c. Static d.Linear
8. Causality condition for an LTI System is [ ]
a.h(n)=0 for n<0 b. h(n)=0 for n>0
9. Sampling Theorem states that [ ]
a.fs2fm b. fsfm c. fs2fm d. fsfm4.
10. if x(n) is a causal sequence then the ROC is the entire Z plane except at [ ]
a.Z= b.Z=0 c. Both a&b d. None
11. Z[(n)]=
a. 0 b. 1 c. >1 d.
12 ________is the set of all values of Z FOR WHICH X(Z) attains a finite value. [ ]
a. Radius of convergence b. Region of convergence

c. Region of divergence d. None
13.Time shifting property of Z transform
a. z{x(n+k)}=Z-KX(Z) b. z{x(n-k)}=ZKX(Z)
-K
c. z{x(n-k)}=Z X(Z) d. None
14. Quantization is necessary to represent the signal in [ ]
(a) Binary code (b) Decimal code (c) Hexa decimal code (d) Octal code
15. A discrete time signal represented by a set of numbers is called [ ]
(a) Continuous signal (b) Step signal (c) Discontinued signal (d) Sequence
16. The Fourier Transform of a finite energy discrete time signal x(n) exist when [ ]
=0
(a) x ( n) < (b) x(n) < (c) x(n) > 0 (d) x(n)
n= n=0 n=0
n=

17. The z transform of K e- n is [ ]
Kz Kz Kz Kz 1
(a) (b) (c) (d)

z e z e z +e z e
18. If the system output at any time n depends on future inputs or outputs then the system is
called [ ]
(d) Non linear
(a) Non - causal (b) Causal (c) Stable
19.Discrete time system is stable if the poles are
(a) Within unit circle |z| < 1 b) Outside the unit circle c) On the Unitcircle d)none
20.if a signal depends on only one independent variable, it is called a one dimension signal.
22.the representation of a signal by mathematical expression is known as Signal Modeling.
23. A signal which can be described by a mathematical equation is called a Deterministic signal.
24.For an even signal, x(-n)=x(n) for all n.
25.A system is said to be stable if every Bounded input producesa bounded output.
26.An Linear System is one which satisfies the properties of homoginity and Superposition.
27. WN is known as twiddle factor.
28. In overlap-save method ,we use circularconvolution.

29. Fourier Transform of x(n)*h(n) is equal to the product of X(w)H(w)
30. DFT allows us to perform frequency analysis on a \digital computer.
31.The DFT is obtained by sampling one period of the fourier transform X(w)
32.Inverse Fourier transform of sgn (w) is Rectangular .
33.DFT supports only Circular convolution
34. (n)=u(n)-u(n-1)
35.Double integration of unit step function leads to Parabolic
36.The Laplace transform of impulses ignal is 1
37.. A signal which can not be described by a mathematical equation is called a randomsignal.
38.The roots of the denominator or the value of z for which X(z) becomes infinite, defines
Poles
39.In two sided sequence, the ROC is entire z plane except at z=0 and z= .
40.In direct form II realization, the number of memory locations required is less than that
of direct form I realization.
41. A Causal System is also known as a Non Anticipative.
42. A non Causal System is also known as a Anticipative.
43. An LTI System is one which satisfies the properties of Linearity, Time Invariance .
44.A System is said to be Stable if every Bounded input produces bounded output.
45.A Linear System Obeys the principle of Superposition.
46.A System which does not have an unique relation between input and output is called Non Invertible.
47.For an Energy Signal E=Finite and P=Zero.
48. For a Power Signal and P= Finite and E=Infinity.
49. For a discrete time system to be stable, its impulse response must be Absolutely Summable.
50.For an anti-causal signal,x(n)=0 for n>0.

UNIT 1I
Assignment questions
1. a)State and explain the properties of discrete fourier series

a. b)Explain how the DFS is represented for a periodic sequence
b. c) State the properties of DFT
2. Determine the eight point DFT of the sequences of x(n) = {1, 0n3 0, 4n
3. Determine the N-point DFT of the sequences x1(n)= x(n)cos 2kn/N and x2(n) = x(n)sin 2kn/N ,
0nN-1 Provided DFT of x(n) = X(k)
4. a. Determine the N-point DFT of signals
b. X(n)= {1,1,1,1,1,0,0}
5. Distinguish between circular and linear convolution
6. Find the DFT of the causal 3-sample average
a. For the x1(n),x2(n) and N given compute linear and circular convolutation of the given sequence
b. X1(n)= (n)+ (n-1)+ (n-2),N=3
c. X2(n)= 2(n)- (n-1)+ 2(n-2)
7. a. What is FFT
a. b.What is meant by Radix-2 FFT.
b. How many multiplication of addition are required to compute N-Pont DFT using Radix-2 FFt.
c. What are the differences of similarities between DIF and DIT algorithms.
8. Develop DIT algorithm for N=4 of draw signal flow graph
9. given input x(n)=(1,2,3,0) and system function h(n) = (1,2,0,0). Use FFT method to calculate output y(n), using DIT
algorithm for FFT
10. Given x(n) = (1,2,3,0) and y(y) = (1,1,-1,-1), use DIF algorithm to compute Y(K) and Y(K)
11. Compute 8-point DFT of the following sequences using

(a) DIF algorithm (b) DIF algorithm
(i) x(n) = {1,-1,1,-1,0,0,0,0} (ii) x(n) = {0.5,0,0.5,0,0.5,0,0.5,0}
(iii) x(n) = {2,1,2,1,2,1,2,1 } (iv) x(n) = {1,2,3,2,1,2,3,2}
12. (a)Compute the circular convolutuion of the sequences
x1(n)={1,2,0,1}and
x2(n)={2,2,1,1} using DFT Approach
13. Compute 4-point DFT of the following sequences using (a) DIT algorithm (b) DIF algorithm
(i) x(n) = {1,2,3,4} (ii) x(n) = {1,1,-1,-1}
(iii) x(n) = {1,2,-1,1} (iv) x(n) = {0,1,2,3}
14. Compute IDFT of the following sequences using (a) DIT algorithm (b) DIF algorithm
(i) X(K) = {1, 1+j, 1-j2, 1, 0, 1+j2, 1+j}
(ii) X(K) = {12, 0, 0, 0, 4, 0, 0, 0}
(iii) X(K) = {5, 0, 1, -j, 0, 1, 0, 1+j, 0}
15. Compute the DFT for N=8 using (a) DIT algorithm (b) DIF algorithm for the following.
(a) x (n) 1 0 n 3
(b) x (0) x (3) 1 and x (1) x (2) 1
(c) x ( n ) n for 0 n 7
= 0 otherwise
16. a. Calculate the number of multiplication needed in the calculation of DFT using FFT algorithm with 32-point
sequence.
b. Evaluate and compare the 8-point for the following sequences testing DIT-FFT algorithm
1 for 3 n 3
(a) x1 (n) otherwise
0
1 for 0 n 6
(b) x2 (n) otherwise
0
17. a. Compute 4-point DFT of a sequence x(n) = {0, 1, 2, 3} using DIT, DIF algorithm.
b. Find the DFT of a sequence x(n) = {1, 2, 3, 4, 4, 3, 2, 1} using. DIT algorithm.
18. a. Draw the flow graph of 16-point DIT-FFT.

b.Construct an 8-point DFT from two 4 point DFTS.
19. a) State & prove circular convolution property of DFT?
b) Perform Linear Convolution of the two sequences x(n)= {1,-1,2,-2,3,-3,4,-4} and h(n)={-1,1}
using overlap add method
20 a.Find the convolution of the sequences x1(n) and x2(n) using overlap add method
x1(n)={3,-1,0,1,2,3,0,1,1,2}
x2(n)={1,1,1
b. Obtain the relationship between DTFT and DFS.
1. Define DFS . State any four properties of DFS? (April-2011)

2. Define DFT&IDFT. State any four properties of DFT? (April-2011)
3. Find 8-Poine DFT of the given sequence x(n)={1,2,3,4} (April-2011)
4. Find the IDFT of the given sequence X(K)={2,2-3j,2+3j,-2} (April-2011)
5. Define convolution . Compare Linear and Circular Convolution Techniques (April-2011)
6. Find the linear convolution of the two sequences x(n)={1,2}and h(n)={1,2,3}using DFT &IDFT
(April-2011)
7. Compare the computational Complexity of DFT and FFT (April-2011)
8. Give the steps involved in implementing Radix-2 DIT-FFT algorithm (April-2011)
9. FindX(K) of the given sequence x(n)={1,2,3,4,4,3,2,1} using DIT-FFT algorithm
April-2011)
10. Find the IFFT of the given X(K)={1,2,3,4,4,3,2,1} using DIF Algorithm (April-2011)
11. 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 (April-2011)
12. Define DFT&IDFT. Prove circular convolution , circular correlation, and time reversal properties of DFT .
(April-2012)
13. Determine the relation between DFT and Fourier Transform of an Aperiodic sequence
(April-2012)
14. Compute Linear Convolution of two given sequences x(n)={1,2,3} and h(n)={2,3} (April-2012)
15. An 8-Point sequence is given by x(n)={1,2,1,2,1,2,1,2}. Compute 8-point DFT of x(n) using Radix-2 DIT-
FFT&DIF-FFT (April-2012)
16. Compute the IFFT for the sequence X(K)={0,1,2,3,0,0,0,0} using DIT algorithm (April-2012)
17. a) State & prove circular convolution property of DFT?
b) Perform Linear Convolution of the two sequences x(n)= {1,-1,2,-2,3,-3,4,-4} and h(n)={-1,1}
using overlap add method (May-2013)
18. (a)Find the convolution of the sequences x1(n) and x2(n) using overlap add method
x1(n)={3,-1,0,1,2,3,0,1,1,2}
x2(n)={1,1,1}
b) Obtain the relationship between DTFT and DFS. (JUNE-2014)
19. (a) State and prove the following Properties of DFT
i) Linearity ii) Frequency Shifting
(b) Explain the following
i) Overlap Add Method ii) overlap
Save Method
21. . (a) Develop the DIT-FFT algorithm for N=8 using the resulting signal flow graph. Compute
8- point DFT of the sequence ( ) = , 0 7.
(b) If x2(n)=x1(-n) without performing FFT find X2(K)using X1(K) (May-
2015)
22. By taking an example compute DFT by using Over-lap save Method (May-2016)
23. (a)Compute the circular convolutuion of the sequences
x1(n)={1,2,0,1}and
x2(n)={2,2,1,1} using DFT Approach
(b)What is FFT? Calculate the no.of multiplications needed in the calculation of DFT using FFT
algorithm with 32 point sequence. (May-2016)
24. Compute the FFT of the sequence x(n)={1,0,1,1,0,1,1,1} (May-2016)
25. Explain the properties of DFT? (Nov/Dec-2016)
26. Explain the Radix-2 Decimation in Time algorithms? (Nov/Dec-2016)
1. The methods used to find the circular convolution of two sequences are ( )
a. a. Concentric circle b. Matrix multiplication
b. c. Both a&b d. none
2. The number of stages for N=16 in DIT-FFT are ( )
a.8 b. 4 c. 2 d.16
3. For DIT-FFT the input sequence is ______ & the output sequence is in _____ order ( )
a. a.Natural, Bit Reversal b. Bit Reversal,Natural
b. c. Bit Reversal, Bit Reversal d. Natural,Natural
4. Applications of FFT Algorithm ( )
a. Linear Filtering b. Correlation c. Spectrum Analysis d. all of the above
5. In radix 2 FFT, the total no. of complex additions are reduced to [ ]
(a) log2N (b) N log2N (c) m log2N (d) N log2 m

6. In direct computation of DFT, the total no. of complex multiplications are given by [ ]
a)N b) N2 c)2N2 d)N/2
7. Circular convolution between two finite length sequence is equal to_________ of their [ ]
correspondingDFTs
a) Sum b) Linear Convolution c)Product D)Difference
8. WN also represent an Nth root of_____________.
9.DFT performs filtering operation in [ ]
a)time domaun b) frequency domain c) both the time and frequency domain d) none
10. Twiddle factor isWN=
a)ej2/N b) ej/N c) e-j2/N d) e-j/N
11.The DTFT of the z transform evaluated along the
a) imaginary axis of z-plane b)real axis of Z-Plane
c) Unit circle in Z-Plane d) entire Z-Plane
12.The no.of complex multiplications involved in the direct evoluation 8-point DFT [ ]
a) 8 b)64 c) 16 d) 56
13. Which of the following is true regarding the number of computations required to compute anN-pointDFT?
a)N2 complexmultiplications and N(N-1) complex additions
2
b)N complexadditions and N(N-1) complex multiplications
c)N2 complexmultiplications and N(N+1) complex additions
2
d)N complexadditions and N(N+1) complex multiplications
14. Which of the following is true regarding the number of computations required to compute DFT at any one value of
k?
a) 4N-2 real multiplications and 4N real additions
b) 4N real multiplications and 4N-4 real additions
c) 4N-2 real multiplications and 4N+2 real additions
d) 4N real multiplications and 4N-2 real additions
15.WNk+N/2=
a)WNk
b)-WNk
c)WN-k
d)None
16. The computation of XR(k) for a complex valued x(n) of N points requires:
2
a) 2N evaluations of trigonometric functions
b) 4N2 real multiplications
c) 4N(N-1) real additions
d) All of the mentioned
17. Divide-and-conquer approach is based on the decomposition of an N-point DFT into successively smaller DFTs. This
basic approach leads to FFT algorithms.
a)True
b)False
18. If the arrangement is of the form in which the first row consists of the first M elements of x(n), the second row consists
of the next M elements of x(n), and so on, then which of the following mapping represents the above arrangement?
a)n=l+mL
b)n=Ml+m
c)n=ML+l
d)None
19.IfN=LM,thenwhatisthevalueoWNmqL?
a)WMmq
b)WLmq
c)WNmq
d)None
20. How many complex multiplications are performed in computing the N-point DFT of a sequence using divide-and-
conquer method if N=LM?
a)N(L+M+2)
b)N(L+M-2)
c)N(L+M-1)
d)N(L+M+1)
21. How many complex additions are performed in computing the N-point DFT of a sequence usingdivide-and-
conquermethodifN=LM?
a)N(L+M+2)
b)N(L+M-2)
c)N(L+M-1)
d)N(L+M+1)
22. Which is the correct order of the following steps to be done in one of the algorithm of divide andconquermethod?
1)Store the signal column wise
2) Compute the M-point DFT of each row
3) Multiply the resulting array by the phase factors WNlq.
4) Compute the L-point DFT of each column.
5) Read the result array row wise.
a)1-2-4-3-5
b)1-3-2-4-5
c)1-2-3-4-5
d)1-4-3-2-5
23. If we store the signal row wise then the result must be read column wise.
a)True
b)False
24. If we store the signal row wise and compute the L point DFT at each column, the resulting array must be multiplied by
which of the following factors?
lq
a)WN
b)WNpq
c)WNlq
d)WNpm
25.Convolution of long sequences can be done using sectionedconvolutions
26.The two methods of section convolution are overlap add method and overlap add method.
27.The Direct computation of DFT requires N2 real multiplications and N(N-1)real additions.
28.The FFT may be defined as an algorithm or computing DFT.
29.The basic FFT algorithms are DIT FFT and DIF FFT.
30..For DIT FFT the input is in Bit Reversal order and the output is in Natural order.
31.For DIF FFT the input is in Natural order and the output is in Bit Reversal.
32.The computation 64 point DFT by radix-2 DIF FFT involves six stages of computation.
33.The number of complex additions involved in direct computation of 8-point DFT is 64.
34.In radix-2 DFT N/2 butterflies per stage are required to present the computational process.
35.The signal flow graph for computing DFT by radix-2 FFT is also called-BUTTERFLY diagram
36.The expansion of FFT is fast fourier transform
37.The main advantage of FFT is computation complexity reduced
38.The number of multiplications needed in the calculation of DFT using FFT with 32-point
sequence 8
39.Nlog N2 number of additions are required to compute N pt DFT using radix 2 FFT
40..Appending zeros to a sequence in order to increase the size or length of the sequence is
called Zero padding
41.In radix 2 FFT, the N point sequence is decimated into two N/2 point DFTs.
42.IN FFT ,the computational efficiency is achieved by adopting a divide and conquer approach.
43.FFT is a faster method of computation,because it exploits the symmetry and periodicity properties of
the phase factor WN.
44.In DFT computation using radix-2 FFT, the value of N should be such that N=2m
45. The computation 32 point DFT by radix-2 DIF FFT involves five stages of computation
46.The convolution by convolution sum formula is called slow convolution.
47.The convolution by FFT is called fast convolution.
48. The no.of complex multiplications involved in the computation of 256-point DFT by radix-2 FFT
Is 1024
49.The DFT X(k) of a 2 sample sequence x(n)={4,2}is{6,2}.
50.For radix-2 FFT , N must be a power of 2

UNIT III
Assignment Questions
1. Design a butter worth high pass filter satisfying the following specifications. p = 1dB, s = 15 dB, p = 0.4 , s
= 0.2
2. Design a butter worth low pass filter satisfying the following specifications
fp = 0.1 Hz, = 0.5 dB, fs = 0.15 Hz, s = 15dB, F= 1Hz
3. Design a band stop butter worth and chebyshev type I filter to meet the following specifications.
a) Stop band 100-600 Hz

b) 20dB attenuation at 200 & 400 HZ.
c) The gain at W=0 is unity.
d) The pass band ripple for the chebyshev filter is 1.1 dB.
e) The pass band ripple attenuation for butter worth filter is 3 dB.
4. Design a chebyshev type I band reject filter with the following specifications.
Pass band dc to 275Hz & 2KHZ to
Stop band 550 HZ to 1000HZ.
p = 1dB, s = 15dB, F = 8KHZ.
5. Design an analog butter worth filter that has a 2 dB Pass band attenuation at a frequency of 20 rad/sec and at least
10 dB stop band attenuation at 30 rad/sec.
6. Given the specification p = 1dB, s = 30dB, p = 200rad/sec, s = 600 rad/sec. Determine the order of the filter.
7. What are the differences between along filter and digital filter.
8. How one can design digital filters from analog filters.
Design a butter worth filter using the bilinear transformation method for the following specifications.
0.8 H (ejw) 1 0 w 0.2
H(ejw) 0.2 0.6 w
9. Bilinear transformation
0.8H (ejw) 1
H (ejw) 0.2
10. What is warping effect? What is its effect on magnitude of phase response?
11. Write a short note on pre wrapping?
12. What is bilinear transportation?
13. What are the properties of the bilinear transformation?
14. What are the advantages and disadvantages of the bilinear transformation?
15. Impulse invariance method

0 w 0.2
0.6w
16. Design a butter worth filter using impulse invariance method for the following specifications.
0.8 H (ejw) 1 0 w 0.2
H(ejw) 0.2 0.6 w
18. What are the conditions to convert digital low pass to digital high pass filter?
19. What are the conditions to convert digital low pass to band pass filter?
20. What is the matched Z- transformation? What are the disadvantages?

1. Discuss IIR Filter design using impulse invariant transformation and list out its advantages and limitations
(April-2011)
2. Discuss IIR filter design using bilinear transformation and hence discuss frequency warping effect
(April-2011)
3. Discuss Analog&Digital transformation Techniques (April-2011)
4. Design a digital butterworth LPF Using Bilinear Transformation technique for the following specifications
0.707H(w) 1 0w0.2
H(w) 0.08 0.4w (April-2011)
5. Compute the poles of an analog chebyshev filter TF that satisfiesthe constraints
0.707H(j) 1 0 2
H(j) 0.14
And determine Ha(s) and hence obtain H(Z) using bilinear transformation (April-2011)
6. How chebyshev filter approximation is superior than butterworth filter approximation
(April-2012)
7. Discuss in detail the procedure of designing an anolog filter using Butterworth approximation Technique
(April-2012)
8. In a speech recording system with a sampling frequency of 10000 hz. The speech is corrupted by random noise. To
Remove noise while preserving speech information,the following specifications are given
Speech Frequency range: 0-3000 khz
Stopband range: 4000-5000 khz
Passband ripple: 3 db
Stopband attenuation : 25db
Determine the filter Order and Transfer function using Butterworth IIR Filter. (April-2012)
9. Discuss the problems encountered in design of digital filter using impulse invariant and bilinear transformation
techniques (April-2012)
10. a) What is Bilinear Transformation and Sketch the mapping of S-Plane into Z-Plane in Bilinear
Transformation.
b) Explain how to convert an analog filter transfer function into digital filter transfer function
using Bilinear Transformation (May-2013)
11. For the analog filter transfer function H(S)=2/(S+1)(S+3). Determine H(Z) USING Bilinear Transformation .Use T=0.1
sec (June-2014)
12. a) Design a digital IIR lowpass Butter worth Filterthat has a 2 db passband attenuation at a frequency of 300 rad/sec
and atleast 60db stopband attenuation at 4500 rad/sec.Use backward reference Transformation
(b) Determine the Order and Poles of a type-I lowpassChebyShev Filter that satisfies the following
Constraints 0.8|H(w)|1 0W0.2
| H(w)|0.2 0.6W
(May-2015)
13. (a)Discuss in detail about spectral transformations
(b) Explain how IIR Digital filters are designed from analog filters. (May-2016)
14. (a)Compare the impulse invariance and bilinear transformation method
(b) Find the order and poles of a lowpassbutterworth filter that has a -3 db bandwidth of 400hz
and an attenuation of 20 db at 1 khz (May-2016)
15. For the analog filter transfer function H(S)==2/(S+2)(S+3).Determine H(z) using Impulse invariance method.
AssumeT=1 Sec. Nov/Dec-2016)
16. Design a digital second order Low pass butterworth Filter with cut-off frequency 2.2 KHZ using Bilinear
Transformation.Sampling rate 8 KHZ. (Nov/Dec-2016)
1. In the Frequency Transformations of the analog domain the transformation is

a. Low Pass to Lowpass
b. Lowpass to Highpass
c. Lowpass to Bandpass
d. Lowpass to Bandreject
2. In the Frequency Transformations of the analog domain the transformation is

a. Low Pass to Lowpass
b. Lowpass to Highpass
c. Lowpass to Bandpass
d. Lowpass to Bandreject
3. The magnitude response of the following filter decreases monotonically as frequency increases
a. Butterworth Filter
b. Chebyshev type - 1
c Chebyshev type - 2
d. FIR Filter
4. The transition band is more in

a. Butterworth Filter
b. Chebyshev type - 1
c. Chebyshev type - 2
d. FIR Filter
5. The poles of Butterworth filter lies on

a. sphere
b. circle
c. ellipse
d. parabola
6. I I R digital filters are of the following nature

a. Recursive
b. Non Recursive
c. Reversive
d. Non Reversive
7. In I I R 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
8. Which of the following is best suited for I I R filter when compared with the FIR filter
a. Lower sidelobes in stopband
b. Higher Sidelobes in stopband
c. Lower sidelobes in Passband
d. No sidelobes in stopband
9. In the case of I I R filter which of the following is true if the phase distortion is tolerable
a. More parameters for design
b. More memory requirement
c. Lower computational Complexity
d. Higher computational complexity
10.A causal and stable I I R filter has

a. Linear phase
b. No Linear phase
c. Linear amplitude
d. No Amplitude
11.Neither the Impulse response nor the phase response of the analog filter is Preserved in the digital
filter in the following method
a. The method of mapping of differentials
b. Impulse invariant method
c. Bilinear transformation
d. Matched Z - transformation technique
12. Out of the given I I R filters the following filter is the efficient one
a. Circular filter
b. Elliptical filter
c. Rectangular filter
d. Chebyshev filter
13.What is the disadvantage of impulse invariant method
a. Aliasing
b. one to one mapping
c. anti aliasing
d. warping
14.Which of the I I R Filter design method is antialiasing method?
a. The method of mapping of differentials
b. Impulse invariant method
c. Bilinear transformation
d. Matched Z - transformation technique
15.The nonlinear relation between the analog and digital frequencies is called
a. aliasing
b. warping
c. prewarping
d. antialiasing
16.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
17.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
18.In the IIR filter Design method by approximation of derivatives as varies
from to , the corresponding locus of a point in the zplane is a circle with radius and center
a. 0,0
b. 1,1
c. 1, - 1
d.none
19.The I I R filter design method thatovercomes the limitation of applicability to only
Lowpass filter and a limited class of bandpass filters is
a. Approximation of derivatives
b. Impulse Invariance
c. Bilinear Transformation
d. Frequency sampling
20._________ is more in Butterworth filter when compared to chebyshev filter ( )
a.Pass band b. Stop band c. Transition Band d. Both a&b
21.For Recursive Realization the current output y(n) is a function of ( )
a. Past outputs, present input &past input
b. Past inputs, Present &past outputs
c. Present & past outputs
d. None
22. Filters designed by considering all the infinite samples of the impulse response are called IIR filters.
23. The physically realizable IIR filters donot have linear phase.
24. The IIR filter specification includes the desired characteristics for the magnitude response only.
25. Filters designed by considering all the finite samples of the impulse response are called FIR filters.
26. The impulse response is obtained by taking the inverse fourier transform of ideal frequency response.
27. The bandwidth of the discrete signal is limited by sampling frequency.
28. The popular methods for design of IIR digital filters uses the techniques of transforming an analog filter in to an
equivalent digital filter.
29. The bandwidth of the real discrete signal is half the sampling frequency.
30.The three techniques used to transform an analog filter to digital filter are approximation of derivatives, impulse
invariant transformation and bilinear transformation.
31. The two properties which are to be preserved in analog to digital transformation are causality and stability.
32. The tolerance in the passband and stopband are called ripples.
33. In impulse invariant transformation the impulse response of digital filter is the sampled version of the impulse response
of analog filter.
34. In impulse invariant transformation , the left half poles of s-plane are mapped into the exterior of unit circle in z-plane.
35. In impulse invariant transformation , the right half poles of s-plane are mapped into the exterior of unit circle in z-
plane.
35. In impulse invariant transformation , any strip of width 2/T in s-plane are mapped into the entire z-plane.
35.The phenomenon of high frequency components acquiring the identity of low frequency components is called aliasing.
35. Aliasing is higher frequencies impersonating low frequencies.
36. Aliasing occurs only in impuse invarient transformation.
37. The impulse invariant mapping is many to one mapping, whereas bilinear mapping is a one to many mapping
38. The distortion in frequency axis due to nonlinear relationship between analog and digital frequencies is called
frequency warping.
39. In bilinear transformation , the effect of warping on magnitude response can be eliminated by pre warping the analog
filter.
40. A linear phase analog filter cannot be transformed into a linear phase digital filter using bilinear transfer function.
41. the two popular techniques used to approximate the ideal frequency response are butterwoth and chebyshev
42. in butterworth approximation ,the magnitude response is maximally flat at the origin and monotonically dicreases with
increase in frequency.
43.at the cutoff frequenc the magnitude of the butterworth filter is 1/2 times the maximum value.
44. in type-1 chebyshev approximation ,the magnitude response is equiripple in the passband and monotonic in the
stopband.
45. in type-2 chebyshev approximation ,the magnitude response is monotonic in the passband and equiripple in the
stopband.
46. the type -2 chebyshev response is also called inverse chebyshev response.
47. in chebyshev approximation the normalized magnitude response has a value of 1/1+2at the cutoff frequency.
48. IIR Filters are recursive type.
49. Butterworth filters have wide band transition region.

50. Type-1 chebyshev filter contains oscillations in the passband.
UNIT-IV:
1. What are the advantages and disadvantages of FIR filters over IIR filters?
2. Determine the coefficients of a linear phase FIR filter of length M=15 has a symmetric unit sample response and a
frequency response that satisfies the conditions
H (2k) = 1, k=0,1,2.3
------
15 =0 ,k=4, 5,6,7
3. Determine the frequency response of FIR filter defined by y(n) = 0.25 x (x) + x(n-1)+0.25 x (n-2). Calculate the
phase delay and group delay.
4. Design an ideal differentiator with frequency response H (e jw) = jw - w. Using Hamming window.
5. Design a filter with Hd(e jw) = e j3w-/4 w /4 = 0/4 <W

Using a hanning window with N=7
6. Design an ideal differentiator with frequency response H (e jw) = jw - w using rectangular window.
7. Design an ideal Hilbert transform using frequency response H (e jw)=j for - w0 using rectangular window = -j
for 0w using rectangular window for N =11
8. Design an ideal Hilbert transform using frequency response Blackman window.
9.Design an FIR low pass filter satisfying the following specifications

p 0.1dB, s 44 dB.
wp = 20 rad/sec, ws = 30 rad/sec, wsf = 100 rad/sec.
10. What are the advantages of raiser window?
11. Give the equation specifying Kaiser window?
12. Suppose the axis of symmetry of impulse response h(x) lies half way between 2 samples, for what kind of
applications this type of impulse response is used.
13. For what kind of applications, the ant symmetrical impulse response can be used?
14. For what kind of applications the symmetrical impulse response?
15. Using a rectangular window technique design a LPF with pass band gain of unity, cutoff frequency of 1000Hz and
working at a sampling frequency of 5 KHZ. The length of the impulse response should
be 7.
16. Determine the filter coefficients h (n) obtained by sampling
Hd (e jw) = e j(N-1)w/z 0 w/2
For N =7 = 0 /2w
17. Design a FIR band pass digital filter satisfying the following specifications.
fp1= 20 Hz. fpz = 30 Hz, fs2 = 40 Hz, F=100Hz
p = 0.5dB, s = 30 dB.
18. Write a short notes about location of the zeros of linear phase FIR filters.
19. Design an ideal band reject filter with a desired frequency response
Hd = (e jw ) = 1 for w /3 &w 2/3=0 other wise
Find the value of h(n) for N=11. Find H(z)
20. Design an ideal high pass filter with a frequency response Hd (e jw) = 1 /4w = 0 W</4
Find the values of h (n) for N=11 find H (Z).

1. Compare various windowing functions (April-2011)
2. Design an FIR Digital lowpass filter using Rectangular window whose cutoff freq is 2 rad/s and length of window
N=9 (April-2011)
3. Design an FIR Digital lowpass filter using Hanningwindow whose cutoff freq is 2 rad/s and length of window N=9
(April-2011)
4. Compare FIR and IIR Filters? (April-2011)
5. Define phase delay and group delay (April-2012)
6. FIR Filters always stable and have linear phase characteristics.Justify (April-2012)
7. What are the desirable characteristics of windowing function to be satisfied in filter design?
(April-2012)
8. The following transfer function characterizes and FIR filter (M=11) . Determine the magnitude response and show
that the phase and group delays are constant
H(Z)=h(n)z-n (April-2012)
9. Design an FIR Digital low pass Filter using blackman-turkey window whose cutoff freq is 1.2 rad/s and length of
window N=5 (April-2012)
10. Design an FIR digital stop band filter using rectangular window whose upper and lower cutoff frequencies are 4&5
rad/s and length of window N=9. Realize the filter using linear phase realization structure
(April-2012)
11. Design FIR Digital High Pass Filter using Hamming Window whose cutoff freq is 1.2 rad/s and
length of window N=5 . Compare the same using Rectangular Window .Draw the frequency
response curve for the both cases. (May- 2013)
12. Design a High Pass FIR Filter whose cutoff freq is 1,.4 rad/s and N=5 using hamming window
(June-2014).
13. (a)Detrmine the transfer function H(Z) of an FIR Filter to implement h(n)=(n)+2 (n-1)+ (n-2)
using freq sampling techniques
(b) Give the comparision between FIR and IIR Filters with Examples (May-2015)
14. (a)Draw and explain the freq response of FIR Digital Filter
(b) Design a high pass filter using hamming window with a cutoff freq of 1.2 rad/s and N=9
(May- 2016)
15. (a) List the design steps of FIR filters using fourier method
(b) Design a low pass digital filter using Kaiser window satisfying the specifications given below.
Pass band cutoff freq=100hz
Stopband cutoff freq=200hz
Passband ripple=0.1 db
Stopband attenuation= 20 db
Sampling freq=1000hz (May- 2016)
16.Using a Rectangular Window technique, Design a LOwpass filter with pass band gain of unity, cut off frequency of
1000 HZ and working at a sampling frequency of 5 KHZ . The length of impulse response should be 7.
(Nov/Dec-2016)
1. An LTI System modifies the input spectrum X(w) according to its frequency response to yield
an output spectrum Y(w).
2. H(w) acts as weighing or spectral shaping function to the different frequency components in the input signal.
3. For a linear phase filter the delay is constant .
4. The phase distortion is due to_non linear phase characteristics of the filter.
5. In FIR filter phase is a linear function of w.
6. Delay distortion is synonymous with phase distortion.
7. The Hd (w) is periodic with periodicity of 0 to 2.
8. The filters are classified according to their impulse , frequency response.
9. Based on impulse response filters are classified as FIR and IIR filters.
10. In FIR filters with constant phase delay the impulse response is symmetric.
11. In FIR filters with constant group and phase delay the impulse response is anti-symmetric
12. Based on frequency response filters are classified as low pass,high pass,band pass and band reject filters.
13. The ideal filters are non causal and hence physically unrealizable.
14. The transition of the frequency response from passband to stopband defines the transition region of the filter.
15. Linear phase filter requires the filter to have both constant group delay and constant phase delay.
16. The frequency response of the filter is the fourier transform of the impulse response of the filter.
17. In linear phase filters when impulse response is anti-symmetric and N is odd , the magnitude function is anti
symmetric.
18. In linear phase filters when impulse response is anti-symmetric and N is even , the magnitude function is
symmetric
19. In linear phase filters when impulse response is symmetric and N is odd , the magnitude function is symmetric
20. In linear phase filters when impulse response is symmetric and N is even , the magnitude function is anti
symmetric.
21. The abrupt truncation of the impulse response leads to oscillations in passband and stopband.
22. The generation of oscillations due to slow convergence of the fourier series near the points of discontinuity is
called Gibbs phenomenon.
23. In fourier series method of FIR filter design,the causality is brought about by multiplying the transfer function with
Z-(N-1)/2
24. The width of the main lobe in window spectrum can be reduced by increasing the length of window sequence
25. The width of the transition region of FIR filter directly depends on the width of main lobe in window spectrum.
26. Gibbs oscillations can be eliminated by replacing the sharp transitions in window sequence by gradual transition.
27. In rectangular window the width of main lobe is equal to 4/N
28. In hamming window spectrum the width of main lobe is double that of rectangular window for the same value of
N.
29. In blackmann window spectrum the width of main lobe is triple that of rectangular window for the same value of
N.
30. In kaiser window spectrum the side lobe magnitude is variable.
31. Blackmann window spectrum has the highest attenuation for side lobes.
32. In hamming window spectrum the increase in side lobe attenuation is achieved at the expense of constant at high
frequencies.
33. In blackmann window spectrum the higher side lobe attenuation is achieved at the expense of increased main lobe
width.
34. The ideal filters are ____________
a) Causal b) Non causal c) May be causal of may not be causal d) None ( )
35. In fourier series method to get the transfer function of realizable filter , H(z) is to be multiplied by
a)Z-(N-1)/2 b) Z(N-1)/2 c )Z-(N-1) d) Z(N-1)
36. The abrupt truncation of fourier series results in oscillations in
a)pass band b) stop band c) both pass and stop bands d) None
37. The frequency response of a digital filter is __________
a) periodic b) non periodic c) may periodic or non periodic d) None
38. For rectangular the main lobe width is equal to________
a)2/N b) 4/N c) 8/N d)12/N
39. For hanning window the main lobe is equal is to _____________
a)2/N b) 4/N c) 8/N d)12/N
40. For hamming window the main lobe is equal is to _____________
a)2/N b) 4/N c) 8/N d)12/N
41. For blackmann window the main lobe is equal is to _____________
a)2/N b) 4/N c) 8/N d)12/N
42. For kaiser window the main lobe is equal is to _____________
a)adjustable b) 4/N c) 8/N d)12/N
43.For rectangular window the peak side lobe magnitude in dB is
a)-13 b)-31 c)-41 d)-58
44. For hanning window the peak side lobe magnitude in dB is
a)-13 b)-31 c)-41 d)-58
45. For hamming window the peak side lobe magnitude in dB is
a)-13 b)-31 c)-41 d)-58
46. For blackmann window the peak side lobe magnitude in dB is
a)-13 b)-31 c)-41 d)-58
UNIT V
1. Show that the signal x(n) can recovered from its sample x(mD) if the sampling frequency
s 2 / D 2 m ( f s 1 2 fm )
D
2. Show that x(n) can be reconstructed using the formula

x ( n) x(kD)h (n kD)
r
a. k
i. Where
sin(2 f c n
hr ( n ) f m fc f s f m
2 n
b.
m c s m
3. Define interpolation and Decimation .Listout the advantages of sampling rate conversion .
4. a. Explain the implementation of polyphase structure for interpolator

b.Explain the process of Decimation using relevant expressions and block diagram
c.Expalin multirate Signal processing
5. Explain the process of interpolation by a factor of I and also discuss how the images are eliminated with a neat
blockdiagram
6. a.Discuss Finite word length effects of implementation of FFT
b.Discuss the effects due to finite wor length in direct form-I and II structures
7. What is meant by Overflow Error and How it can be Avoided?
8. Plot the signals and their corresponding spectra for rational sampling rate conversion by
(a )I/D =5/3
(b) I/D=3/5
9. Assume that the spectra of input signal x(n) occupies the entirerange -x
10. (a)Discuss the sampling Rate Conversion by a factor I/D
b) A Sequence x(n) is upsampled by I=2 , it passes through an LTI System H1(z) and then
downsampled by D=2 . Can we replace this process with a single LTI System H2(z) ?if yes,
Determine the system function of this system.
11. Explain the process of intrerfacing of digital systems eith different sampling rates with a neat block diagram
12. a.What are the Deadband Effects?Discuss.
1. Define interpolation and Decimation .Listout the advantages of sampling rate conversion .
(April-2011)
2. Explain the implementation of polyphase structure for interpolator (April-2011)
3. Explain the process of Decimation using relevant expressions and block diagram (April-2011)
4. Expalinmultirate Signal processing (April-2012)
5. Explain the process of interpolation by a factor of I and also discuss how the images areeliminated with a neat
blockdiagram (April-2012)
6. Discuss Finite word length effects of implementation of FFT algorithms (April-2012)
7. Discuss the effects due to finite wor length in direct form-I and II structures (April-2012)
8. What is meant by Overflow Error and How it can be Avoided? (April-2012)
9. Plot the signals and their corresponding spectra for rational sampling rate conversion by
(a )I/D =5/3
(b) I/D=3/5 (April-2012)
Assume that the spectra of input signal x(n) occupies the entirerange -x
10. (a)Discuss the sampling Rate Conversion by a factor I/D
b) A Sequence x(n) is upsampled by I=2 , it passes through an LTI System H1(z) and then
downsampled by D=2 . Can we replace this process with a single LTI SystemH2(z) ?if yes,
Determine the system function of this system. (May- 2013)
11. Explain the process of intrerfacing of digital systems eith different sampling rates with a neat block diagram
(NOV-2013)
12. (a)Explain the process of decimation using relevant expressions and blockdiagram
b) Explain the implementation of polyphase filter structure for interpolator. (JUNE-2014)
13.. Short notes on
a) Limit Cycles
b) Overflow Oscillations
c) Dead band effects (JUNE-2014)
14. (a) Consider a single stage interpolator with the following specifications:
Original Sampling rate=1Khz
Interpopolator Factor L=2
Frequency of interest =0-150 Hz
Passband ripple= 0.02 db
Stopband Attenuation=45 db
i) Draw the blockdiagram for the interpolator
ii) Determine the window type filter length and cutoff freq, if the window method is used for the Anti-image FIR Filter
design
(b) Explain the MultiRate Signal Processing an give its examples
(May- 2015)
15. (a)Explain the characteristics of a limit cycle Oscillation with respect to the system described by the
equation y(n)=0.85y(n-2)+0.72y(n-1)+x(n)
(b) Determine the deadband of the filter x(n)-3/4(n) (May- 2015)
16. (a) What are the DeadbandEffects?Discuss.
(b) What is meant by sampling rate conversion?Explain.
(May- 2016)
17. What are Limit Cycles and discuss various types of Limit cycles in brief (May- 2016)
18.Consider a second order IIR filter H(Z)=1/(1-0.5z-1)(1-0.45z-1) . Find the effect on quantization on pole locations of
the given system function in direct form and cascade form. Assume b=3 bits. (Nov/Dec -2016)
19. Explain how reduction of product round-off error is achieved in digital filters? (Nov/Dec -2016)
20. Explain in fetail decimation and interpolation? (Nov/Dec -2016)

1. In single ratesystems single sampling rate is used .
2. The two basic operations in multi rate signal processing are Decimation and Interpolation
3. Decimation reduces the sampling rate ,whereas Interpolation
increases the sampling rate .
4. Down sampling reduces the sampling rate whereas Upsampling increases the sampling rate.
5. The complete process of Filtering and then Down sampling is referre to as decimation.
6. The filter used to band limit the signal prior to down sampling is called as Anti Aliasing filter.
7. The sampling rate of a discrete time signal can be Increased by a factor I by placing I-1 equally spaced zeros
between each pair of samples.
8. The up sampler and down sampler are time Variant systems.
9. The phenomenon of getting image spectra in the output of an up sampler in addition to the scaled input spectra is
called Imaging.
10. The additional spectra introduced at the output of an up sampler is called Image Spectra
11. Interpolation is the complete process of UpSampling and filtering to remove image spectra.
12. The low pass filter which is used after the up sampler to remove the image spectra is called the Anti imaging
filter.
13. The scaling of discreter time signals and their addition at the nodes are independent of the Sampling rate.
14. The delay of D sample periods before a down sampler in the same as a delay of one sample period after the down
sampler.
15. A sampling rate conversion by a factor I/D can be achieved by Cascading a factor of I interpolator and factor of D
decimator.
16. A cascade of a factor of D down sampler and a factor of I up sampler is interchangeable with no change in the
input an output relation if and only if I and D are Co Prime
17. The Transpose of a decimator is an interpolator anviceversa.
18. Filter banks may be Analysis-filter banks or Synthesis
19. The D-channel synthesis filter bank is the Dual of D-channel analysisfilter bank.
20. In digital audio the different sampling rates used are 32kHz for broadcasting 44.1 kHz for compact disc and 48 kHz
for audio tape.
21. While designing multirate systems effects of Aliasing for decimation and Pseudo Images for interpolation should
be provided.
22. The decimator is also known as sub, down sampler or under sampler.
23. The reciprocal of the Nyquist rate is called the Nyquist period
24. In sampling rate conversion first Interpolation is to be performed and then Decimation is to be performed.
25. Decimation results in
a) Decrease in sampling rate
b) increase in sampling rate
c) No change in sampling rate
d) Random change in sampling rate
26. Interpolation results in
a) Decrease in sampling rate
b) increase in sampling rate
c) No change in sampling rate
d) Random change in sampling rate
27. Anti aliasing filter is to be kept
a) Before down sampler
b) After down sampler
c) Before up sampler
d) After up sampler
28. Up sampler and down sampler are
a) Time varying systems
b) Time invariant systems
c) Unpredicatable systems
d) May be time varying of time invariant
29. Down sampling by a factor of D skips
a) D samples
b) D-1 samples
c) No samples
d) D/2 samples
30. Up sampling by a factor of I introduces
a) I zeros between samples
b) I-1 zeros between samples
c) No zeros
d) I/2 zeros between samples
31. Down sampling by a factor of D introduces how many additional images.

a) D images
b) D-1 images
c) No images
d) D/2 images
32. Up sampling by a factor of D introduces how many additional images?

a) I images
b) I-1 images
c) No images
d) I/2 images
33. If x(n) ={ 1,2,3,4,5,6,7,} then x(n/2)=
a) {1,0,2,0,3,0,4,0,5,0,6,0,}
b) {1/2,2/2,3/2,4/2,5/2,6/2,.}
c){1,3,5,7,.}
d) {2,4,6,8,10,}
34. If x(n)={1,2,3,4,5,6,7,..}then x(2n)=
a) {2,4,6,8,10,.}
b){1,0,2,0,3,0,4,0,5,0.}
c) {1,3,5,7,..}
d) {1,0,0,2,0,0,3,0,0,4,0,0,5,0,0}
35. Roundoffnoiseis that errorin the lter output that resultsfrom rounding or
truncatingcalculations within the lter.
36. for recursive lterswith a zero or constant input, this nonlinearity can cause
spurious oscillations called limit cycles.
37. The term overow oscillation, sometimes also called adder overow limit Cycle.
38.input quantization error, product quantization error, coefficient quantization error the errors arises due to
quantization of numbers.
39. truncation error is introduced when the number is represented by reduced no.of bits.
40. rounding error is introduced whenever the number is rounded off to the nearest digital level.
41.Dead band is the range of output amplitudes over which limit cycle oscillations takes place.
42. Overflow can be avoided by scaling the internal signal levels with the help of scaling multipliers.
43. Zero input Limit cycle Oscillations, Overflow Limit cycle Oscillations are two types of limit cycles.
44. Saturation arithmetic eliminates limit cycles due to overflow.
45. When the addition operation is performed in the stable IIR filter,its output oscillates between max and min amplitude,
such oscillations are called as Overflow Limit cycle Oscillations

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DIGITAL SIGNAL PROCESSING

Able to obtain different Continuous and Discrete time signals.

Discrete Fourier series and Fast Fourier Transform

Fast Fourier Transforms (FFT)

T3:LoneyLudermanFundamentals of Digital Signal Processing. John wiley, 2009

R3: S. SalivahananDigital Signal Processing TMH, 2000

R6: "Digital Signal Processing" -RAMESH BABU Ed- 4thedition,scitech publications

UNIT -I: Introduction and Realization of Digital Filters

T1: 2.2; T2: BB

Linear Constant Coefficient T1: 2.4.3 ; T2: BB

L8 System Function, Frequency Response BB

Explanation for the use of Difference BB

Frequency Response of Stable To compute and Plot Amplitude and BB

Realization of Digital Filter using Form I T1: 7.3.1 ; OHP

Realization of Digital FILTER Using T1: 7.3.3 ; OHP

T1: 7.3.4 ; OHP

UNIT -II: Discrete Fourier Series and Fast Fourier Transform

DFS Representation of Periodic DFS Representation of T2: 8.2; R1:3.2 BB

T2: 8.3 ; R6: BB

T1: 7.1 ; T2: BB

Linear Convolution of Sequences T2: 8.9; R6: BB

Concept of Butterfly Diagram OHP

Solution using Butterfly T1: 6.1.3 ; T2: BB

T2: 9.6 ; R1:

UNIT -III: IIR Digital Filters

Explanation for the use of BB

Need to convert in Digital BB

Step & Impulse Invariance Backward Derivative, Forward R6: 5.12.4 ; BB

Low Pass to High Pass , Low R6: 5.13 ; BB

UNIT -IV: FIR Digital Filters

Filter Definition , OHP

Frequency Response L44 Frequency Response BB

Design of FIR Filter using BB

Comparison of IIT and FIR Comparison of IIR and FIR OHP

UNIT -V: Multirate Digital Signal Processing

Concept and Formulation of T1:10.2 ; R6: BB

Concept and Formulation of T1: 10.3 ; R6: BB

T1: 10.4 ; R6: BB

Sampling rate conversion of OHP

Design of phase splitter, BB

Introduction to Limit Cycles BB

Computational output round off BB

R3: 10.9 ; R6: OHP

Measurement of Coefficient Measurement of Coefficient BB

T3:LoneyLudermanFundamentals of Digital Signal Processing. John wiley, 2009

R3: S. SalivahananDigital Signal Processing TMH, 2000

R6: "Digital Signal Processing" -RAMESH BABU Ed- 4thedition,scitech publications

a) (n)=u(n)-u(n-1) b) u(n)= (k)(and k=-to n)=(n-k)(and k=0 to )

4) Compute the convolution y(n)=x(n)*h(n) of the following signals

6) Determine the response of the relaxed system characterized by impulse response

8) find the convolution of the following sequences

x(n)=u(n) and h(n)=anu(-n-1),a>1

9) Compute and sketch the convolution y(n) of the sequences

1. x(n ) = {1,2,4} h(n ) = {1,1,1,1,1}

2. x(n ) = {1,2,3,4} h(n ) = {4,3,2,1}

Y (n) 4y(n-1) = x(n ) - x(n - 1 )

11) a) Determine if the system is time invariant or not

i. y(n) = T[x(n)] = x(n) x(n-1)

ii. y(n) = T[x(n)] = nx(n)

1. y(n) = nx(n) 2. y(n)= x(n) 3. y(n ) = x(2n)

(b) A digital system is characterized by the following difference equation:

17) An LTI System is described by the equation y(n)=x(n)+0.8x(n-1)+0.7x(n2)

followingdifference equation y(n)=0.6y(n-1)-0.08y(n-2)+x(n).