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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:
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:
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.
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
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
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
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
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
T1:2.2.6 ; 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 ;
T1: 4.2.1 , BB
4.2.3 ;
T2: 2.6 ; R1:
L7 Impulse and Step Response
Response of Stable System 1.5
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
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;
T1: 7.2; BB
T2:8.7;
Properties of DFT L21 Linearity , Shifting Property
R6: 3.6 ; R3:
6.3.2
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
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
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
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
R6: 5.12.3 ; BB
Bilinear Transformation Method L39 Mathematical Representation
T1: 8.3.3
R6: 5.12.2 ; BB
LTI L41 Impulse Invariance Techniques
T1: 8.3.2
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
T1: 10.1 ; BB
Introduction and application of
Introduction L49 R6:8.1; R3:
Multirate DSP
11.1
Trade between Round off and Trade between Round off and BB
L58 R3: 10.5
Overflow Noise Overflow Noise
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
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
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
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 :
3) Consider the system y(n)=T[x(n)]=x(n) determine whether the system is invariant or time variant?
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)
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)
10) a) Determine the impulse response of system described by second order difference equation
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
b) Determine if the systems are linear, non linear, causal, and non causal
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)
y(n)=x(n)+ay(n-1) assuming that the system is relaxed initially , determine its impulse response?
18a ) Determine the impulse and unit step response of the systems described by the
2 Z 3 Z 3 Z 1
(i) (ii) (iii)
x ( n ) { 3, 2, 1, 0,1}
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)
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)
a) h(n) =0 for n=0 b) h(n) =0 for n>0 c) h(n) =0 for n<0 d) none
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
10. if x(n) is a causal sequence then the ROC is the entire Z plane except at [ ]
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. 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
(a) Binary code (b) Decimal code (c) Hexa decimal code (d) Octal code
(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=
Kz Kz Kz Kz 1
18. If the system output at any time n depends on future inputs or outputs then the system is
called [ ]
(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.
23. A signal which can be described by a mathematical equation is called a Deterministic signal.
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.
31.The DFT is obtained by sampling one period of the fourier transform X(w)
34. (n)=u(n)-u(n-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
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.
46.A System which does not have an unique relation between input and output is called Non Invertible.
49. For a discrete time system to be stable, its impulse response must be Absolutely Summable.
Assignment questions
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)
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.
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
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
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
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.
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
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
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
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
48. The no.of complex multiplications involved in the computation of 256-point DFT by radix-2 FFT
Is 1024
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.
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.
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?
14. What are the advantages and disadvantages of the bilinear transformation?
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?
(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)
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
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
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
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.
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.
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.
Assignment Questions
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.
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
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?
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. 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)
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.
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