Bhoj Reddy Engineering College for Women

III B Tech II Sem., Objective (Question Bank) Questions with Answers

Branch and Section: ECE A, B & C Subject: Digital Signal Processing DSP

UNIT-I

Choose the correct alternative:

1. y(n)=x(n+2) is [ c ]
a) Linear system b) Dynamic system c) both (a) and (b) d) Non
causal system
2. y(n)=x(2n) is [ c ]
a) Time-invariant system b) Non Linear system c) Linear system d) Static system
3. y(n)=x(-n) is [ a ]
a) Linear system b) Causal system c) Time-invariant system
d) Static system
4. The process of converting a signal from a given rate to a different rate is known as
[ c ]
a) Sampling b) Normalizing
c) Sampling rate conversion d) Quantisation
5. In which of the following, sampling rate conversion is used? [ d ]
a) Narrow band filters b) Digital filter banks
c) Quadrature mirror filters d) All of the above
6. A system whose output y(n) at time n depends on any number of past output values is
called a [ a ]
a) Recursive system b) Non-recursive system
c) Causal system d) Non-Causal system
7. A system whose output y(n) at time n depends only on present and past input values is
called a [ b ]
a) Recursive system b) Non-recursive system
c) Causal system d) Non-Causal system
8. The process of reducing the sampling rate by a factor D is called as [ a ]
a) Decimation b) Interpolation
c) Sampling rate conversion d) Sub band coding

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9. If Fx and Fy are the sampling rates of the input and output signals respectively, then
what is the value of Fy/Fx? [ b ]
a) D-I b) I/D c) I.D d) I2D
10. Which of the following is the advantage of sampling rate conversion by converting the
signal into analog signal? [ b ]
a) Less signal distortion b) New sampling rate can be arbitrarily selected
c) Quantization effects d)All of the above

Fill in the Blanks

11. Discrete time signals are discrete in time and continuous in amplitude.

12. Static systems are also called memoryless systems.

13. Dynamic systems are also called memory systems.

14. A causal system is one whose output depends on present and past values of input.

15. A system is said to be stable if every bounded input produces a bounded output.

16. The process of increasing the sampling rate of a signal by a factor I is called
Interpolation.

17. The systems that employ multiple sampling rates are called multi-rate DSP systems.
(True/False) True

18. A system whose output y(n) at time n depends only on the present and past inputs is
called a non-recursive system.

19. The Frequency-domain relationships can be obtained by combining the results of the
interpolation and decimation processes.

20. The basic element used to construct the block diagram of a discrete-time system are
adder, constant multiplier, unit delay element.

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UNIT-II

Choose the correct alternative:

1) Discrete Fourier Transform is applied to [ b ]
a. Infinite sequences b) Finite discrete sequences
b. Continuous infinite signals d) Continuous finite sequences
2) Discrete Time Fourier Transform is a periodic function with a period of [ c ]
a.  b) 0 c) 2 d) infinity
3) The Discrete Time Fourier Transform is the Z-transform evaluated along the [ c ]
a. a) Imaginary axis of z-plane b) real axis of z-plane
c)unit circle in z-plane d) entire z-plane
4) If the DFT of a real signal x(n) is X(k)={1, 2-j, A, 5, 3+j2, 2+j}, the value of A is[b]
a. 3+j2 b) 3-j2 c)-3-j2 d) -3+j2
5) Let x(n)={0, 1, A, 2,3,4}. If X(0)=10, A=______________ [ a ]
a. A=0 b) A=1 c) A=2 d) A=3
6) The Discrete Fourier Transform of a 2 sample sequence x(n)= {4,2} is [ d ]
a. {2, 1} b) {4, 2} c) {8, 4} d) {6, 2}
7) The number of stages in the computation of 1024-point Discrete Fourier Transform
by radix-2 Fast Fourier Transform is [ d ]
8) a)1024 b) 32 c) 8 d) 10
9) The number of butterflies in each stage of computation of 64-point radix-2 Fast
Fourier Transform is [ b ]
a. 64 b) 32 c) 16 d) 8
10) For radix-2 Fast Fourier Transform, N must be a power of [ c ]
a. N b) 4 c) 2 d) N/2
11) The number of complex multiplications involved in the computation of 256-point
Discrete Fourier Transform by radix-2 Fast Fourier Transform is [ b ]
a. 256 b) 1024 c) 512 d) 128

Fill in the Blanks

1) The Discrete Time Fourier Transform is a periodic continuous function of  with a
period of 2.

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 2) The Discrete Time Fourier Transform is nothing but the Z-transform evaluated at a
finite number of equally spaced points on the unit circle centred at the origin of z-
plane.

3) ___________ is known as the twiddle factor.

4) The periodicity property of Discrete Fourier Transform says that X(k)=X(k+N).

5) The linearity property of Discrete Fourier Transform says that

a. DFT{ax1(n)+bx2(n)}=aX1(k)+bX2(k).

6) The Fast Fourier Transform may be defined as an algorithm for computing Discrete
Fourier Transform.

7) In Fast Fourier Transform, the computational efficiency is achieved by adopting a

divide and conquer approach.

8) The basic Fast Fourier Transform algorithms are DIT FFT and DIF FFT.

9) For DIT Fast Fourier Transform, the input is in bit reversed order and the output is
in normal order.

10) For DIF Fast Fourier Transform, the input is in normal order and the output is in bit
reversed order.

UNIT-III

Choose the correct alternative:

1) Advantages of IIR filter [ a ]
a) Infinite sequences b) Finite discrete sequences
c) Continuous infinite signals d) Continuous finite sequences

2) The cascade realisation of IIR systems involves [ b ]

1) The transfer function broken into product of transfer functions
2) The transfer function divided into addition of transfer functions
3) Factoring the numerator and denominator polynomials
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 4) Derivatives of the transfer functions
a) 1, 2 and 3 are correct
b) 1 and 3 are correct
c) 3 and 4 are correct
d) All the four are correct

3) In direct form for realisation of IIR filters, [ c ]

1) Denominator coefficients are the multipliers in the feed forward paths
2) Multipliers in the feedback paths are the positives of the denominator coefficients
3) Numerator coefficients are the multipliers in the feed forward paths
4) Multipliers in the feedback paths are the negatives of the denominator
coefficients
a) 1, 2 and 3 are correct
b) 1 and 2 are correct
c) 3 and 4 are correct
d) All the four are correct

4) The Chebyshev filters have [ c ]

1) Flat pass band
2) Flat stop band
3) Equiripple pass band
4) Tapering stop band
a) 1 and 2 are correct
b) 2 and 4 are correct
c) 2 and 3 are correct
d) All the four are correct

5) The magnitude response of Butterworth filter has [ a ]

1) Flat stop band
2) Flat pass band
3) Tapering pass band
4) Tapering stop band
a) 1 and 2 are correct
b) 2 and 4 are correct
c) 2 and 3 are correct
d) All the four are correct
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6) The impulse invariant method is obtained by [ a ]
a) Sampling the impulse response of an equivalent analog filter
b) Taking backward difference for the derivative
c) Mapping from s-domain to z-domain
d) Approximation of derivative

7) Th filter that may not be realized by approximation of derivatives techniques are

1) Band pass filters [ b ]
2) High pass filters
3) Low pass filters
4) Band reject filters
a) 1, 2 and 3 are correct
b) 2 and 4 are correct
c) 2 and 3 are correct
d) All the four are correct

8) IIR filter design by approximation of derivatives has the limitations [ c ]

1) Used only for transforming analog high pass filters
2) Used for band pass filters having smaller resonant frequencies
3) Used only for transforming analog low pass filters
4) Used for band pass filters having high resonant frequencies
a)1, 2 and 3 are correct
b)1 and 2 are correct
c) 2 and 3 are correct
d) All the four are correct

9) The IIR filter designing involves [ b ]

a) Designing of analog filter in analog domain and transforming into digital domain
b) Designing of digital filter in analog domain and transforming into digital domain
c) Designing of analog filter in digital domain and transforming into analog domain
d) Designing of digital filter in digital domain and transforming into analog domain

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10) For a system function H(s) to be stable [ c ]
a) The zeros lie in left half of the s plane
b) The zeros lie in right half of the s plane
c) The poles lie in left half of the s plane
d) The poles lie in right half of the s plane

Fill in the Blanks

1) The magnitude response of the Butterworth filter decreases as the frequency
increases from zero to infinite.

2) The poles of the Chebyshev filter lie on the ellipse.

3) Bilinear transformation avoids aliasing effect in frequency components.

4) Stable continuous system can be mapped into realizable stable digital systems using
Bilinear transformation.

5) Analog filters are constructed from active and passive components.

6) The transition band is more in Butterworth filter when compared to Chebyshev filter

7) In the frequency response characteristics of FIR filter, the number of bits per
coefficient should be increased in order to maintain the same error.

8) In IIR Filter design by the Bilinear Transformation, the Bilinear Transformation is a

mapping from S-plane to Z-plane.

9) Transition band is the region between stop band and the pass band frequencies in the
magnitude frequency response of a low pass filter.

10) In bilinear transformation, the left-half s-plane is mapped inside the unit circle |z|=1
in the z-domain.

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UNIT-IV

Choose the correct alternative:

1) Which among them is an advantage of FIR Filter? [ c ]

a) Necessity of computational techniques for filter implementation
b) Requirement of large storage
c) Incapability of simulating prototype analog filters
d) Presence of linear phase response

2) What is the general system function of an FIR system? [ c ]

3) Which of the following is an method for implementing an FIR system? [ d ]

a) Direct form
b) Cascade form
c) Lattice structure
d) All of the mentioned

4) The realization of FIR filter by frequency sampling realization can be viewed as

cascade of how many filters? [ a ]
a) Two
b) Three
c) Four
d) None of the mentioned

5) The zeros of the system function of comb filter are located: [ b ]

a) Inside unit circle
b) On unit circle
c) Outside unit circle
d) None of the mentioned

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 6) Which of the following is the first method proposed for design of FIR filters?
[ c ]
a) Chebyshev approximation
b) Frequency sampling method
c) Windowing technique
d) None of the mentioned

7) Which of the following technique is more preferable for design of linear phase FIR
filter? [ b ]
a) Window design
b) Chebyshev approximation
c) Frequency sampling
d) None of the mentioned

8) The anti-symmetric condition with M even is not used in the design of which of the
following linear-phase FIR filter? [ a ]
a) Low pass
b) High pass
c) Band pass
d) Bans stop

9) A filter is said to be linear phase filter if the phase delay and group delay are [ d ]
a) High
b) Moderate
c) Low
d) Constant

10) In FIR filters, which among the following parameters remains unaffected by the
quantization effect? [ d ]
a) Magnitude Response
b) Phase Characteristics
c) Both a and b
d) None of the above

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Fill in the Blanks

1) The difference equation of the FIR filter of length M, input x(n) and output y(n) is

2) The unit sample response of a FIR filter inorder to have a linear phase should satisfy
±h(M-1-n) n=0,1,2…M-1

3) FIR filters are always stable because all its pole are at the origin.

4) FIR filters can be easily designed to have a perfect linear phase.

5) The anti-symmetric condition with M even is not used in the design of low pass
linear-phase FIR filter.

6) The central lobe of the frequency response of the window should be narrow with
more energy.

7) FIR filters are of non recursive type where as IIR filters are of recursive types.

8) Filters with any arbitrary magnitude response can be realized using FIR sequence.

9) The major methods for designing FIR filters with linear phase are windows method,
frequency sampling method and optimal or minimax design.

10) In Barlett window, the triangular function resembles the tapering of rectangular
window sequence linearly from the middle to the ends.

11) In FIR filter design, parameters that are separately controlled by using Kaiser
window are order of filter (M) and transition width of main lobe.

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UNIT-V

Choose the correct alternative:

1. The only signal whose ROC is the entire z-plane is [ a ]

a) (n) b) u(n) c) r(n) d) an
1
2. The Z-transform of a signal with X(s)= 𝑠
is [ a ]
𝑧 1 1 1
a. b) c) d)
𝑧−1 𝑧−1 1−𝑧 1+𝑧

3. Which of the following use quadrature mirror filters? [ c ]

a) Sub band coding b) Trans-multiplexer
c) Both a and b d) None of the above
4. The structure which uses less number of delay elements is [ b ]
a) Direct form-I b) Direct form-II
c) Cascade form d) Parallel form
5. The number of multipliers required for the realization of Finite Impulse Response
systems is reduced if we choose [ d ]
a) Direct form b) Cascade form
c) Parallel form d) Linear phase realization
6. _________structure provides a direct relation between time domain and z-domain
equations. [ a ]
a) Direct form-I b) Direct form-II c) Cascade d) Parallel
7. The sampling rate of 2fh samples per second is the ______and its reciprocal ½fh is
the ______ [ d ]
a) Sampling period, Sampling rate b) Sampling rate, Sampling period

c) Nyquist period,Nyquist rate d) Nyquist rate, Nyquist period

8. Which of the following realization is less sensitive to the process of quantization
[ c ]
a) Direct form b) Canonic form
c) Cascade form d) Parallel form
9. Error due to finite word length registers in digital filters is/are [ d ]
a) Input quantization error b) Coefficient quantization error
c) Product quantization error d) All of the above
10.If the z-transform of a sequence x(n) is X(Z) then the z-transform of a sequence
upsampled by a factor L is [ a ]
a) Y(z)=X(zL) b) Y(z)=X(z/L) c) Y(z)=X(z*L) d) Y(z)=X(z-L)

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Fill in the Blanks

11.Truncation is defined as the process of discarding all bits less significant than least
significant bit that is retained.

12.Zero input limit cycles and Overflow limit cycles are the types of limit cycles.

13.Zero input limit cycles are of lower amplitudes in comparison with overflow limit
cycles.

14.In practice, 1-Dimensional Finite Impulse Response (FIR) filters are employed in
filtering problems where there is a requirement for a linear-phase characteristic
within the passband of the filter.

15.The dead band of a single pole filter with a pole at 1/2 and represented by 4 bits is
(-1/16, 1/16).

16.The range of values of z for which X(z) converges is called the ROC of X(z).

17.The frequency response of a system is obtained by substituting ________ in H(z).

18.Zero input limit cycles are of lower amplitudes in comparison with overflow limit
cycles.

19.The Overflow error occurs when two large numbers of the same sign are added and
the result exceeds the word-length.

20.Limit cycles can occur due to overflow in digital filters implemented with finite
precision arithmetic.

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 Bhoj Reddy Engineering College for Women, Hyderabad

Department of Electronics and Communication Engineering

DIGITAL SIGNAL PROCESSING (DSP)

Academic Year: 2020-2021 (R18) III BTech II semester Branch: ECE A, B and C

UNIT-I:

Introduction: Introduction to Digital Signal Processing: Discrete Time Signals & Sequences, conversion
of continuous to discrete signal, Normalized Frequency, Linear Shift Invariant Systems, Stability, and
Causality, linear differential equation to difference equation, Linear Constant Coefficient Difference
Equations, Frequency Domain Representation of Discrete Time Signals and Systems

Multirate Digital Signal Processing: Introduction, Down Sampling, Decimation, Upsampling,

Interpolation, Sampling Rate Conversion, Conversion of Band Pass Signals, Concept of Resampling,
Applications of Multi Rate Signal Processing.

TEXT BOOKS:
B1: Digital Signal Processing, Principles, Algorithms, and Applications: John G. Proakis, Dimitris
G. Manolakis. Pearson Education / PHI. 2007.
B2: Discrete Time Signal Processing-A. V. Oppenheim and R.W. Schaffer. PHI, 2009
B3: Digital Signal Processing - P. Ramesh Babu, Fourth Edition.
B4: Digital Signal Processing – A.Anand Kumar –PHI Learning 2013.
B5: Digital Signal Processing - S.Salivahanan. and A.Vallavaraj TMH.2003.

Question Bank:

1. Define various elementary discrete-time signals, indicate them graphically. (B4, PP: 5)
2. What are the Basic Elements of a Digital Signal Processing System explain with block diagram?
(B1, PP: 4)
3. What are Advantages and Limitations of Digital Signal Processing over Analog Signal Processing
(B1, PP: 5)
4. What are the classification of Discrete-Time Signals and explain each classification with example.
(B3, PP: 1.33)
5. What are the Basic Operations on Signals? (B3, PP: 1.43)
6. What are the classifications of Discrete –Time Systems? (B4. PP: 33)
7. Determine if the system described by the following input-output equations is linear or non-
linear (i) y(n)=x(n)+1/(x(n-1)) (ii) y(n)=x2(n) (iii) y(n)=nx(n) (B3, PP :1.55)
8. Determine if the following systems are time-invariant or time variant
9. (i) y(n)=x(n)+x(n-1) (ii) y(n)=x(-n) (B3, PP : 1.57)
10. What are stable and unstable systems ? (B4, PP: 49)
11. Derive the expression for Linear constant coefficient Difference Equations. (B3, PP: 1.83)
12. Determine the solution of the difference equation y(n) = 5/6y(n-1) – 1/6y(n-3) + x(n) for
x(n)=2nu(n) (B3, PP: 1.98)
13. What is Frequency Domain representation of Discrete Time Signals and Systems? (B3, PP: 1.99)

1
14. Define sampling rate conversion, Multirate digital Signal Processing?Explain the methods of
sampling rate conversion? (B1, pp: 750)
15. Explain the process of down sampling or decimation by a factor D ? (B1, pp: 755-758)
16. Explain the process of up sampling or interpolation ? (B1, pp: 755-758)
17. Discuss the general cases of sampling rate conversion by a factor I/D? (B1, pp: 763-765)
18. Consider a causal and stable LTI system whose difference equation is y(n)-1/6y(n-1)-1/6 y(n-
2)=x(n) .Determine impulse response h(n) for the system. (B3,pp:1.40)

19. The difference equation y(n)=5/6 y(n-1)-1/6 y(n-2)+x(n).Determine impulse response and also step
response of the system. ( B 3, pp:1.40)

20. Determine the response y(n), n≥ 0 of the system described by the second-order difference equation
Y(n)-4y(n-1)+4y(n-2)=x(n)-x(n-1) when the input is x(n)=(−1)𝑛 u(n) and the initial conditions are
y(-1)=y(-2)=1 (B3, pp:1.45)

UNIT-II:
Discrete Fourier series: Fourier series, Fourier Transform, Laplace Transform and Z-Transform
Relation, DFS Representation of Periodic Sequences, Properties of Discrete Fourier series, Discrete
Fourier Transforms: Properties of DFT, Linear Convolution of Sequences using DFT, Computation
of DFT: Over-Lap Add Method, Over-Lap Save Method, Relation between DTFT, DFS, DFT and
ZTransform.

Fast Fourier Transforms: Fast Fourier Transforms (FFT) - Radix-2 Decimation-in-Time and
Decimation-in-Frequency FFT Algorithms, Inverse FFT.

TEXT BOOKS:
B1: Digital Signal Processing, Principles, Algorithms, and Applications: John G. Proakis, Dimitris
G. Manolakis. Pearson Education / PHI. 2007.
B2: Discrete Time Signal Processing-A. V. Oppenheim and R.W. Schaffer. PHI, 2009
B3: Digital Signal Processing - P. Ramesh Babu, Fourth Edition.
B4: Digital Signal Processing – A.Anand Kumar –PHI Learning 2013.
B5. Digital Signal Processing - P. Ramesh Babu, Second Edition
B6.Digital Signal Processing-S.Salivahanan,C.Gnanapriya ,2nd Edition

Question Bank:

1. Write about the relation between DFT and Z-Transform, DTFT and DFT (B4, PP: 419,426)
2. Explain the properties of DFS (B4, PP:3.2)
3. Explain the following DFT properties (i)Linearity (ii)Time shifting (iii)Symmetry (B4, PP:418)
4. Explain the following DFT properties (i)Linearity (ii)Circular shift of a sequence (iii)DFT of Even
and Odd sequences (iv)Time reversal properties (B2, PP: 101)
5. Explain the following DFT properties (i)Complex Conjugate Property (ii)DFT of Delayed Sequence
(iii)DFT of Real Valued Sequence (B4, PP: 434)
6. Explain the following DFTproperties (i)Multiplication (ii)Circular convolution (iii)Parseval’s
Theorem (iv)Circular correlation (B5, PP: 436)
7. Find the DFT of a sequence x(n)={1, 1, 0, 1} and find the IDFT of Y(k)= {1, 0, 1, 0}
(B3, PP: 3.12)
8. Find IDFT of the sequence X(k) = {5, 0, 1-j, 0, 1+j, 0} (B3, PP: 3.21)
2
 9. What are the methods to evaluate circular convolution of two sequences (B3, PP:3.42)
10. Find the 8-point DFT of x(n) = {1, 1, 0, 0, 0, 0, 0, 0}. Use the property of conjugate symmetry
property (B4, PP: 430)
11. Find the circular convolution of the two sequences x1(n) = {1, 2, 2, 1} and x2(n) = {1, 2, 3, 1} using
(a) concentric circle method (b) matrix method (B3, PP: 3.45)
12. Find the output y(n) of a filter whose impulse response is h(n)={1, 1,1} and input signal x(n) = {3,
-1, 0, 1, 3, 2, 0, 1, 2, 1}using overlap-save method (B3, PP: 3.54)
13. Find the output y(n) of a filter whose impulse response is h(n)={1, 1,1} and input signal x(n) = {3,
-1, 0, 1, 3, 2, 0, 1, 2, 1}using overlap-add method (B3, PP: 3.54)
14. State the difference between overlap-save method and over add method, Linear Convolution and
circular convolution (B3, PP: 3.86, 3.88)
15. Compute the DFT of a sequence (-1)n for N=4 (B3, PP: 3.61)
16. Describe the efficient method for computation of DFT? (B1, PP: 511-512)
17. What is FFT?Bring out its 2 classes of FFT algorithms? (B5, PP: 4.2)
18. Illustrate the principle of Decimation in time FFT Algorithm? (B2, PP 290-294)
19. Illustrate the principle of Decimation in frequency FFT Algorithm? (B2, PP 290-294)
20. Summarize the steps of radix 2 DIT-FFT Algorithms? (B5, PP: 4.12)
21. Draw the flow graph of 16 point DIT-FFT? (B5, PP: 4.16)
22. Compare the computational complexity for the direct computation of DFT versus FFT Algorithm?
(B1, PP: 522)
23. List out the requirements for the direct computation of DFT? (B1, PP:512-8.1.1 )
24. What is bit reversal process?Tabulate the bit reversal process for N=8? (B5, PP: 4.10)
25. What are the applications of FFT? (B1, PP: 538)
26. Compute the 8 point DFT of the sequence x(n)=1 for 0<=n<=7, otherwise by using DIT,DIF
Algorithm? (B5, PP: 4.26)
27. Compute the 4 point DFT of the sequence x(n)={0,1,2,3}using DIT,DIF Algorithm? (B5, PP: 4.27)
28. If x(n)={1,2,3,4,4,3,2,1},Find X(k)using DIF FFT Algorithm? (B6, PP: 396)
29. Draw the basic butterfly diagram and flow graph for the computation of DIT -FFT and
DIF FFT ? (B6, PP: 388-396)
30. Determine DFT (8 point) for a continuous time signal x(t)=sin(2∏ft) with f=50HZ?(B6, PP: 402)
31. When do we use compose Radix FFT,Develop a radix-3 DIT and DIF-FFT algorithm for N=9?
(B6, PP: 404)
32. Develop a DIF-FFT algorithm for decomposing the DFT for N=6 and draw the flow diagrams
33. for a)N=3.2 b)N=2.3 (B6, PP: 415)
34. Given X(K)={36,-4+j9.656,-4+j4,-4+j6.56,-4,-4-j1.656,-4-j4,-4-j9.656},Find x(n)? (B6, PP: 402)
35. Describe the method of computing an inverse DFT by doing a direct DFT? (B6, PP: 399)

UNIT-III:

IIR Digital Filters: Analog filter approximations – Butterworth and Chebyshev, Design of IIR Digital
Filters from Analog Filters, Step and Impulse Invariant Techniques, Bilinear Transformation Method,
Spectral Transformations.

TEXT BOOKS:

B1. John G. Proakis, Dimitris G. Manolakis, “Digital signal processing, principles, Algorithms and
applications”, Pearson Education/PHI, 4th ed., 2007.
3
B2. Sanjit K Mitra, “ Digital signal processing , A computer base approach”, Tata Mcgraw Hill, 3rd
edition, 2009.
B3. A.V. Oppenheim and R.W. Schaffer, “Discrete Time Signal Processing”, 2nd ed., PHI.
B4. P. Ramesh Babu, “Digital Signal Processing”, Scitech Publications, 2nd ed, 2005.

Question Bank:

1. What are different types of frequency selective filters? Explain in detail. (B4, pp: 5.2)
2. Explain the indirect method used for designing IIR digital filters. (B4, pp: 5.4)
3. Differentiate analog and digital filters and list out advantages and disadvantages of digital filters.
i. (B4, pp:5.5)
4. Derive the order equation for low pass Butterworth analog filter. (B4, pp:5.7)
5. Given the specification α p =1 dB, α s =30 dB, Ω p =200 rad/sec, Ωs =600 rad/sec. Determine the
order of the filter. (B4, pp: 5.13)
Ωp Ωs
Ωc = =
((10 α p− 1) /2 N ) ((10 αs )1 /2 N )
0.1 1 0.1
6. Prove that (B4, pp: 5.14)
2
H (s)=
7. Apply bilinear transformation to [(s+1)(s+2)] with T=1 sec and find H(Z) (B4, pp: 5.52)
2
H (s)=
8. For the analog transfer function [(s+1)(s+2)] determine H(Z) using impulse invariance
method. Assume T=1 sec. (B4, pp: 5.53)
10
H ( S)= 2
9. An analog filter has a transfer function S + 7 S+ 10 . Design a digital filter eqivalent to this
using impulse invariant method. (B4, pp: 5.59)
α
10. Design a Chebyshev low pass filter with specifications p =1 dB ripple in the pass band 0 ω 2 Π
α
and s =15 dB in the stop band 0.3 Π⩽ω⩽ Π using (a). bilinear transformation (b). Impulse
Invariance (B4, pp: 5.91)
11. Design a Butterworth filter using the impulse invariance and impulse bilinear transformation method
for following specifications

a. 0.8⩽|H (e j ω )|⩽1 0⩽ ω⩽ 0.2 Π

b. H (e j ω )⩽0.2 0.6 Π⩽ ω⩽ Π (B4, pp: 5.97, 5.98)
(s+0.1)
H (S)=
12. Convert the analog filter with system function ((s+0.1)2+9) into a digital IIR filter by means
of impulse invariance method(B1, pp: 710)
(s+0.1)
H (S)=
13. Convert the analog filter with system function ((s+0.1)2+16) into a digital Iir filter by means
of bilinear transformation at resonant frequency П/2. (B1, pp:715)
14. Explain the basic concepts involved in frequency transformation in analog domain. (B1, pp: 730)
Ωp
H ( S)=
Transform the single pole LP Butterworth filter with system function S +Ω p into a
band pass filter with upper and lower band edge frequencies Ωu and Ω l respectively. (B1, pp:732)
4
15. Explain the basic concepts involved in frequency transformation in digital domain. (B1, pp:732)
(0.245(1+z− 1))
H (Z)=
(1− 0.509 z− 1)
16. Convert the single pole LP Butterworth filter with system function into aband
ω ω
pass filter with upper and lower cut off frequencies ω u and l respectively. The LP filter
has 3 dB bandwidth, ω p = 0.2 П (B1, pp: 733)
17. Compare Butterworth and Chebyshev filter (B4, pp: 5.24)
18. What are the steps to design analog Butterworth and Chebyshev low pass filter (B4, pp: 5.27)
19. Design a Chebyshev filter with a maximumpass band attenuation of 2.5 dB at Ω p =20 rad/sec and the
stop band attenuation of 30 dB at Ωs =50 rad/sec. (B4, pp: 5.28)
20. What are steps to design a digital filter using Impulse Invariance method (B4, pp: 5.37)
21. What are steps to design a digital filter using bilinear transform technique. (B4, pp: 5.42)
22. Design an analog Butterworth filter that has a 2 dB pass band attenuation at a frequency of 20
rad/sec an at-least 10 dB stop band attenuation at 30 rad/sec. (B4, pp: 5.15)

UNIT-IV:

FIR Digital Filters: Characteristics of FIR Digital Filters, Frequency Response, Design of FIR Filters:
Fourier Method, Digital Filters using Window Techniques, Frequency Sampling Technique, Comparison of
IIR & FIR filters.

TEXT BOOKS:

B1. John G. Proakis, Dimitris G. Manolakis, “Digital signal processing, principles, Algorithms and
applications”, Pearson Education/PHI, 4th ed., 2007.
B2. Sanjit K Mitra, “ Digital signal processing , A computer base approach”, Tata Mcgraw Hill, 3rd
edition, 2009.
B3. A.V. Oppenheim and R.W. Schaffer, “Discrete Time Signal Processing”, 2nd ed., PHI.
B4. P. Ramesh Babu, “Digital Signal Processing”, Scitech Publications, 2nd ed, 2005.

Question Bank:

1. Explain the symmetrical impulse response of FIR filters for N odd. (B4, pp: 6.7)
2. Explain the symmetrical impulse response of FIR filters for N even. (B4, pp: 6.8)
3. Explain the antisymmetrical impulse response of FIR filters for N odd. (B4, pp: 6.9)
4. Explain the antisymmetrical impulse response of FIR filters for N even. (B4, pp: 6.10)
5. Explain the Fourier series method in designing FIR filters (B4, pp: 6.16)
6. Design an ideal LPF with a frequency response d
H (e jω)= 1 for − Π /2 ⩽ ω ⩽ Π /2 and
H d (e jω)= 0 for − Π /2 ⩽ |ω|⩽ Π . Find the values of h(n) for N=11. Find H(Z). Plot the
magnitude response. (B4, pp: 6.18)
jω
7. Design an ideal HPF with a frequency response
H d (e )= 1 for − Π /4 ⩽ |ω|⩽ Π and
H d (e jω)= 0 for |ω|<Π /4 . Find the values of h(n) for N=11. Find H(Z). Plot the magnitude
response. (B4, pp: 6.22)

5
 8. Design an ideal BPF with a frequency response
H d (e jω)= 1 for − Π /4 ⩽ |ω|⩽ 3 Π /4 and
H d (e jω)= 0 otherwise. Find the values of h(n) for N=11. Find H(Z). Plot the frequency
response. (B4, pp: 6.25)
9. Design an ideal BRF with a frequency response
H d (e jω)= 1 for |ω|⩽ Π /3 and |ω|⩾2 Π /3
H d (e jω)= 0 otherwise. Find the values of h(n) for N=11. Find H(Z). Plot the magnitude response.
(B4, pp: 6.28)
10. Design an ideal HPF using (a). Hanning window and (b). Hamming window with a frequency
response d
H (e jω)= 1
for − Π /4 ⩽ |ω|⩽ Π and d
H (e jω)= 0
for |ω|<Π /4 . Find the values of
h(n) for N=11. Find H(Z). Plot the magnitude response. (B4, pp: 6.47)
11.
H (e jω)= e− j 3 ω
Design a filter with d for − Π /4 ⩽ ω ⩽ Π /4 and
H d (e jω)= 0 for Π /4 <|ω|⩽ Π
using Hanning window with N=7. (B4, pp: 6.53)
H d (e jω)= jω
12. Design a filter with for − Π ⩽ ω ⩽ Π using (i) Rectangular Window (ii)Hamming
window with N=7. Plot frequency response in both the cases. (B4, pp: 6.55)
jω
13. Design an ideal Hilbert transform having frequency response H (e )= j for − Π ⩽ ω ⩽ 0 and
H (e jω)= − j for 0 ⩽ ω ⩽ Π using (i). rectangular window (ii). Blackman window for N=11. Plot
frequency response in both cases. (B4, pp: 6.61)
14. Design a FIR LP filter satisfying the following specifications α p⩽0.1 dB , α s≥ 44.0 dB ,
ωp = 20 rad /sec ωs = 30 rad/sec and ωsf = 100 rad / sec (B4, pp: 6.72)
15. Design a FIR band pass digital filter satisfying the following specifications α p⩽0.1 dB ,
α s≥ 44.0 dB , f p1 = 20 Hz , f p2 = 30 Hz , f s 1= 10 Hz , f s 2= 40 Hz and F=100Hz. (B4, pp: 6.72)
α p= 0 .5 dB
16. Design a FIR band pass digital filter satisfying the following specifications ,
α s = 30 dB f p1 = 20 Hz f p2 = 30 Hz f s 1= 10 Hz f s 2= 40 Hz
, , , , and F=100Hz. (B4, pp: 6.76)
17. Determine the filter coefficients h(n) obtained by sampling

Hd(e jω)= e− j(N−1)/2 0⩽|ω|⩽ pi/2

Hd (e j)= 0 pi/2⩽|ω|⩽ pi for N=7.
(B4, pp: 6.89)
18. Using frequency sampling method design a band pass filter with the following specifications
sampling frequency F=8000Hz, Cut-off frequency fc1= 1000Hz, fc2=3000Hz. Determine the
filter coefficients for N=7. (B4, pp: 6.92)
19. Use frequency sampling method to design a FIR LPF with ωc=pi/4 for N=15. Plot the
magnitude response. Repeat part (a) by selecting an additional sample |H ( k)| =0.5 in transition
band . Plot magnitude response. (B4, pp: 6.94)
20. List out the differences between FIR and IIR filters. (B4, pp: 6.99)
jω − jαω
Hd(e )=e |ω|⩽ pi/6
H d (e j
)=0 pi/6⩽|ω|⩽ pi
21. Design a FIR filter approximating the ideal frequency response
22. Determine the filter coefficients for N=13. (B4, pp: 6.100)

6
23. Do the above example using Hamming window. (B4, pp: 6.101)
24. Determine the frequency response of FIR filter defined by y(n)=0.25x(n)+x(n-1)+0.25x(n-2).
Calculatethe phase delay and group delay. (B4, pp: 6.104)

UNIT-V:
Realization of 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.

Finite Word Length Effects: Limit cycles, Overflow Oscillations, Round-off Noise in IIR Digital Filters,
Computational Output Round off Noise, Methods to Prevent Overflow, Trade-off between Round Off and
Overflow Noise, Measurement of Coefficient Quantization Effects through Pole-Zero Movement, Dead
Band Effects.

TEXT BOOKS:
B1: Digital Signal Processing, Principles, Algorithms, and Applications: John G. Proakis, Dimitris G.
Manolakis. Pearson Education / PHI. 2007,4th edition
B2: Discrete Time Signal Processing-A. V. Oppenheim and R.W. Schaffer. PHI, 2009
B3: Digital Signal Processing - P. Ramesh Babu, Fourth Edition.
B4. Digital Signal Processing - P. Ramesh Babu, Second Edition
B5.Digital Signal Processing- S.Salivahanan, C.Gnanapriya,2nd Edition

Question Bank:
1. Show that if r=1 the Z-Transform expression is equal to the Fourier transform of the sequence.
(B2, PP: 46)
2. What are the Properties of ROC in Z-Transform with example (B1, PP: 148)
3. Explain the following z-Transform properties (i)Linearity (ii)Time shifting (iii)Multiplication
(B3, PP:2.8)
4. Explain the following z-Transform properties (i)Differentiation (ii)Time reversal (iii)convolution
theorem (B3, PP: 2.10)
5. Explain the following z-Transform properties (i)Complex Convolution Theorem (ii)Parseval’s
relation (iii)Correlation (B3, PP:2.12)
6. Explain initial value theorem and final value theorem with proof (B3, PP: 2.14)
7. Determine the inverse z-transform using power series expansion of X(z)=1/(1-1.5z-1+0.5z-2) when
ROC: |z|>1 and ROC:|z|<0.5 (B1, PP:182)
8. Determine the Partial fraction expansion of X(z) = (1-Z-1) / (1-z-1+-.5z-2) (B1, PP: 187)
9. Derive the expression for system function, and what are pole-zero system, zero system and pole
system? (B3, PP: 2.23)
10. Compute the response of the system y(n) = 0.7y(n-1) - 0.12y(n-2) + x(n-1) + x(n-2) to input
x(n)=nu(n). Is the system is stable? (B3, PP: 2.56)
11. Determine the frequency response, magnitude response, phase response and time delay of the
system given by y(n) + ½ y(n-1) = x(n) – x(n-1) (B5, PP: 224)
12. What are basic realization block diagram and the signal-flow graph and canonic and non-canonic
structures? (B5, PP:242)
13. Explain Direct form realization of IIR systems with block diagram (B5, PP:244)
14. Explain Cascade Realization of IIR systems with block diagram (B5, PP:251)
7
15. Obtain a cascade realization of the system characterized by the transfer function
16. H(z) = (2(z+2)) / (z(z-0.1) (z+0.5) (z+0.4)) (B5, PP: 252)
17. Explain Parallel Realization of IIR systems with block diagram (B5, PP: 253)
18. Determine the parallel realization of IIR digital filter transfer functions H(z) =
(3(2z2 + 5z + 4) / ((2z+1) (2z+2)) (B5, PP:254)
19. Expalin basic structures for FIR systems, Direct form Realisation of FIR systems, cascade form
Realization of FIR systems. (B5, PP: 271)
20. Define sampling rate conversion, MultiMate digital Signal Processing?Explain the methods of
sampling rate conversion? (B1, pp: 750)
21. Explain the process of down sampling or decimation by a factor D ? (B1, pp: 755-758)
22. Explain the process of up sampling or interpolation ? (B1, pp: 755-758)
23. Discuss the general cases of sampling rate conversion by a factor I/D? (B1, pp: 763-765)
24. Obtain the two-fold expanded signal y(n) of the input signal x(n)=n for n>0,0otherwis(B5, pp: 594)
25. What are the errors that arise due to quantization of numbers? (B3, pp: 7.1)
26. Describe the forms that are used to represent the numbers in a digital computer? (B3, pp: 7.3-7.7)
27. Compare floating point and fixed point arithmetic? (B3, pp: 7.8)
28. Discuss floating point numbers and blocked floating point numbers? (B3, pp: 7.7-7.9)
29. What are the methods of quantization?Explain them. (B3, pp: 7.9-7.10)
30. Draw the quantizer characteristics for probability density function? (B3, pp: 7.15-7.16)
31. Find the steady state variance of the noise in the output due to quantization of input for the first
order filter? (B3, pp: 7.20)
32. Realize the first order transfer function,draw its quantization noise model. Find the steady state
noise power due to product roundoff.? (B3, pp: 7.20)
33. What is zero input limit cycle oscillations,consider first order difference equation and explain
about it? (B3, pp: 7.30)
34. What is signal scaling.Derive its formula? (B3, pp: 7.33-7.35)
35. Discuss finite word length effects in FIR digital filters?(B3, pp:7.41)
36. Explain about the Dead band effect? (B3, pp:7.31)
37. Realize the IIR Digital filter by floating point quantization? (B3, pp:7.37)
38. The input to the system is y(n)=0.999y(n-1)+x(n) is applied to D.C.What is the power produced by
the quantization noise at the output of the filter if the input is quantized to a)8 bits b)16 bits?
(B3, pp:7.37)
39. Explain the characteristics of a limit cycle oscillation with respect to a system described by
the difference equation y(n)=0.95y(n-1)+x(n).Determine the dead band of the filter? (B3, pp:7.55)
40. A digital system is characterized by the difference equation y(n)=0.9y(n-1)+x(n) with x(n)=0,
and initial condition y(-1)=12.Determine the dead band of the system? (B5, pp:571)