QUESTION BANK

DIGITAL SIGNAL PROCESSING

PART A (2 MARKS)
1. Define DFT of a discrete time sequence.
2. What are the differences and similarities between DIF and DIT algorithms?
3. Define the properties of convolution.
4. Draw the basic butterfly diagram of radix-2 FFT.
5. State and prove parsevals relation for DFT.
6. What do you mean by the term bit reversal as applied to FFT
7. What are the advantages of FFT algorithm over direct computation of DFT?
8. The first five DFT coefficients of a sequence x(n) are x(0) = 20, x(1) = 5+j2, x(2) = 0,
x(3)=0.2+j0.4, X(4) = 0. Determine the remaining DFT coefficients.
9. Define Complex Conjugate of DFT property.
10. What is FFT?
11. State sampling theorem?
12. What is BIBO Stability? What is necessary and sufficient condition for BIBO stability?
13. How will you perform linear convolution via circular convolution?
14. How many multiplications and additions are required to compute N-point DFT using radix-2
FFT?
15. What is decimation-in-time algorithm?
16. What is decimation-in-frequency algorithm?
17. Derive the necessary and sufficient condition for an LTI system to be BIBO stable.
18. Define DFT pair?
19. Give any two properties of DFT
20. Explain Linearity property of DFT
21. What are the applications of FFT algorithms?
22. What are twiddle factors of the DFT?
23. How many additions and multiplications are needed to compute N-point FFT?
24. Calculate the number of multiplications in 64 point DFT using FFT?
25. Find the values of WN, when N=8 and k=2 and also for k=3.
26. Determine the circular convolution of the sequence x1(n) = {1,2,3,1} and x2(n) = {4,3,2,1}.
27. Find the linear convolution of {1,0,1} and {2,0,2}.
28. What is zero padding ? What are its uses?
29. Distinguish between linear and circular convolution of two sequences.
2 Linear convolution Circular convolution
30. What is meant by sectioned convolution?
31. What are the two methods used for the sectioned convolution?
32. Write briefly about overlap-add method?
33. State the difference between (i) overlap-save method (ii) overlap-add method.
34. What are the steps involved in calculating convolution sum?
35. How to obtain the output sequence of linear convolution through circular convolution?
36. Define circular convolution.
37. What are the differences and similarities between DIF and DIT algorithms?

PART B(16 MARKS)
1. Determine the DFT of the sequence
x (n) =1/4, for 0<=n <=2
0, otherwise (16)
2. Derive the DFT of the sample data sequence x (n) = {1, 1, 2, 2, 3, and 3} and compute the
corresponding amplitude and phase spectrum. (16)
3. Given x(n) = {0,1,2,3,4,5,6,7} find X(k) using DIT FFT algorithm. (16)
4. Given X (k) = {28,-4+j9.656,-4+j4,-4+j1.656,-4,-4-j1.656,-4-j4,-4-j9.656}, find x (n) using inverse
DIT FFT algorithm. (16)
5. Find the inverse DFT of X (k) = {1, 2, 3, and 4} (16)
6. a) Compute 4- point DFT of casual three sample sequence is given by,
x(n) = 1/3, 0n2
= 0, else (10)
b) State and prove shifting property of DFT. (6)
7. Derive and draw the radix -2 DIT algorithms for FFT of 8 points. (16)
8. Compute the DFT for the sequence {1, 2, 0, 0, 0, 2, 1, 1}. Using radix -2 DIF FFT and radix -2
DIT- FFT algorithm. (16)
9. 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 and overlap save method. (16)
10. In an LTI system the input x(n) = {1, 1, 1}and the impulse response h(n) = {-1, - 1 } Determine
the response of LTI system by radix -2 DIT FFT (16)
11. 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 (16).
12. Find 4-point DFT of the following sequences
(a) x(n)={1,-1,0,0}
(b) x(n)={1,1,-2,-2} (AU DEC 06)
(c) x(n)=2n
(d) x(n)=sin(n/2)
13. Find 8-point DFT of the following sequences
3 (a) x(n)={1,1,1,1,0,0,0,0}
(b) x(n)={1,2,1,2}
14. Determine IDFT of the following
(a)X(k)={1,1-j2,-1,1+j2}
(b)X(k)={1,0,1,0}
(c)X(k)={1,-2-j,0,-2+j}
15. Find the circular convolution of the following using matrix method and concentric circle method
(a) x1(n)={1,-1,2,3}; x2(n)={1,1,1};
(b) x1(n)={2,3,-1,2}; x2(n)={-1,2,-1,2};
(c) x1(n)=sin n/2; x2(n)=3n 0n7
16.Calculate the DFT of the sequence x(n)={1,1,-2,-2} Determine the response of the LTI system by
radix2 DIT-FFT? If the impulse response of a LTI system is h(n)=(1,2,3,-1)
17. Determine the impulse response for the cascade of two LTI systems having impulse responses
h1(n)=(1/2)^n* u(n),h2(n)=(1/4)^n*u(n)
18. Determine the circular convolution of the two sequences x1(n)={1, 2, 3, 4} x2(n)={1, 1, 1, 1}
and prove that it is equal to the linear convolution of the same.
19. Find the output sequence y(n)if h(n)={1,1,1,1} and x(n)={1,2,3,1} using circular convolution
(AU APR 04)
20. State and prove the following properties of DFT.
1) Cirular convolution 2) Parsevals relation
2) Find the circular convolution of x1(n)={1,2,3,4} x2(n)={4,3,2,1}
UNIT II
DIGITAL FILTER DESIGN
PART A (2 MARKS)
1. Show that the filter with h (n) = [-1, 0, 1] is a linear phase filter.
2. What are the merits and demerits of FIR filters?
3. In the design of FIR digital filters, how is Kaiser window different from other windows?
4. State the condition for a digital filter to be causal and stable?
5. What is the condition satisfied by linear phase FIR filter?
6. Give any two properties of Butterworth filter and chebyshev filter.
7. What are the desirable and undesirable features of FIR Filters?
8. Define Hanning and Blackman window functions.
9. Write the magnitude function of Butterworth filter. What is the effect of varying order of N on
magnitude and phase response?
10. Mention the necessary and sufficient condition for linear phase characteristics in FIR filter
11. What is linear phase? What is the condition to be satisfied by the impulse response in order to
have a linear phase?
12. List the characteristics of FIR filters designed using window functions.
13. Give the Kaiser Window function.
4 14. What are the different types of filters based on impulse response?
15. What are the different types of filters based on frequency response?
16. What are the advantages and disadvantages of FIR filters?
17. What are the design techniques of designing FIR filters?
18. What is Gibbs phenomenon?
19. What are the desirable characteristics of the window function?
20. List the steps involved in the design of FIR filters using windows.
21. What are the advantages of Kaiser Window?
22. What is the principle of designing FIR filter using frequency sampling method?
23. For what type of filters frequency sampling method is suitable?
24. Draw the direct form realization of FIR system.
25. When cascade form realization is preferred in FIR filters?
26. State the equations used to convert the lattice filter coefficients to direct form FIR Filter
coefficient.
27. Draw the direct form realization of a linear Phase FIR system for N even.
28. Draw the direct form realization of a linear Phase FIR system for N odd
29. Draw the M stage lattice filter.
30. State the equations used to convert the FIR filter coefficients to the lattice filter Coefficient.
31. What are the properties of FIR filter?
32. What do you understand by linear phase response?
33. What are the disadvantages of Fourier series method?
34. What is the need for employing window technique for FIR filter design? OR What is window and
why it is necessary?
35. Define Rectangular and Hamming window functions.
36. Compare FIR and IIR filters.
PART B (16 marks)
1. With a neat sketch explain the design of IIR filter using impulse invariant transformation. (16)
2. Apply impulse invariant transformation to H(S) =2/(S +1) (S + 2)
With T =1sec and find H (Z). (16)
3. For a given specifications of the desired low pass filter is
0.707 |H ()| 1.0, 0 0.2
|H ()| 0.08, 0.4
Design a Butterworth filter using bilinear transformation. (16)
4. Explain the procedural steps the design of low pass digital Butterworth filter and List its
Properties. (16)
5. The normalized transfer function of an analog filter is given by,
Ha (Sn) = 1/Sn 2+1.414Sn+1 with a cutoff frequency of 0.4 , using bilinear Transformation. (16)
6. Design a low pass filter using rectangular window by taking 9 samples of w(n) and with a cutoff
frequency of 1.2 radians/sec. Using frequency sampling method, design a band
5 pass FIR filter with the following specification. Sampling frequency Fs =8000 Hz, Cutoff
frequency fc1 =1000Hz, fc2 =3000Hz. Determine the filter coefficients for N =7. (16)
7. Design an ideal high pass filter with
Hd(e j ) = 1 ; /4 | |
= 0 ; | | /4 Using Hamming window with N =11 (16)
8. Design and implement linear phase FIR filter of length N =15 which has following unit Sample
sequence H (k) = 1; for k = 0, 1, 2, 3 0; for k =4, 5, 6, 7 (16)
9.a).The Analog Transfer function H(s)=2/(S+1)(S+2).Determine H(Z) .Using Impulse Invariant
Transformation .Assume T=1sec. (8)
b).Apply Bilinear Transformation to H(s) =2/(S+2) (S+3) with T=0.1 sec.
10. a) Derive bilinear transformation for an analog filter with system function
H(S)=b/S+a
b) Design a single pole low pass digital IIR filter with-3 Db bandwidth of 0.2 by using bilinear
transformation
11. a) Obtain the direct form I, Direct form II, cascade and parallel realization for the following
Systems
y(n)=-0.1x(n-1)+0.2y(n-2)+3x(n)+3.6x(n-1)+0.6x(n-2)
b) Discuss the limitation of designing an IIR filetr using impulse invariant method
12. Determine H(Z) for a Butterworth filter satisfying the following specifications:
0.8 H(e j) 1, for 0/4
H(e j) 0.2, for /2
Assume T= 0.1 sec. Apply bilinear transformation method
13. Determine digital Butterworth filter satisfying the following specifications:
0.707 H(e j) 1, for 02
H(e j 0.2, for3/4.
Assume T= 1 sec. Apply bilinear transformation method.
14. Design a Butterworth high pass filter satisfying the following specifications.
p =1 dB; s=15 dB
p =0.4; s =0.2
15. Design a Butterworth low pass filter satisfying the following specifications.
p=0.10 Hz; p=0.5 dB
s=0.15 HZ; s=15 dB:F=1Hz.
16. Design (a) a Butterworth and (b) a Chebyshev analog high pass filter that will pass all radian
frequencies greater than 200 rad/sec with no more that 2 dB attuenuation and have a 100 rad/sec.
17. Design a digital filter equivalent to this using impulse invariant method
H(S)=10/S2+7S+10
18. Briefly explain about bilinear transformation of digital filter design.











1. State sampling theorem.
2. Distinguish between power and energy signal with an example.
3. State and prove Parseval's theorem.
4. Compute the DFT of the four point sequence x(n ) = [0,1,2,3].
5. What is meant by warping?
6. What are the limitations of impulse invariance method?
7. List out the conditions for the FIR filter to be linear phase.
8. What is meant by limit cycle oscillations?
9. Let x(n ) = [1,2,3,4,5,6]. Sketch x (n 2) and x (3n ).
10. Give any two image enhancement methods.
11. (a) (i) Suppose a LTI system with input x(n ) and output y(n ) is
characterized by its unit sample response h(n ) (0.8) u(n ) = n . Find
the response y(n ) of such a system to the input signal
x(n ) = u(n ) .
(8)
(ii) A causal system is represented by the following difference
equation
( 1)
2
1
( 1) ( )
4
1
y(n ) + y n = x n + x n
Compute the system function H (z ) and find the unit sample
response of the system in analytical form. (8)
Or
(b) (i) Compute the normalized autocorrelation of the signal
x(n ) = a u(n ),0 < a <1 n . (8)
(ii) Determine the impulse response for the cascade of two LTI
system having impulse responses ( ) (1 / 2) ( ) 1 h n u n = n and
( ) (1 / 4) ( ) 2 h n u n = n . (8)
12. (a) By means of the DFT and IDFT, determine the response at the FIR
filter with the impulse response h(n ) = |1,2,3| and the input sequence
x(n ) = |1,2,2,1|.
Or
(b) Compute the DFT of the following sequence x(n ) using the
decimation in time FFT algorithm x(n ) = |1,1,1,1,1,1,1,1|.
13. (a) (i) Find the H (z ) corresponding to the impulse invariance design
using a sample rate of 1/T samples/sec for an analog filter H (s)
specified as follows :

(6)
(ii) Design a digital low pass filter using the bilinear transform to
satisfy the following characteristics (1) Monotonic stop band
and pass band (2) 3 dB cutoff frequency of 0.5 rad (3)
magnitude down at least 15 dB at 0.75 rad . (10)
Or
(b) Design an IIR filter using impulse invariance technique for the given
17 12
1
( )
2 + +
=
s s
H s a . Assume T = 1 sec. Realize this filter using direct
form I and direct form II. (16)
14. (a) Design and obtain the coefficients of a 15 tap linear phase FIR low
pass filter using Hamming window to meet the given frequency
response (16)
()
(`

((
(}
)
o
o
=



| |
6
0 for
6
1 for | |
( )
w
w
H w d .
Or
(b) (i) Determine the coefficients of a linear phase FIR filter of length
M = 15 which has a symmetric unit sample response and a
frequency response that satisfies the conditions
()
(`

((
(}
)
=
=
=
= |
.
|


0 5,6,7
0.4 4
1 0,1,2,3
15
2
k
k
k
k
Hr

. (8)
(ii) The output of A/D converter is applied to digital filter with the
system function
0.5
0.5
( )

=
z
z
H z . Find the output noise power
from the digital filter when the input signal is quantized to have
8 bits. (8)
15. (a) Derive and explain the frequency domain characteristics of the
Decimator by the factor M and interpolator by the factor L. (16)
Or
(b) With neat diagram explain any two applications of adaptive filter
using LMS algorithm. (16)