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IMPORTANT QUESTIONS
UNIT – I INTRODUCTION
PART – A
1. What is signal and Signal processing?
2. List the advantages of Digital Signal processing.
3. Mention few applications of Digital Signal processing.
4. Classify discrete time signals.
5. What are Energy and Power signals?
6. What do you mean by periodic and Aperiodic signals?
7. When a signal is said to be symmetric and Anti symmetric?
8. What are deterministic and random signals?
9. What are the elementary signals?
10. What are the different types of representation of discrete time signals?
11. Draw the basic block diagram of digital signal processing of analog signals.
12. What are the basic time domain operations of discrete time signal?
13. What is the significance of unit sample response of a system?
14. Classify discrete time systems.
15. Whether the system defined by the impulse response h(n) = 2n u(-n) + 2-n u(n)
is causal ?
Justify your answer.
16. Compute the energy of the signal x(n) = 2-n u(n)
17. Compute the energy of the signal x(n) = (0.5)n u(n)
18. Define convolution.
19. List out the properties of convolution.
20. What do you mean by BIBO stable?
21. What is linear time invariant system?
22. Compute the convolution of x(n) = {1,2,1,-1} and h(n) = {1,2,1,-1} using
↑
tabulation method.
23. Check whether the system defined by h (n) = [5 (1/2)n + 4 (1/3)n ] u(n) is stable.
24. Differentiate between analog, discrete, quantized and digital signals.
25. Differentiate between analog
26. and digital signals.
27. Differentiate between one dimensional and two dimensional signal with an
example for
each.
28. Name any four elementary time domain operations for discrete time signals.
29. For the signal f (t) = 5 cos (5000πt) + sin2 (3000πt), determine the minimum
sampling rate
for recovery without aliasing.
Ω1 = 5000π = 2πF1 Ω2 = 3000π = 2πF2
F1 = 2.5 kHz F2=1.5 kHz
Fmax = 2.5 kHz
According to sampling theorem Fs ≥ 2 Fmax
So, Fs = 5 kHz
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30. For the signal f (t) = cos2 (4000πt) + 2 sin (6000πt), determine the minimum
sampling rate
for recovery without aliasing.
Ω1 = 4000π = 2πF1 Ω2 = 6000π = 2πF2
F1 = 2 kHz F2 =3 kHz
Fmax = 3 kHz
According to sampling theorem Fs ≥ 2 Fmax
So, Fs = 6 kHz
31. What is sampling?
32. State sampling theorem and what is Nyquist frequency?
Sampling theorem - Fs ≥ 2 Fmax ,Nyquist frequency or Nyquist rate FN = 2 Fmax
33. What is known as aliasing?
34. Define the criteria to perform sampling process without aliasing.
35. Differentiate between anti aliasing and anti imaging filters.
36. What are the effects of aliasing?
37. What is anti aliasing filter? What is the need for it?
38. Draw the basic block diagram of a digital processing of an analog signal.
39. Draw the basic structure of linear constant difference equation.
40. What is sample and Hold circuit?
41. If a minimum signal to noise ratio (SQR) of 33 dB is desired, how many bits per
code word
are required in a linearly quantized system?
SQR =1.76 +6.02b, SQR given is 33 Db
1.76+6.02b = 33 dB
6.02 b = 31.24
b = 5.18 = 6 bits
42. Determine the number of bits required in computing the DFT of a 1024 – point
sequence
with an SQR of 30 dB
The size of the sequence is N = 1024 = 210
SQR is σX2 / σq2 = 22b / N2 10 log [σX2 / σq2] = 10 log [22b / N2]
N2 = 22010 log [σX2/ σq2] = 10 log [22b / 220]
SNR = 10 log [22b - 20] = 30 dB 3(2b-20) =30 ,b=15 bits is the precision for both
multiplication and addition
43. Determine the system described by the equation y (n) = n x (n) is linear or not.
44. What is the total energy of the discrete time signal x(n) which takes the value of
unity at n
=-1,0,1?
45. Draw the signal x(n) = u(n) – u(n-3)
PART - B
1. For each of the following systems, determine whether the system is static stable,
causal,
linear and time invariant
a. y(n) = e x(n)
b. y(n) = ax(n) +b
c. y(n) = 𝑛0 𝑥(𝑘)
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d. y(n) = 𝑛+1
−∞ 𝑥(𝑘)
e. y(n) = n x2(n)
f. y(n) = x(-n+2)
g. y(n) = nx(n)
h. y(n) = x(n) +C
i. y(n) = x(n) – x(n-1)
j. y(n) = x(-n)
k. y(n) = Δ x(n) where Δ x(n) = [x(n+1) – x(n)]
l. y(n) = g(n) x(n)
m. y(n) = x(n2)
n. y(n) = x2(n)
o. y(n) = cos x(n)
p. y(n) = x(n) cos ω0n
2. Compute the linear convolution of h(n) = {1,2,1} and x(n) ={1,-3,0,2,2}
3. Explain the concept of Energy and Power signals and determine whether the
following are
energy or power signals
a. x(n) = (1/3)n u(n)
b. x(n) = sin (π / 4)n
4. The unit sample response h(n) of a system is represented by
h (n) = n2u(n+1) – 3 u(n) +2n u(n-1) for -5≤ n ≤5. Plot the unit sample response.
5. State and prove sampling theorem. How do you recover continuous signals from
its samples? Discuss the various parameters involved in sampling and
reconstruction.
6. What is the input x(n) that will generate an output sequence y(n) =
{1,5,10,11,8,4,1} for a system with impulse response h(n) = {1,2,1}.
7. Check whether the system defined by h(n) = [5 (1/2)n +4(1/3)n] u(n) is stable?
8. Explain the analog to digital conversion process and reconstruction of analog
signal from digital signal.
9. What are the advantages and disadvantages of digital signal processing compared
with analog signal processing?
10. Classify and explain different types of signals.
11. Explain the various elementary discrete time signals.
12. Explain the different types of mathematical operations that can be performed on
a discrete time signal.
13. Explain the different types of representation of discrete time signals.
14. Determine whether the systems having the following impulse responses are
causal and stable
a. h(n) = 2n u(-n)
b. h(n) = sin nπ / 2
c. h(n) = sin nπ + δ (n)
d. h(n) = e2n u(n-1)
15. For the given discrete time signal
x (n) = { -0.5,0.5, for n = -2, -1
1, n = 0
3, 2, 0.4 n > 0}
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10. Derive DIF – FFT algorithm. Draw its basic butterfly structure and compute the
DFT
x(n) = (-
1)n using radix 2 DIF – FFT algorithm. (16)
11. Perform circular convolution of the sequence using DFT and IDFT technique
x1(n) = {0,1,2,3} x2 (n) = {1,0,0,1} ( 8 )
12. Compute the DFT of the sequence x (n) = 1/3 δ (n) – 1/3 δ (n-1) +1/3 δ (n -2)
(6)
13. From the first principles obtain the signal flow graph for computing 8 – point
DFT using
radix-2 DIT - FFT algorithm. Using the above compute the DFT of sequence x(n) =
2 sin nπ
/ 4 for 0 ≤ n ≤ 7 (16)
14. What is circular convolution? Explain the circular convolution property of DFT
and
compute the circular convolution of the sequence x(n)=(2,1,0,1,0) with
itself (8)
15. Perform circular convolution of the sequence using DFT and IDFT technique
x1(n) = {0,1,2,3} x2 (n) = {1,0,0,1} ( 8 )
16. i) Compute the DFT of the sequence x (n) = (-1)n (4)
ii) What are the differences and similarities between DIT – FFT and DIF – FFT
algorithms? (4)
17. From the first principles obtain the signal flow graph for computing 8 – point
DFT using
radix-2 DIT - FFT algorithm. Using the above compute the DFT of sequence x(n) =
cos nπ / 4
for 0 ≤ n ≤ 7 (16)
18. Compute 4-point DFT of the sequence x (n) = (0, 1, 2, 3) (6)
19. Compute 4-point DFT of the sequence x (n) = (1, 0, 0, 1) (6)
20. Explain the procedure for finding IDFT using FFT algorithm (6)
21. Compute the output using 8 point DIT – FFT algorithm for the sequence
x(n) = {1,2,3,4,5,6,7,8} (16)
22.Determine the 8-point DFT of the sequence x(n) = {0,0,1,1,1,0,0,0}
23. Find the circular convolution of x(n) = 1,2,3,4} and h(n) = {4,3,2,1}
24. Determine the 8 point DFT of the signal x(n) = {1,1,1,1,1,1,0,0}. Sketch its
magnitude and phase.
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UNIT - IV DESIGN OF DIGITAL FILTERS
PART – A
1. An analog filter has a transfer function H(s) = 1 / s+2. Using impulse invariance
method, obtain pole location for the corresponding digital filter with T = 0.1s.
2. What is frequency warping in bilinear transformation?
3. If the impulse response of the symmetric linear phase FIR filter of length 5 is
h (n) = {2,3,0,x,y}, find the values of x and y.
4. What is prewarping? Why is it needed?
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5. Find the digital transfer function H (z) by using impulse invariance method for
the analog
transfer function H(s) = 1 / s+2.
6. What are the different structures of realization of FIR and IIR filters?
7. What are the methods used to transform analog to digital filters?
8. State the condition for linear phase in FIR filters for symmetric and anti
symmetric
response.
9. Draw a causal FIR filter structure for length M= 5.
10. What is bilinear transformation? What are its advantages?
11. Write the equation of Barlett (or) triangular and Hamming window.
12. Write the equation of Rectangular and Hanning window.
13. Write the equation of Blackman and Kaiser window.
14. Write the expression for location of poles of normalized Butter worth filter.
Pk = ± Ωc ej (N+2k +1) π / 2N
Where k = 0, 1, ……. (N-1) and for a normalized filter Ωc = 1 rad / sec
15. Write the expression for location of poles of normalized Chebyshev filter.
16. Draw the magnitude response of 3rd order Chebyshev filter.
17. Draw the magnitude response of 4th order Chebyshev filter.
18. Draw the basic FIR filter structure.
19. Draw the direct form – I structure of IIR filter.
20. Draw the direct form – II structure of IIR filter.
21. Draw the cascade form realization structure of IIR filter.
22. Draw the parallel form realization structure of IIR filter.
23. When cascade form realization structure is preferred in filters?
24. Distinguish between FIR and IIR filters.
25. Compare analog and digital filters.
26. Why FIR filters are always stable?
Because all its poles are located at the origin.
27. State the condition for a digital filter to be causal and stable.
28. What are the desirable characteristics of windows?
29. Give the magnitude function Butterworth filter. What is the effect of varying the
order of N
on magnitude and phase response?
30. List out the properties of Butterworth filter.
31. List out the properties of Chebyshev filter.
32. Give the Chebyshev filters transfer function and draw its magnitude response.
33. Give the equation for the order ‗N‘ and cut off frequency Ωc of Butterworth
filter
34. Why impulse invariance method is not preferred in the design of IIR filter other
than low
pass filters?
35. What are the advantages and disadvantages FIR filters?
36. What are the advantages and disadvantages IIR filters?
37. What is canonic structure?
If the number of delays in the structure is equal to the order of the difference
equation or order of transfer function, then it is called canonic form of realization.
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14. Design a digital Butterworth filter satisfying the following constraints with T= 1
sec,
using Bilinear transformation.
0.707 ≤ H (ω) ≤ 1.0, 0 ≤ ω ≤ π/2
H (ω) ≤ 0.2, 3π/4 ≤ ω ≤ π (16)
15. Design a digital Chebyshev filter satisfying the following constraints with T= 1
sec,
using Bilinear transformation.
0.707 ≤ H (ω) ≤ 1.0, 0 ≤ ω ≤ π/2
H (ω) ≤ 0.2, 3π/4 ≤ ω ≤ π (16)
18. Draw and explain cascade form structure for a 6th order FIR filter. (6)
19. Explain impulse invariance method of digital filter design. (10)
20. Derive an expression between s- domain and z- domain using bilinear
transformation.
Explain frequency warping. (10)
21. Draw the structure for IIR filter in direct form – I and II for the following
transfer Function
H (z) = (2 + 3 z-1) (4+ 2 z-1 +3 z-2) / (1+0.6 z-1) (1+ z-1+0.5 z-2) (10)
22. Design a filter with
Hd(ω) = e-j2ω - π/4 ≤ ω ≤ π/4
= 0 π/4 ≤ ω ≤ π Using a Hamming window with N= 7 (16)
23. Discuss about frequency transformations in detail. (8)
24. Design a LPF with
Hd(ω) = e-j3ω - 3π/4 ≤ ω ≤3π/4
= 0 3π/4 ≤ ω ≤ π Using a Hamming window with N= 7 (16)
25. Using the bilinear transformation and a low pass analog Butterworth prototype,
design a low pass digital filter operating at a rate of 20 KHz and having pass band
extending to a 4 KHz with a maximum pass band attenuation of 0.5 dB and stop
band
starting at 5KHzwith a minimum stop band attenuation of 10 dB. (16)
26. Using the bilinear transformation and a low pass analog Chebyshev type I
prototype,
design a low pass digital filter operating at a rate of 20 KHz and having pass band
extending to a 4 KHz with a maximum pass band attenuation of 0.5 dB and stop
band starting at 5KHzwith a minimum stop band attenuation of 10 dB. (16)
27. Obtain the cascade realization of linear phase FIR filter having system function
H(z) =
( 1+1/2 z-1 + z-2) (2 + ¼ z-1 +2z-2) using minimum number of multipliers.(8)
28. Design an ideal Hilbert transformer having frequency response
H(ejω) = j for -π ≤ ω ≤ 0
= -j for 0≤ ω ≤ π for N=11, using i. rectangular window ii. Blackmann
window
29. Obtain the direct form – I, direct form – II, cascade and parallel form of
realization
for the system y(n) = -0.1 y9n-1) + 0.2 y(n-2) + 3 x(n) + 3.6 x (n-1) + 0.6 x(n-2)
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30. Using Bilinear transformation and a low pass analog Butterworth prototype,
design a low pass digital filter operating at the rate of 20k Hz and having pass band
extending to 4 kHz with maximum pass band attenuation of 10 dB and stop band
starting at 5 kHz with a minimum stop band attenuation of 0.5 dB
31. Using Bilinear transformation and a low pass analog Chebyshev type I
prototype,
design a low pass digital filter operating at the rate of 20k Hz and having pass band
extending to 4 kHz with maximum pass band attenuation of 10 dB and stop band
starting at 5 kHz with a minimum stop band attenuation of 0.5 dB
32. Design a low pass filter using Hamming window for N=7 for the desired
frequency
Response D (ω) = ej3ω for -3π / 4 ≤ ω ≤3π / 4
= 0 for 3π / 4 ≤ ω ≤ π
33. Design an ideal differentiator for N=9 using Hanning and triangular window
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