DSP QB
DSP QB
CAMPUS
QUESTION BANK
and h(n) = .
10. Explain the need for Discrete Fourier Transform (DFT). Define
DFT for a sequence x(n).
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11. Define Z Transform for a discrete time signal x(n). Explain the
significance of ROC in Z Transform.
13. Show that DFT and IDFT form a consistent Discrete Fourier
Transform pairs.
14. Let X(k) denote the N point DFT of an N point sequence x(n).If
the DFT X(k) is computed to obtain a sequence x1 (n) , determine x1 (n) in
terms of x(n).
20. Calculate the 8 point DFT of the sequence x(n) = {1, 1, 1, 1}.
21. Compute the DFT of the following standard signals: (a) x(n) =
δ(n); (b) x(n) = a n for 0 n N-1 ;
22. State and prove the following properties of DFT: (a) Periodicity;
(b) Linearity; (c) Time shifting property; (d) Time reversal of a
sequence; (e) Complex conjugate property; (f) Multiplication of 2
DFT’s – Circular Convolution property; (g) Circular Correlation
property; (h) Multiplication of two sequences; (i) Parseval’s Theorem;
(j) Circular Time Shift property; (k) Circular Frequency Shift property;
25. Give the steps to get the result of linear convolution from the
method of circular convolution.
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27. Convolve the following sequences using (a) overlap – add
method; (b) overlap – save method; for x(n) = {1, -1, 2, 1, 2, -1, 1,
3, 1} and h(n) = {1, 2, 1}.
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,0n2
28. Determine the DFT of the sequence h(n)= 3 .
0, otherwise
29. Compute the inverse DFT for the sequence X(k) = e-k for k = 16.
Module 2
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12. Discuss the computational efficiency of radix 2 FFT algorithm.
13. Explain how you would use FFT algorithm to compute IDFT.
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3. Given | Ha ( j)|2 , determine the analog filter system function
1 646
Ha (s) .
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8. Using analog frequency transformations design a second order
bandpass Butterworth filter with passband of 200 Hz to 300 Hz.
15. Use the backward difference for the derivative to convert the
analog low pass filter with the following system function, using impulse
invariant transformation H(s) = 1/(s+2).
16. Transform the analog filter with the transfer function shown
below into a digital filter, using backward difference for the derivative:
H(s) = 1/(s+2)(s+3).
17. Convert the analog filter to a digital filter whose system function
36
is H(s) = . The digital filter should have a resonant
(s + 0.1)2 + 36
frequency of ωr = 0.2 π . Use bilinear transformation.
18. Convert the analog filter to a digital filter whose system function
1
is H(s) = using bilinear transformation.
(s + 2)2 (s +1)
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19. Design and realize a digital LPF using bilinear transformation to
satisfy the following requirements: (i) monotonic passband and
stopband; (ii) – 3 dB cut off frequency at 0.6π radians; (iii) magnitude
down at 16 dB at 0.75π radians
Module 4
3. Realize a linear phase FIR filter with the following impulse response.
Give necessary equations: h(n) = δ(n) + ½ δ(n-1) – ¼ δ(n-2) + δ(n-
4) + ½ δ(n-3).
4z 2
7. H(z) = 3 + . (i) Does this H(z) represent a FIR or IIR
z - 0.5 z - 0.25
filter? Why?; (ii) Give a difference equation realization of this system
using direct form I; (iii) Draw the block diagram for the direct form II
canonic realization, and give the governing equations for
implementation.
8. Explain the method for designing FIR filters using the window
technique.
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9. Explain Gibb’s phenomenon with respect to window technique for
designing FIR filters.
10. Name the different types of window functions? How are they
defined (mathematically).
13. Design a bandpass filter which approximates the ideal filter with
cut off frequencies at 0.2 rad/sec and 0.3 rad/sec. The filter order is M
= 7. Use the hanning window function.