AKSHEYAA COLLEGE OF ENGINNERING

CHENNAI
DEPARTMENT OF B.E(ECE)

IT1252
DIGITAL SIGNAL PROCESSING

QUESTION BANK

Prepared By,
Karthikeyan, Lect / ECE

UNIT I
PART A (2 Marks)
1. Draw the general block diagram of DSP.
2. What are the steps involved in DSP?
3. What are the advantages of DSP?
4. What are the disadvantages of DSP?
5. List the difference between analog signal processing and digital signal processing.
6. List the different types of signal representation.
7. Define the auto correlation and cross correlation.
8. Define any two discrete standard signals.
9. What is meant by Nyquist rate?
10. What is an aliasing? How to overcome this effect?
11. Give the expression and the graphical representation of unit ramp signal.
12. Give the expression for unit step and unit impulse discrete signal.
13. List the operations performed on the signals.
14. Sketch the signal u(n-4) and u(n+2).
15. Sketch the discrete time signal x(n) = 4(n+4) + (n) + 2 (n-1) + (n-2) -5 (n-3).
16. State sampling theorem.
17. What are energy and power signal?
18. Define symmetric and antisymmetric signals.
19. If the signal x(n) is odd, what is y(n)=x2(n)?
20. Find whether the signal x(t)=5sin250t + 6cos300t is periodic. If so, find the fundamental
period.
n
n
sin
.
8

21. Find the period N of the sequence x(n) = cos

2 n
.
7

22. Find the periodicity of x(n) = cos

23. Determine which of the following signals are periodic and compute the fundamental period.
(a) sin

2 t

24. Define a system.

(b) sin20t + sin5t.

25. Determine if the system described by the following equation is casual or non casual:
y(n)=x(n)2.
26. Find whether the signal y = n2x(n) is linear.
27. Find whether the signal y(n)= log[x(n)] is linear and time invariant.
28. How you will find that the given system is time invariant?
29. Determine whether the signal y(n)=x(n)u(n) is time invariant.
30. What is the condition for a system to be stable?
31. What is causal and non-casual system?
32. What is the causality condition for an LTI system?
33. What is zero padding?
34. What is an anti imaging and anti aliasing filter?
35. Determine the z-transform of x(n?)= anu(n).
36. Determin Z transform for x(n) = -nan u(-n-1).
37. Find the Z transform of unit step and unit impulse response.
38. Define ROC.
39. List the properties of ROC.
40. Define the transfer function H(z).
41. Define the time shifting property of z transform.
42. State the convolution property of z transforms.
43. State the differential property of z transforms.
44. How will you perform linear convolution?
45. Define commutative and associative law of convolutions.
46. Distinguish between linear convolution and circular convolution.
47. List the properties of convolution.
48. Compute y(n) =x(n) * y(n) where x(n) = h(n) = {1,2,1,-1}.
49. Find the poles of the system y ( n)

1
1
1
y (n 1) y (n 2) y (n 3) 2 x(n) 3x (n 1) and
4
4
16

determine if the system is stable.

50. Determine the response of the system with y(n) = x(n-1) for the input signal for
x(n) = | n |, for 3 n 3

PART B (16 MARKS)

1. Determine whether the following systems are linear, time-invariant, causal and static
(a) y(n)=Ax(n) +B

(c) y(n)=nx2(n)

(b) y(n)=x(2n)

(d) y(n)= ax(n)

2. Check whether the following systems are linear, time-invariant, causal, and static.
(a) y(n)=x(n+2) (b) y(n) = cos[x(n)]

(c) y(n)=x(n)cos[x(n)]

(d) y(n)=x(-n+2)

3. The unit sample response h(n) of a system is represented by

h(n) = nu(n-1) n2u(n+2) + 3u(n-3) for -5 n 5. Plot the unit sample response. (8)
4. Let x(n) = {1, 2, 3, 2, 1}. Find xe(n) =

x ( n ) x ( n)
x ( n) x ( n)
and xo(n) =
. Obtain
2
2

x(n) in terms of xe(n) and xo(n).

(8)

5. Discuss the merits and demerits of digital signal processing over analog signal
Processing.

(8)

5. State and prove the properties of z-transform.

6. Find the z-transform of the following signals
(a) x(n) = ancos0n

(b) x(n)= 4-nu(n-2)

7. Compute the z-transform of x(n) =

1

3
1

2

(c) x(n)=n2u(n)

(d) x(n)=sin0n

u (n), n 0

(10)

u (n), n 0

8. Find the z-transform of x(n) = anu(n) + b-nu(-n-1).

(6)

9. Find the z-transform of x1(n)={3, 5, 7} and x2(n)={3, 0, 5, 0, 7}. What is the relation
between X1(z) and X2(z).

(6)

10. Find the z-transform of

(i) x(n) = 2n u(n-2)

(ii) x(n) = n2u(n).

11. Find the inverse z-transform of X(z)=

z 0.6
.
( z 3)( z 4)( z 5)

12. Determine the inverse z-transform of X(z) =

z ( z 1)
for |z| > 0.5.
( z 0.5)

(6)
(8)
(8)

13. Find the inverse z-transform of

(a) X(z) =

(16)

1
1
1

1 z 1 1 z 1
2
4

(b) X(z) =

z 3
z 1 z 2 z 4

1
14. Find the inverse z-transform of X(z) =

1 1
1
.
z 1 z 1
2
4

15. Find the input casual signal x(n) where

(a) X(z) =

(8)
(16)

z2
z 1 z 2

(b) X(z) =

z 0.2
z 0.5 z 1

11 1
z
4
16. Find the inverse z- transform of X(z) =
using (i) residue method and
1
1 z 2
9
1

(ii) Convolution method.

(16)

z 1

17. Find the inverse z-transform of X(z) =

3z

18. Find the inverse z-transform of X(z) =

z ( z 2)
( z 0.2)( z o.5)( z 3)

19. Determine IZT of X(z) =

4 z 1

using partial fraction method. (8)

1
.
(1 z )(1 z 1 ) 2
1

(8)
(8)

1 z 1
1 1
1 1 .
20. A causal system has the transfer function H(z) =
1 z 1 z
2
4

(i)

Find impulse response h(n).

(ii)

Find X(z) of an input x(n) that will produce output

1 1
y(n)=

3 4

u ( n)

4
2 n u (n 1) .
3

(16)
21. Find the convolution and correlation of the signal x(n) = {0, 1, -2, 3, -4} and
h(n)={0.5, 1, 2, 1, 0.5}.
22. Find the convolution of the following sequence x(n) = u(n) and h(n) = u(n-3).

(8)
(8)

23. Find the output of an LTI system if the input is x(n) = (n+2) for 0 n 3 and
h(n) = anu(n).

(8)

24. Find the response of the system for the input signal
x(n) = {1,2,2,3} and h(n) = {1,0,3,2}.

(4)

25. Find the response of the system whose input is {1,4,6,2} and impulse response is
{1,2,3,1}.

(4)

26. Find xy and yx for x={1,0,2,3} and y={4,0,1,2}.

(8)

27. Check whether the system y(n) = ay(n-1)+x(n) is linear, time invariant, static, stable and
causal.

(8)

28. Find the convolution of the sequence {5 , 4 , 3 ,2} and {1 , 0 , 3 , 2}.

29. What is the input x(n) that will generate the output sequence y(n) = {1, 5, 10, 11, 8, 4, 1}
for a system with impulse response h(n)={1, 2, 1}?

30. Check whether the system defined by h(n) = 5 2

1

(10)
1
4
3

u (n) is

stable.

(6)

31. Find the convolution of the following sequence x(n) = {1, 2, -1, 1} and
h(n) = {1, 0, 1, 1}.

(4)

32. Find the convolution y(n) of the signals

x(n) = n for -3 n 5

and

h(n) = 1 for 0 n 4.

(8)

33. Find the convolution sum of

x(n) = 1 for n = -2,0,1 and h(n) = (n) - (n-1) + (n-2)- (n-3).
34. Find the output sequence y(n) if h(n) ={1, 1, 1} and x(n) = {1, 2, 3, 1}.
2

35. Find the discrete convolution of the sequence x(n) = e n and h(n) = 3n2.

(8)
(4)
(8)

36. Evaluate the convolution y(n) = x(n) * h(n) of the sequence x(n) = an u(n) and
h(n) = bn-m for m n.

(8)

37. Determine the output response of the casual system y(n) y(n-1) = x(n) + x(n-1) to inputs
(i) x(n) = u(n) and

(ii) x(n) = 2-nu(n). Also test the stability.

(16)

38. The difference equation of the system is given as

y ( n)

3
1
y ( n 1) y (n 2) x( n) x( n 1) .
4
8

a. Find the transfer function of the system.

b. Find the impulse response of the system.
c. Find the output y(n) for the input x(n) = 2nu(n).

(16)

39. Find the output of the system whose input-output is related by the difference equation
y ( n)

5
1
1
( n 1) y (n 2) x(n) x(n 1) for the step input.
6
6
2

(12)

40. Find the output of the system whose input-output is related by the difference equation
y ( n)

5
1
1
( n 1) y (n 2) x(n) x(n 1) for x(n) = 4nu(n).
6
6
2

(16)

41. A casual LTI system is described by the difference equation y(n) = y(n-7) + y(n-2) + x(n-1)
(i) Find the system function of the system.
(ii) Find the unit impulse of the system.
(iii)Find the unit step response of the system.
5
6

42. The system is described by y (n) (n 1)

(16)

1
y ( n 2) x (n) . Determine the impulse
6

response of the system.

(8)

43. Using z-transform, compute the response of the system

y(n) = 0.7y(n-1) 0.12y(n-2) + x(n-1) + x(n-2) to the input x(n) = nu(n).
1

44. The input to a casual LTI system is x(n) u ( n 1)

(12)

u ( n) , the z-transform of the

1 1
z
2
.
1

1 z 1 1 z 1
2

output if the system is Y ( z )

a. Determine H(z).
b. Determine the output response of the system y(n).

(16)

45.Consider the following linear constant coefficient difference equation

y ( n)

3
1
y (n 1) y (n 2) 2 x(n 1). Determine y(n) when x(n) = (n). (10)
4
8

46. Find the frequency and impulse response of the following causal system
y ( n)

1
1
x( n) x( n 1) x( n 2) .
2
2

(6)

47. Consider an LTI system with difference equation

y ( n)

1
1
y (n 1) y (n 2) x(n) 3 x( n 1) 2 x(n 2) .
4
8

a. Find the impulse response.

b. Find y(n) for the input x(n) = 2-nu(n).

(16)

48. Determine the step response of the difference equation

y ( n)

1
1
y (n 1) y (n 2) x(n) 3 x( n 1) 2 x(n 2) .
4
8

(12)

49. Find the system function and the impulse response of the system described by the difference
equation y(n) = x(n) + 2x(n-1) 4x(n-2) + x(n-3).

(8)

50. Determine the transfer function and impulse response of the system
y ( n)

3
1
1
y (n 1) y (n 2) x(n) x( n 1) .
4
8
3

(8)

UNIT II
PART A (2 Marks)
1. Define DTFT pair.
2. Determine the DTFT of a sequence x(n) = anu(n).
3. Give the equation for DFT and IDFT.
4. List the properties of DFT.
5. Define the periodicity property of DFT.
6. Define DFT pairs.
7. State and prove Parsevals relation for DFT.
8. State complex conjugate DFT property.
9. What is the difference between DFT and DTFT?
10. Establish the relation between DFT and z-transform.
11. Assume two finite duration sequences x1(n) and x2(n) are linearly combined. Let
x3(n) = ax1(n) + bx2(n). What is the DFT of x3(n)?
12. Find the DFT of x(n) = (-1)n for 0 n 3.
13. Find the DFT of the sequence x(n)= (n).
14. Find the DFT of the sequence x(n) = u(n).
15. Determine the DFT of a sequence x(n) = anu(n).
16. Compute DFT of x(n) = (n-n0).
n

1
17. Calculate the DFT of the sequence x(n) = u(n).
4
k
18. Find the values of WN when N=8 and k=2 and also for k=3.
12
7
19. Find the value of W16 and W8 .

20. Find the DFT of the sequence x(n) = {1,1,0,1}.

21. Find the DFT of x(n) = 1, for N= 4.
22. Find the IDFT of Y(k) = {1,0,1,0}.
23. Find the DFT of x(n) = {1,-1,1,-1}.
24.

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.

25. What is FFT?

26. Why FFT is needed?

27. What is meant by radix-2 FFT algorithm?
28. What is bit reversal in FFT?
29. What are the two methods used in FFT?
30. Compare DIT and DIF.
31. Draw the basic butterfly diagram for radix-2 DIT-FFT and DIF-FFT.
32. What do you mean by in-place computation in FFT?
33. Draw radix-4 FFT-DIT butterfly diagram.
34. Calculate the number of multiplications needed in the calculation of DFT and FFT with
64-point sequence.
35. How many multiplication and additions are required to compute N point DFT using radix 2
FFT?
36. Calculate the number of multiplications needed in the calculation of 512 point radix 2 FFT
when compared to Direct DFT.
37. How many multiplication and additions are required to compute 1024 point DFT using radix 2
FFT and using Direct DFT?
38. Calculate the number of multiplications needed in the calculation of DFT and FFT with
32-point sequence.
39. What are the advantages of FFT over DFT?
40. What is zero padding? What are its uses?
41. Define circular convolution.
42. Distinguish between linear convolution and circular convolution.
43. Determine the circular convolution of the sequence x1(n)={1,2,3,4} and x2(n) = {4,3,2,1}.
44. Find the circular convolution of two finite duration sequences x1(n) = { 1,-1,-2,3},
x2(n) = {1,2,3}.
45. Obtain the circular convolution of the following sequence x(n) = {1,2,1}; h(n) = {1,-2,1}.

PART B (16 MARKS)

1. Perform the circular convolution of the following sequences:

(8)

x1(n) = {2,1, 2,1} and x2(n) ={1,2,3, 4}.

2. Using DFT and IDFT find the circular convolution of x1(n) = {1, 1, 2, 2} and
x2(n) = {1, 2, 3, 4}.

(8)

3. Discuss the properties of DFT.

(10)

4. Compute the DFT of x(n) = e-0.5n for 0 n 5.

(8)

4. Discuss the use of FFT algorithm in linear filtering.

(6)

n
.
2

(16)

6. Find the 4 point DFT of the sequence x(n) = cos

n
.
4

(8)

7. Calculate the DFT of the sequence x(n) = {1, 1, -2, -2}.

(16)

7. Find the IDFT of X(k) = {1, 2, 3, 4}.

(8)

8. Find DFT for x(n) = {1, 2, 3, 4, 1}.

(8)

9. Find the IDFT of X(k) = { 1, 4, 3, 1}.

(8)

10. Find the DFT of the sequence x(n) = {1, 1, 2, 0, 1, 2, 0, 1}using DIT FFT algorithm.

(16)

11. Derive and draw the radix-2 DIT algorithm for FFT 8 points.

(16)

12. Derive and draw the 8-point FFT- DIF butterfly structure.

(16)

13. Find the DFT of the sequence x(n) = {0.5, 0.5, 0.5, 0.5, 0, 0, 0, 0}using DIT algorithm.

(16)

14. Compute the DFT of the sequence x(n) = {1, 1, 1, 1, 1, 1, 0, 0} using DIT algorithm.

(16)

15. Given x(n) = {1, 2, 3, 4, 4, 3, 2, 1}, find X(k) using DIT FFT algorithm.

(16)

5. Find the 8 point DFT of the sequence x(n) = cos

16. Compute the DFT for the sequence (1, 2, 0, 0, 0, 2, 1, 1} using radix-2 DIF FFT algorithm. (16)
17. Using the DIF FFT signal flow graph find the DFT of the sequence
n
for 0 n 7.
4

x(n) = cos

(16)

18. Two finite duration sequences are given by

n
x(n) = sin
for n = 0, 1, 2, 3.
2
h(n) = 2n for n= 0, 1, 2, 3.
(i) Calculate 4-point DFT X(k)
(ii) Calculate 4-point DFT H(k)
(iii) If Y(k) = X(k)H(k), determine the inverse DFT y(n) of Y(k).

(16)

19. Find the DFT of x(n) = 1 for 0 n 3.

(8)

20. Find X(k) using DIT FFT radix-2 algorithm for the sequence
x(n) = {0, 1, 2, 3, 4, 5, 6, 7}.

(16)

21. Compute the DFT of the sequence x(n) = 2n for N= 8 using DIT radix-2 algorithm.

(16)

22. Given x(n) = {0, 1, 2, 3}, find X(k) using DIT FFT algorithm.

(16)

23. Compute the DFT using DIT algorithm for the sequence x(n) = n+1 for N=8.

(16)

24. Find the DFT of the sequence x(n) = {0.5, 0.5, 0.5, 0.5, 0, 0, 0, 0}using DIFalgorithm.

(16)

25. Compute the DFT of the sequence x(n) = {1, 1, 1, 1, 1, 1, 0, 0} using DIF algorithm.

(16)

26. Given x(n) = {1, 2, 3, 4, 4, 3, 2, 1}, find X(k) using DIF FFT algorithm.

(16)

27. Compute the DFT for the sequence (1, 2, 0, 0, 0, 2, 1, 1} using radix-2 DIT FFT algorithm. (16)
28. Using the DIT FFT signal flow graph find the DFT of the sequence
n
for 0 n 7.
4

x(n) = cos

(16)

29. Find X(k) using DIF FFT radix-2 algorithm for the sequence
x(n) = {0, 1, 2, 3, 4, 5, 6, 7}.

(16)

30. Compute the DFT of the sequence x(n) = 2n for N= 8 using DIF radix-2 algorithm.

(16)

31. Given x(n) = {0, 1, 2, 3}, find X(k) using DIF FFT algorithm.

(16)

32. Compute the DFT using DIF algorithm for the sequence x(n) = n+1 for N=8.

(16)

33. Determine DFT (8 point) for continuous time signal x(t) = sin(2t) where f = 50Hz. (16)
34. For the 8 sample sequence x(n) = {1, 2, 3, 5, 5, 3, 2, 1} find the DFT using DIT.

(16)

35. Find the DFT of the sequence x(n) = {1, -1, -1, -1, 1, 1, 1, -1} using DIT FFT.

(16)

36. Compute the FFT for the sequence x(n) = n2 +1 where N= 8 using DIT algorithm.

(16)

37. Compute the DFT of the sequence x(n) = an where N=8 and a=3.

(16)

38. Compute the 8-point DFT for the sequence x(n) = {0, 1, 2, 3} using DIT algorithm.

(16)

39. Compute the linear and the circular convolution of the two sequence x1(n) = {1, 2, 2, 2} and
x2(n) = {1, 2, 3, 4}.

(16)

40. Draw the butterfly diagram using 8 point DIF-FFT for the following sequence
x(n) = {1, 0, 0, 0, 0, 0, 0, 0}.

(16)

41. For the 8 sample sequence x(n) = {1, 2, 3, 5, 5, 3, 2, 1} find the DFT using DIF.

(16)

42. Find the DFT of the sequence x(n) = {1, -1, -1, -1, 1, 1, 1, -1} using DIF FFT.

(16)

43. Compute the FFT for the sequence x(n) = n2 +1 where N= 8 using DIF algorithm.

(16)

44. Compute the 8-point DFT for the sequence x(n) = {0, 1, 2, 3} using DIT algorithm.

(16)

45. By means of DFT and IDFT, determine the response of an FIR filter with impulse response
h(n) = {1, 2, 3} to the input sequence x(n) = {1, 2, 2, 1}.
1, 0 n 7
.
0, otherwise

46. Find the 8-point DFT of the sequence x(n) =

(16)
(16)

47. Determine the output response y(n) if h(n) = {1, 1, 1, 0} ; x(n) = {1, 2, 3, 1} by using liner
convolution and circular convolution.

(8)

48. Find 8-point DFT of x(n) = { 1, -1, 1, -1, 1, -1, 1, -1}using DIT algorithm.
1, 0 n 7
using DIF.
0, otherwise

(16)

49. Find the 8-point DFT of the sequence x(n) =

(16)

50. Find 8-point DFT of x(n) = { 1, -1, 1, -1, 1, -1, 1, -1}using DIF algorithm.

(16)

UNIT III
1.

PART A (2 Marks)
Give the magnitude function of Butterworth filter.

2. Give the properties of Butterworth filters.

3. What are the properties of Chebyshev filter?
4. Give the equation for the order of N and cutoff frequency c of Butterworth filter.
5. Calculate the transfer function of a Butterworth filter of order 2.
6. Give the equation of transfer function of a Butterworth filter of order 3.
7. Determine the order of the analog filter that has a -2db pass band attenuation at a frequency of
20rad/sec and atleast -10db stop band attenuation at 30rad/sec.
8. Find the order and poles of Butterworth LPF that has 3 dB bandwidth of 500Hz and an
attenuation of 40dB at 1000Hz.
9. How you will find the transfer function H(s) of a Chebyshev filter.
10. Give the equation for the order n, major, minor axis of an ellipse in case of Chebyshev filter.
11. Give the expression for the location of poles of a Chebyshev type I filter.
12. Give the expression for location of poles of normalized Butterworth filter.
13. Distinguish between Butterworth and Chebyshev type I filter.
14. Draw the response curve for Butterworth, Chebyshev filters.
15. What are the properties that are maintained same in the transfer of analog filter into a digital
filter?
16. How can you design digital filters from analog filters?
17. Write the equation for frequency transformation from low pass to band pass filter.
18. Mention any two procedures for digitizing the transfer function of an analog filter.
19. How you will convert a stable analog filter to a stable digital filter?
20. What is meant by impulse invariant method of designing IIR filter?
21. What are the limitations of impulse invariant mapping technique?
22. What is the relationship between analog and digital frequency in impulse invariance method.
23. Find the digital transfer function using impulse invariance method for an analog filter
H(s) =

1
for T = 0.5 sec.
s2

24. By impulse invariant method obtain the digital filter transfer function of analog filter

H(s) =

1
.
s 1

25. Obtain the impulse response of digital filter corresponding to an analog filter with impulse
response ha(t) = 0.5e-2t and with a sampling rate of 1KHz, using impulse invariance method.
26. Find digital filter equivalent for H ( s )

1
.
s 8

27. Find the digital transfer function using impulse invariance method for an analog filter
H(s) =

1
for T = 0.1 sec.
s2

28. What is bilinear transformation?

29. Give the bilinear transform equation between s-plane and z-plane.
30. What is warping effect?
31. Write short note on prewarping effect.
32. State two advantages of bilinear transformation.
33. Distinguish between recursive realization and non-recursive realization.
34. What are the different types of structures for realization of IIR system?
35. Give the direct form I and direct form II structures of second order system realization.
36. How many number of additions, multiplications and memory locations are required to realize a
system H(z) having M zeros and N poles in direct form I and direct form II realization?
37. What is the main advantage of direct form II realization when compared to direct form I
realization?
38. Draw the parallel form structure of IIR filter.
39. Realize y ( n) y (n 1)

1
y (n 2) x(n) in cascade form.
4

40. Draw the direct form I structure for the second order system function
H(z) =

H (e j ) 0.2 for

3
0
.
4

41. Draw the direct form I structure for the system y(n) = 0.5x(n) + 0.9y(n-1).

PART B (16 MARKS)

1. Find H(s) for a 3rd order low pass Butterworth filter.

(6)

2. Derive the equation for calculating the order of the Butterworth filter.

(6)

3. Design a low pass Butterworth filter that has a 3 dB cut off frequency of 1.5 KHz and an
attenuation of 40 dB at 3.0 KHz.

(16)

4. Design a single pole low pass digital IIR filter with -3 db bandwidth of 0.2 by use of
bilinear transformation.

(8)

5. Write the digital frequency transformation formula for converting prototype low pass digital
filter into digital band pass and band stop filters.

(4)

6. Design a second order band reject filter with W1 and W2 as cut off frequency and sampling
interval as T.

(6)

7. Determine the order of Chebyshev filter that meets the following specification
(1) 1 dB ripple in the pass band 0 || 0.3b
(2) atleast 60 dB attenuation in the stop band 0.35 || .

(6)

8. Discuss the limitation of designing an IIR filter using impulse invariant method.

(5)

9. Use the impulse invariance method to design a digital filter from an analog prototype that
has a system function

Ha(s) =

sa
.
(( s a ) 2 b 2 )

10. Describe impulse invariant mapping technique for designing IIR filter.
11.

Design a digital filter with H(s) =

If H(s) =

1
using T = 1 sec.
s 7 s 12
2

(10)
(6)
(8)

1
, find H(z) using impulse invariance method for sampling frequency of
( s 1)( s 2)

5 samples/sec.

(6)

Find H(z) using impulse invariant technique for the analog system function
H(s) =

1
.
( s 0.5)( s 0.5s 2)
2

(5)

12. Using impulse invariant mapping technique, convert the following analog transfer function
into digital. Assume T = 0.1 sec for H(s) =

b
.
sa

(8)

13. Design a digital filter using bilinear transform for H(s) =

1
with cut off frequency
s 7 s 12
2

as 100 rad/sec and sampling time = 1.2 ms.

14. Design a digital filter using H(s) =

(8)

1
with T = 0.2sec.
s 9 s 18

(10)

15. Design a digital filter using bilinear transform method for H(s) =

2
for cutoff
( s 1)( s 2)

frequency of 100 rad/sec, and sampling period T = 1.4ms.

(16)

16. Describe bilinear transformation mapping for designing IIR filter.

17. Derive bilinear transformation for an analog filter with system function H(s) =

(8)
b
.
sa

(8)

18. Design a digital Butterworth filter satisfying the constraints

0.707 H (e j ) 1 for 0
H (e j ) 0.2 for

3
0 .
4

with T = 1 sec using bilinear transformation.

(16)

19. Design a Butterworth filter satisfying the constraints

0.9 H ( ) 1 for 0

H ( ) 0.2 for

3

4
.

with T = 1 sec using bilinear transformation.

(16)

20. Design a digital Chebyshev IIR filter for the following desired frequency response H() by
using bilinear transformation with sampling time period Ts = 1 sec.
1

H ( ) 1, 0
5
2
H ( ) 0.1,

.
2

21. Determine H(z) for Butterworth filter satisfying the constraints

0.8 H (e j ) 1 for 0

(16)

H (e j ) 0.1,

.
2

Assume T=0.1sec. Apply Bilinear transformation method.

(16)

22. Explain how you will develop cascade realizations from direct form realization. (8)
23. Realize the IIR system given by the difference equation
y(n) = -0.1y(n-1) +0.72y(n-2) + 0.7x(n) 0.252x(n-2) in direct form I, direct form II,
cascade form and parallel form.

(16)

24. Realize the following filter using cascade and parallel form with direct form I structure
1 z 1 z 2 5 z 3
.
(1 z 1 )(1 2 z 1 4 z 2 )

(10)

25. Develop a cascade and parallel realization of the system described by the difference equation
3
3
1
y ( n) y (n 1)
y (n 2)
y (n 3) x( n) 3 x(n 1) 2 x(n 2) .
8
32
64

(10)

26. Obtain direct form II, cascade and parallel realization of the discrete-time IIR system having
systems function
H(z) =

(1 z 1 )3
.
(1 0.5 z 1 )(1 0.125 z 1 )

(12)

27. Obtain the direct form I, direct form II, cascade and parallel realization for the following
systems:

y(n) = -0.1y(n-1) + 0.2y(n-2) + 3x(n) + 3.6x(n-1) + 0.6x(n-2).

(11)

28. Realize the given transfer function using direct form I format
8 z 2 5 z 1 1
7 z 3 8 z 2 1

(8)

29. Realize the given transfer function using direct form II and parallel structure.

(10)

H(z) =

H(z) =

4 z 2 11z 2
( z 1)( z 3)

30. Obtain the direct form II, cascade form structures for the system
8 z 3 4 z 2 11z 2
H(z) =
.
( z 0.25)( z 2 z 0.5)

(11)

31. Determine the cascade realization for the system described by the system function

H(z) =

1 1
3 1

1
10 1 z 1 z 1 2 z
2
2

3
1

1 1 1
1
1 z 1 1 z 1 1
j z 1

4
8
2 2
2

(5)

32. Consider the system described by y ( n)

1
j
2

z 1

3
1
1
y (n 1) y (n 2) x(n) x( n 1) . Determine
4
8
3

and draw all possible realization structures.

(12)

UNIT IV
PART A (2 Marks)
1. Distinguish between FIR and IIR filters.
2. Define FIR filter.
3. What are the advantages of FIR filter?
4. What are the design techniques used for designing FIR filters?
5. Show that the filter with h(n) = {-1,0,1} is linear phase.
6. What is a linear phase response?
7. What is window? Why it is necessary?
8. Distinguish between recursive ad non recursive realization.
9. What are the disadvantages of FIR filters?
10. What is the condition for linear phase of a digital filter?
11. What are the condition for the constant phase delay of a linear phase FIR filter?
12. Define group delay and phase delay.
13. What are the conditions of a linear phase FIR filter to be symmetric and antisymmetric.
14. Write the magnitude and phase function of FIR filter when impulse response is symmetric
and N is even.
15. Write the magnitude and phase function of FIR filter when impulse response is symmetric
and N is odd.
16. Write the magnitude and phase function of FIR filter when impulse response is
antisymmetric and N is even.
17. Write the magnitude and phase function of FIR filter when impulse response is
antisymmetric and N is odd.
18. List the important characteristics of physically realizable filters
19. What are Gibbs oscillations?
20. Write the procedure for designing FIR filter using windows.
21. Write the equation for Hamming window.
22. What are the desirable features of a window function?
23. Name the different types of windowing functions.
24. In the design of FIR filter, how Kaiser window differs from other windowing techniques?
25. Give the equation for Kaiser window.

26. Define a rectangular window.

27. What are the advantages of Kaiser window?
28. What do you mean by frequency sampling method?
29. Give the relationship between z-transform and DFT?
30. Differentiate Type I and Type II design of frequency sampling method of FIR filter.
31. Give the equation for an impulse response of Type I design.
32. Give the equation for an impulse response of Type II design.
33. Draw the direct realization of FIR system.
34. Give the different methods of realization of FIR system.

PART B (16 MARKS)

1. Write short note on designing FIR filter using windowing techniques.

(8)

2. Compare FIR and IIR filters.

(6)

3. Prove that an FIR filter has linear phase if the unit sample response satisfies the condition
h(n) = h(M-1-n), n = 0,1,M-1. Also discuss symmetric and antisymmetric cases of FIR
filter.

(10)

4. List the three well-known methods of design techniques for FIR filters and explain
any one.

(16)

5. Explain the need for the use of window sequence in the design of FIR filter. Describe the
window sequences generally used and compare their properties.

(6)

6. Explain the steps involved in the design of an FIR linear phase filter using window
method.

(16)

7. Design an ideal low pass filter for a cut off frequency c = 0.4 rad using (8-tap) window
design method.
.

(8)

8. Design a linear phase FIR digital filter

for the given specifications using Hamming

window of length M = 7.

j 3
, for
e
6
Hd() =

0, for
6

j 3
, for
e
4
4
9. Design a filter with Hd(ej) =

0, for

(16)

with N = 7.

using a Hanning window

(16)

3
3
j 3
e
,
for

4
4
10. The desired response of a low pass filter is Hd(ej) =
. Determine
3

0, for

H(ej) for M = 7 using Hamming window.

(16)

1, 4
11. Design an ideal high pass filter with Hd(ej) =

0,

using Hamming window

with N = 11.

(16)

1, 0 4
12. Design an ideal low pass filter with a frequency response Hd(ej) =
. Find
0,

the value of h(n) for N= 11 using (i) Hamming window , (ii) Hanning window.
13. Explain in detail triangular window method of FIR filter design.

(16)
(16)

14. Explain the Type I and Type II design of FIR filter using frequency sampling
technique.

(8)

15. Illustrate the steps involved in the design of linear phase FIR filter by the frequency
sampling method.

(16)

16. A low pass filter has the desired response as given below :

j 3
,0
e
2
Hd(ej) =

0,

Determine the filter coefficients h(n) for M = 7 using frequency sampling technique.

(8)

17. Design and implement linear phase FIR filter of length N = 15 which has following unit
sample sequence
1, fork 0,1, 2,3

0, fork 4,5, 6, 7

H(k) =

(16)

18. Determine the coefficients of a linear phase FIR filter of length M = 15 which has
symmetric unit sample response and a frequency response that satisfies the conditions
1 fork 0,1, 2,3

2 k
Hr
0, fork 4
15
0 fork 5, 6, 7

19. Using frequency sampling method, design a band pass FIR filter with the following
specification. Sampling frequency Fs = 8000Hz, cut off frequency fc1 = 1000 Hz,

(16)

fc2 = 3000 Hz. Determine the filter coefficients for N = 7.

(6)

20. Design an ideal lowpass filter with a frequency response

Hd(ej) = 1
=0

for
for

2
2

Find the values of h(n) for N = 11. Find H(z).

(16)

UNIT V
PART A (2 MARKS)
1. What do you mean by limit cycle oscillations in digital filter?
2. What are the types of limit cycle oscillation?
3. How can overflow limit cycles be eliminated?
4. What is dead band?
5. What is zero input limit cycle oscillation?
6. Explain briefly the need for scaling in the digital filter realization.
7. Explain briefly the musical sound processing.
8. What are the three quantization errors due to finite word length registers in digital filters?
9. Define quantization step size.
10. What is the steady state noise power at the output of an LTI system due to the quantization at
the input to L bits?
11. What is rounding?
12. Define truncation error for an sign magnitude representation and for 2s complement
representation.
13. Express the fraction 7/8 and -7/8 in sign magnitude, 2s complement and 1s complement .
14. What is truncation?
15. Draw a sample/hold circuit and explain its operation.
16. Compare fixed point and floating point arithmetic.
17. What are the two types of quantization employed in digital system?
18. Represent 15.75 in fixed point and floating point representation.
19. Explain briefly quantization noise.
20. List error due to finite word length in filter design.

PART B (16 MARKS)

1. Two first order low pass filter whose system functions are given below re connected in
cascade. Determine the overall output noise power.
H1(z) =
2.
3.

1
1
.
1 and H2(z) =
1 0.9z
1 0.8z 1

(16)

Describe the quantization errors that occur in rounding and truncation in twos
complement.

(6)

Draw a sample/hold circuit and explain its operation.

(4)

4. What is vocoder? Explain with a block diagram.

(6)

5.

Explain in detail about finite word length effects in the filter design.

(16)

6.

Explain in detail about analog to digital conversion with suitable block diagram and
reconstruct the analog signal.

7.

(16)

The output of a A/D converter is applied to a digital filter with the system function
H(z) =

0.5 z
. Find the output noise power for the digital filter when the input signal is
z 0.5

quantized to 8 Hertz.

(16)

8. A digital system is characterized by the difference equation y(n) = 0.95y(n-1) + x(n).

Determine the dead band of the system when x(n) =0 and y(-1) = 13.

(8)

9. Determine the variance of the round off noise at the output of the two cascade realizations of
filter with system function. H(z) = H1(z)H2(z) where H1(z) =
H2(z) =

1
and
1 0.9z 1

1
.
1 0.8z 1

(10)

10. Describe briefly about limit cycle oscillations in recursive systems.

11. Define rounding. Discuss its effects on all types of number representation.
12. Write short notes on 1s complement and 2s complement form with example.

(6)
(8)
(8)

13. Explain in detail about the round-off effects in digital filters.

(6)

14. Explain briefly effects of coefficient quantization in filter design.

(6)

15. Explain briefly effects of product round off error in filter design.

(6)

16. Explain the different representation of fixed and floating point representation.

(8)

17. Discuss about quantization noise and derive the equation for finding quantization noise
power.

(8)
m

2
18. Explain product quantization error and prove err 2 oi .

(16)

i 1

19. Find the output round off noise power for the following transfer function H(z) = H1(z)H2(z)
where

H1(z) =

1
1
and a1 = 0.5 and a2 = 0.6.
1 and H2(z) =
1 a1 z
1 a2 z 1

(16)

20. Compute the steady state noise power in the output due to input quantization for a first order
discrete-time system having difference equation y(n) = ay(n-1) + x(n).

(10)

21. What is meant by truncation? Explain the error that arises due to truncation in floating point
number.

(6)