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DSP - QUES TION BANK
SUB.NAME: DIGITAL SIGNAL PROCESSING
SUB.CODE: CS2403 BRANCH : CSE
YEAR : IV SEM : VII

UNIT-I

SIGNAL AND S YSTEMS
PAR T-A
1. Define S ignal.

2. Define a system.

3. What are the advantages of DSP?

4. Give some applications of DSP

5. What are the various methods of representing discrete time signal?

6. Define the impulse and unit step signal.

7. How will you classify the discrete time signals?

8. When a discrete time signal is called periodic?

9. What are even and odd signals?

10. What is discrete time system?

11. What is impulse response and what is its and its significance?

12. Define the transfer function of an LTI system.

13. What are the various methods available to determine the response of LTI system?

14. Write few properties of discrete convolution.

15. List the various methods of classifying discrete s ystem.

16. What is a static and dynamic system? Give examples.

17. Define time invariant system.

18. What is linear and nonlinear system?

19. What is casual and non-casual system?

20. What is importance of causality?

21. What is BIBO stability? What is the condition to be satisfied for stability?

22. List any four properties of Fourier transform.

23. What is relation between Fourier transform and z- transform?

24. What is frequency response of LTI system?

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25. What are the steps involved in digital signal processing? Or draw the general block diagram to
show the schematic representation of a DSP system.
26. Define linear convolution and correlation.

27. Distinguish between linear convolution and circular convolution.

28. Define z-transform pair.

29. Define region of convergence.

30. How the stability is determined for a system in terms of z-transform?

31. State persavels relation.

32. State initial and final value theorem.

33. What are the different methods of evaluating inverse z-transform?

34. What are the standard discrete-time signals?

35. What are the operations involved in convolution?

PAR T-B

1. Determine whether the following system are linear, time- invariant. (16)

i) y(n) = A x(n) +B

ii) y(n) = x (2n)
iii) y(n) =n x
2
(n)
iv) y(n) = a x(n)
2. Check for following systems are linear, causal, time in variant, stable, static. (16)

i) y(n) = x (2n)

ii) y(n) = cos (x(n))

iii) y(n) = x(n) cos (x(n)

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

v) y(n) =x(n) +n x (n+1)

3.a) For each impulse response determine the system is i) stable ii) causal. (8)

i) h(n)= sin ( n / 2)

ii) h(n) = (n) + sin n

iii) h(n) = 2 n u(-n)

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b) F ind the periodicity of the signal x (n) =sin (2n / 3) + cos ( n / 2) (8)

4. Explain in detail about A to D conversion with suitable block diagram and to

reconstruct the signal. (16)

5. (i) State and proof of sampling theorem. (8)
(ii)What are the advantages of DSP over analog signal processing? (8)
6. (i) Explain successive approximation techniques. (8)
(ii)Explain the sample and hold circuit. (8)
7. (i) State and proof the properties of Z transform. (8)
(ii)) F ind the Z transform of (8)
a) x(n) = [(1/2)n (1/4)n] u(n)
b) x(n) = n (-1)n u(n)

c) x(n) (-1)n cos (n/3) u(n)

d) x (n) = () n-5 u(n-2) +8(n-5)

8. (i) F ind the Z transform of the following sequence and ROC and sketch the pole zero

diagram (8)

a) x(n) = an u(n) +b n u(n) + c n u(-n-1) , |a| <|b| <| c|

b) x(n) =n2 an u(n)

(ii) F ind the convolution of using z transform (8)
(a) x1(n) ={ (1/3) n, n>=0
(b) (1/2) - n n<0}
(c) x2(n) = (1/2) n
9. Find the inverse z transform (16)
X(z) = log (1-2z) z < |1/2 |
X(z) = log (1+az-1) |z| > |a|

X(z) =1/1+az-1 where a is a constant

X(z)= z2/(z-1)(z-2)
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X(z) =1/ (1- z-1) (1-z-1)2

X(z)= Z+0.2/(Z+0.5)(Z-1) Z>1 using long division method.
X(z) =1- 11/4 z-1 / 1-1/9 z-2 using residue method.
X(z) =1- 11/4 z-1 / 1-1/9 z-2 using convolution method.

10. A causal LTI system has impulse response h(n) for which Z transform is given by

H(z)= 1+ z -1 / (1-1/2 z -1 ) (1+1/4 z -1 ) (16)

i) What is the ROC of H (z)? Is the system stable?

ii) F ind THE Z transform X(z) of an input x(n) that will produce the output
y(n) = - 1/3(-1/4)n u(n)- 4/3
iii) F ind the impulse response h(n) of the system.

11. (i)The impulse response of LTI system is h(n)=(1,2,1,-1).F ind the response of the system to

the input x(n)=(2,1,0,2) (8)
(ii). Determine the response of the causal s ystem y(n) y(n-1) =x(n) + x(n-1) to inputs
x(n)=u(n) and x(n) =2 n u(n).Test its stability (8)

12. Determine the magnitude and phase response of the given equation

y(n) =x(n)+x(n-2) (16)

13. (i)Determine the frequency response for the system given by

y(n)-y3/4y(n-1)+1/8 y(n-2) = x(n)- x(n-1) (8)
(ii)Determine the pole and zero plot for the system described difference equations
y(n)=x(n)+2x(n-1)-4x(n-2)+x(n-3) (8)

14. Find the output of the system whose input- output is related by the difference equation

y(n) -5/6 y(n-1) +1/6 y(n-2) = x(n) -1/2 x(n-1) for the step input. (16)

15. Find the output of the system whose input- output is related by the difference equation

y(n) -5/6 y(n-1) +1/6 y(n-2) = x(n) -1/2 x(n-1) for the x(n) =4 n u(n). (16)

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

and h(n) =an u(n) for all n (16)

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17. Find the convolution sum of x(n) =1 n = -2,0,1
= 2 n= -1
= 0 elsewhere

and h(n) = (n) (n-1) + ( n-2) - (n-3) (16)

18. (i) perform circular convolution of the two sequences x
1
(n) = {2,1,2,1) and x
2
(n) = {1, 2 ,3,4} (8)
(ii)F ind the linear and circular convolution of the sequences x (n) = {1, 0.5} and h (n) = {0.5, 1} (8)
19. (i) The input x(n) and impulse response h(n) of a LTI system are given by,
x(n) = {-1,1,2,-2} ; h(n) = {0.5,1,2,0.75}
Determine the response of the system a) using linear convolution b) using circular convolution. (8)

(ii) F ind correlation of the following sequences. (8)
Cross correlation x (n) = {1, 1, 2, 3, 4}, y (n) = {1, 1, 2, 1}
Circular correlation x (n) = {1, 2, 3, 4}, h (n) = {1, 2, 2, 1}
Auto correlation y (n) = {1, 2, 3, 4}
UNIT II
FREQENCY TRANSFORMATIONS
PAR T-A

1. Define DF T of a discrete time sequence.

2. Define IDF T.

3. What is the relation between DTF T and DF T?

4. What is the draw back in Fourier transform and how it is overcome?

5. Give any two applications of DFT (or mention the importance of DFT).

6. State the properties of DF T.

7. When an N-point periodic sequence is said to be even or odd sequence?

8. What is relation between Z-transform and DFT?

9. What is zero padding? Why it is needed?

10. Why circular convolution is important in DSP?

11 . How w ill yo u pe r for m linear co nvo lut io n via c irc ular convolution?

12. What is sectioned convolution?

13. What are the two methods used for the sectional convolution?

15. What is overlap-save method?
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16. What are differences between overlap-save and overlap-add methods.

17. What way zero padding is implemented in overlap save method?

18. What is FF T?

19. Why FF T is needed?

21 . How ma ny mult ip licat io ns a nd add it io ns are r eq uired to co mp ute N po int DF T using
22. What is DIT algorithm?

23. What DIF algorithm?

24. What are the applications of FF T algorithm?

25. Why the computations in FF T algorithm is said to be in place?

26. What are the differences and similarities between DIF and DIT algorithms?

27. What is phase factor or twiddle factor?

28. Draw and explain the basic butterfly diagram or flow graph of DIT radix-2 FF T.

29. What is DIT radix-2 FF T?

30. Draw and explain the basic butterfly diagram or flow graph of DIF radix-2 FFT.

31. Compare the DIT and DIF radix-2 FF T.

32. What is direct or slow convolution and fast convolution?

PAR T-B

1. (i) compute 4-point DF T of casual three sample sequence given by
x (n) = 1/3 ; 0n2
= 0 ; else (8)

(ii) Compute the DF T of the sequence x (n) = {0, 1, 2, 3}.sketch the magnitude and phase spectrum. (8)

2.Compute the DF T for the sequence.(1,1,1,1,1,1,0,0) (16)

3. Find the DF T of the sequence x (n)

1 for 0 n2
x (n)= for the total of 8 number of sequences. (16)
0 otherwise
4.Find the DF T of a sequence x(n)=(1,1,0,0) and find IDF T of Y(k) =(1,0,1,0) (16)

5. Perform the linear convolution of the following sequences by (i) overlap add method and

(ii) overlap save method.

x (n)= {1,-1,2,-2,3,-3,4,-4}; h(n) = {1,-1}

Also sketch the output sequence. (16)

6. Find the output y (n) of a filter whose impulse response is h (n) = {1, 1, 1} and input
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signal x(n) = {3, -1, 0, 1, 3, 2, 0, 1, 2, 1} using O verlap add and O verlap save method. (16)

7. By means of DF T and IDF T, 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}. (16)

8. Derive and draw the 8 point FF T-DIT butterfly structure. (16)

9. Derive and draw the 8 point FF T-DIF butterfly structure. (16)

10.Find the 8 point DF T of the given sequence x(n) =(0,1,2,3,4,5,6,7,) using DIF,radix-2,FFT

algorithm. (16)

11. From the first principles obtain the signal flow graph for computing 8 point DF T using
radix-2 DIF-FFT algorithm. An 8 point sequence is given by x(n)={2,2,2,2,1,1,1,1}
compute its 8point DFT of x(n) by radix-2 DIF-FFT. (16)

12.Compute the eight point DFT for the sequence x(n)= {0.5,0.5,0.5,0.5,0,0,0,0} using the inplace

13. Find the 8-point DFT of the sequence

1, 0n7
x (n)= using decimation in-time FF T algorithm. (16)
0, otherwise

14. x (n) = sin (n/2) at N=8. F ind X(k) using DIT FFT algorithm. (16)

15. From the first principles obtain the signal flow graph for computing 8 point DF T using radix-2

DIT - FFT algorithm. Using the above compute the DF T of sequence

x (n) = 2 sin n / 4 for 0 n 7 (16)

16. Given x (n) =2
n
and N=8, find X (k) using DIT FF T algorithm. (16)

17. Compute the FF T for the sequence x (n) = n+1 where N=8 using the inplace radix 2 decimation

In frequency algorithm. (16)

18. Determine the response of LTI system when the input sequence is x (n) = {-1, 1, 2, 1,-1} by

radix 2 DIT FFT. The impulse response of the system is h (n) = {-1, 1,-1, 1}. (16)

19. Find the DF T of a sequence x(n)={1,2,3,4,4,3,2,1} using DIT algorithm. (16)

20. (i) Discuss the properties of DF T. (10)
(ii) Discuss the use of FFT algorithm in linear filtering. (6)
21. An LTI system has 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)

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UNIT- III

IIR FILTER DESIGN
PAR T-A
1. Define an IIR filter.

2. Distinguish between IIR and FIR filters.

3. Compare IIR and FIR filters.

4. Classify the filters based on frequency response.

5. What are the properties that are maintained same in the transformation of analog to digital filter?

6. What are the requirements for an analog filter to be stable and causal?

7. What are the requirements for a digital filter to be stable and causal?

8. Define ripples in a filter.

9. Write a brief note on the design of IIR filter. (Or how a digital IIRfilter is designed?)

10. Mention any two techniques for digitizing the transfer function of an analog filter.

11. Compare the digital and analog filter.

13. Mention the important features of IIR filters.

14. What is impulse invariant transformation?

15. How analog poles are mapped to digital poles in impulse invariant transformation (or in bilinear
transformation)?
16. What is the relation between digital and analog frequency in impulse invariant transformation?

17. What is aliasing?

18. What is aliasing problem in impulse invariant method of designing digital filters? Why it is
absent in bilinear transformation?
19. What is bilinear transformation?

20. What is the relation between digital and analog frequency in bilinear transformation?

21. How analog fr eq ue nc y is mapped to d igita l freq ue nc y in b iline ar t ra ns fo r ma t io n?

22. Wha t is freq ue nc y warp ing?

24. What is prewarping? Why it is employed?

25. Explain the technique of prewarping.

26. Compare impulse invariant and bilinear transformations.

27. What is butterworth approximation?

28. Write the properties of Butterworth filter.

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29. What is C hebyshev approximation?

30. Give the properties of C hebyshev filter.

31. State the structure of IIR filter
32. State the advantage of direct form structure over direct form structure.

33. What do you understand by backward difference?

34. What are the properties of bilinear transformation?

PAR T-B

1. Obtain the cascade and parallel form realizations for the following systems (16)
Y (n) = -0.1(n-1) + 0.2 y (n-2) + 3x (n) +3.6 x (n-1) +0.6 x (n-2)
2. (i) Obtain the Direct form II

y (n) = -0.1y(n-1) + 0.72 y(n-2) + 0.7x(n) -0.252 x(n-2) (8)
(ii) F ind the direct form II
H (z) =8z-2+5z-1+1 / 7z-3+8z-2+1 (8)

3. Obtain the i) Direct forms ii) cascade iii) parallel form realizations for the following systems

y (n) = 3/4(n-1) 1/8 y(n-2) + x(n) +1/3 x(n-1) (16)

4. Find the direct form I, cascade and parallel form for (16)
H(Z) = z -1 -1 / 1 0.5 z-1+0.06 z-2
5. Explain the method of design of IIR filters using bilinear transform method. (16)

6. (i) For the analog transfer function H
a
(s) = 2 / (s+1) (s+3), determine H (z) if (a) T=1 sec

and (b) T=0.1 sec (8)
(ii) ) Convert the analog filter with system transfer function
H
a
(s) = (s+0.1) / (s+0.1)
2
+ 9

Into a digital IIR filer means of the impulse invariant method. (8)

7. (i) apply the bilinear transformation to H
a
(s) = 2 / (s+1) (s+2) With T=1 sec and find H (z). (8)
(ii) Convert the analog filter with system function H
a
(s) = into digital filter using bilinear
Transformation, H
a
(s) = (s+3)/(s+0.3)
2
+ 16 (8)

8. (i)The normalized transfer function of an analog filter is given by

H
a
(s
n
) = 1/ s
n
2
+1.414 s
n
+1.

Convert analog filter to digital filter with cut off frequency of 0.4 using bilinear

transformation. (8)
(ii) H
a
(s) = 1/(s+0.1)
2
+ 9 convert the analog BPF into digital IIR filter using backward
difference for the derivate. (8)

9. (i) Design a single pole low pass digital IIR filter with -3db bandwidth of 0.2 by using

bilinear transformation. (8)
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(ii) Design a C hebyshev filter with a maximum pass band attenuation of 2.5 dB; at

p
= 20 rad/sec a nd the stop band attenuation of 30 dB at
s
10. For the constraints
0.8 |H (e
j
)| 1.0

; 0 0.2

|H (e
j
)| 0.2 ;0.6 with T= 1 sec .
Design a Butterworth digital filter using impulse invariant transformation. (16)

11. The specification of the desired lowpass filter is

0.8 |H (e
j
)| 1.0 ; 0 0.2

|H (e
j
)| 0.2 ; 0.6

Design a C hebyshev digital filter using bilinear transformation. (16)

12. The specification of the desired lowpass filter is

1/2 |H ()| 1.0 ; 0 0.2

|H ()| 0.08 ; 0.4

Design a Butterworth digital filter using impulse invariant transformation. (16)

13. The specification of the desired lowpass filter is

0.9 |H ()| 1.0 ; 0 0.25

|H ()| 0.24 ; 0.5

Design a C hebyshev digital filter using bilinear transformation. (16)

14. Design a digital C hebyshev low pass filter satisfying the following specifications
0.707 |H (e
j
)| 1,

0 0.2

|H (e
j
)| 0.1 0.5 with T= 1 sec
using impulse invariant transformation. (16)

15. Design a digital Butterworth filter satisfying the following specifications

0.9 |H (e
j
)| 1, 0 /2

|H (e
j
)| 0.2, 3/4 with T= 1 sec.

using bilinear transformation. (16)

16. Design a realize a digital filter using bilinear transformation for the following specifications
i) Monotonic pass band and stop band
ii) -3.01 db cut off at 0.5 rad

iii) Magnitude down at least 15 db at = 0.75 rad. (16)

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UNIT IV

FIR FILTER DESIGN
PAR T-A
1. What are the different types of filters based on impulse response?

2. How phase distortion and delay distortion are introduced?

3. What are FIR filters?

4. Write the steps involved in FIR filter design.

6. What are the design techniques of designing FIR filters?

7. What is the necessary and sufficient condition for the linear phase characteristic of a FIR filter?

8. What are the conditions to be satisfied for constant phase delay and constant group delay in linear
phase F IR filters?
9. What is Gibbs phenomenon (or Gibbs oscillation)?

10. Write the steps involved in the design of FIR filters using windows.

11. What are the desirable characteristics of the window function?

1 2 . W ha t is t he p r i n c ip l e o f d e s i g n i n g F I R f i l t e r u s i n g f r e q ue nc y s a mp l i n g
method?
12. Write the procedure for FIR filter design by fr e q ue nc y s a mp l i n g method.

13. What is the draw back in FIR filter design using windows and frequency sampling method? How
it is overcome?
14. Distinguish between FIR filters and IIR filters.

15. Write the characteristic features of rectangular and triangular window.

16. Why triangular window is not a good choice for designing FIR filters?

17. List the characteristics of F IR filters designed using windows.

18. List the features of hanning and hamming window spectrum.

19. List the features of Blackman and Kaiser Window spectrum.

20. What are the advantages of Kaiser Window?

21. Draw the direct form realization of FIR system.

22. Draw the direct form realization of a linear P hase F IR system for N even.

23. Draw the direct form realization of a linear P hase F IR system for N odd.

24. When cascade form realization is preferred in F IR filters?

25. What is transposition theorem & transposed structure?

26. What are the different types of arithmetic in digital systems?

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27. What is meant by fixed point number? Write the types of fixed point arithmetic.

28. What is meant by sign magnitude and floating point representation?

29. What is meant by 1s and 2s complement form?
30. What are the quantization errors due to finite word length registers in digital filters?

31. What is input quantization error?

32. What is product quantization error?

33. What is the different quantization methods employed in digital system?

34. What is truncation?

35. What is rounding?

36. What are the two types of limit cycle behavior of DSP?.

38. What is zero input and over flow limit cycle?

39. What are the methods to prevent overflow (or how overflow limit cycles can be eliminated)?

PAR T-B

1. (i) 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 anti
symmetric cases of FIR filter. (8)
(ii) Explain the need for the use of window seque nce in the design of F IR filter and Describe
the window sequence generally used. (8)

2. Design a lowpass filter using rectangular window by taking 9 samples of w(n) and with a

cutoff frequency of 1.2 radians/sec. (16)

3. Design a high pass filter using hamming window, with a cut off frequency of 1.2 rad/sec and

N=9. (16)

4. Design a bandpass filter to pass frequencies in the range 1 to 2 rad/sec using hanning window,

with N=5. (16)

5. Design a bandstop filter to reject frequencies in the range 1 to 2 rad/sec using rectangular

window, with N=7. (16)

6. Determine the coefficient of a linear phase FIR filter of length N= 15 which has a symmetric

unit sample response and a frequency response that satisfies the conditation (16)

1 k=0, 1, 2, 3

H
r
2k/15 = 0.4 k=4

0 k= 5, 6, 7

7. Design a linear phase lowpass FIR filter with a cut-off frequency of /2 rad/sec.Take N=17. (16)

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8. The desired response of a low pass filter is

e
- 3 j
-3 /4 3/4

H d (e
j
) = 0 3 /4

Determine frequency response h (e
j
) for M=7 using a Hamming window. (16)

9. Design a filter with (16)
Response H
d
(e
j
) = e
-j2
for - / 4 / 4
0 for / 4

using Hanning window for M=11

10. Design an ideal differentiator with frequency response

H
d
e
j
= j -

using rectangular and hamming window with N=7 (16)

11. (i) Determine the direct form of following system (8)
H (z) =1+2z
-1
- 3z
-2
+ 4z
-3
- 5z
-4

(ii) Obtain the cascade form realizations of FIR systems (8)
H (z) = 1+5/2 z
-1
+ 2z
-2
+2 z
-3

12. (i) Explain the characteristics of a limit cycle oscillation with respect to the system described

By the equation y(n) = 0.95 y(n-1) + x(n). Determine the dead band of the filter. (8)
(ii) Explain Gibbs phenomenon (or Gibbs oscillation). (8)
13.(i) The output of A/D converter is applied to digital filter with the system function

H (z) = 0.5 z/z-0.5. F ind the output noise power from the digital filter when the input signal is
quantized to have 8 bits. (8)
(ii) Compare hamming window and Kaiser Window. (8)
14. 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)

H
d
(w) =
1 for w /6

0 for /6 w
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15. Design an ideal Hilbert transformer having frequency response

H (e
j
) = j - 0

-j 0
Using rectangular window and Blackman window for N=11. (16)

16. Design an ideal f ilter with
1 /4
H d (e
j
) = 0 /4

Using rectangular window with N=11. (16)

17. Design a F IR filter whose frequency response (16)
H (e
j
) = 1 /4 3/4
=0 | | 3 /4.

Calculate the value of h (n) for N=11 and hence find H (z).

1. What is multirate signal processing?

2. Define down sampling.

3. What is meant by up sampling?
UNIT- V
APPLICATIONS
PAR T-A
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Page 15

4. What is the need for anti-aliasing filter prior to down sampling?

5. What is the need for anti- imaging filter after up sampling?

6. Define sampling rate conversion.

7. Mention two applications of multirate signal processing.

8. Draw the block diagram of Q uadrature mirror filter.

9. What is decimator?

10. What you mean by sub band coding?

11. What is an anti- image filter?

12. Explain the need for scaling in the digital filter realization.

13. Explain briefly the musical sound processing.

14. What are the basic operations of multirate signal processing?

15. What is meant by image enhancement?

16. Give some examples of image enhancement process.

17. Mention the basic approaches of image enhancement.

18. Mention some applications image sharpening.

19. What is bit-plane slicing?

20. What is a histogram?

21. What are the applications of histogram processing?
22. Mention some applications of image sharpening.

23. What are the various enhancement techniques in image processing?

24. List various voice compression coding techniques.

26. Mention some applications of adaptive filters.

27. What types of algorithms used in adaptive filters?

PAR T-B

1. Explain the concept of decimation by a factor D and interpolation by factor I (16)

2. With help of equation explain sampling rate conversion by a rational factor I/D (16)

3. Explain the following application (16)

i) speech compression
ii) sound processing
4. With neat diagram explain any two applications of adaptive filter using LMS

algorithm. (16)

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5. Explain speech vocoders and subband coding. (16)

6. Explain how image enhancement restoration and coding can be done using signal

processing. (16)

7. Explain the methods of speech analysis and synthesis in detail. (16)

8. Explain in detail the design and various implementations (structure) steps for filter using

sampling Rate conversion system. (16)

9. Describe how multirate dsp concepts are applied to basic music processing. (16)
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