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DIGITAL SIGNAL PROCESSING -QUESTION BANK

KONGUNADU COLLEGE OF ENGINEERING AND TECHNOLOGY


DEPARTMENT OF ECE
EC2302 DIGITAL SIGNAL PROCESSING
UNIT-1
PART A
1. What is decimation in time algorithm?
2. What is decimation in frequency algorithm?
3. Draw the basic butterfly diagram for radix 2 DIT-FFT and DIF-FFT.
4. What is FFT?
5. What is the need of FFT?
6. What is meant by bit reversal?
7. List out the properties of DFT.
8. State and prove time shifting property of DFT.
9. State and prove Parsevals theorem.
10. State the time reversal property of DFT
11. How many multiplications and additions are required to compute N-point DFT using radix-2
FFT?
12. How many stages of decimations are required in the case of a 64 point radix 2DIT FFT
algorithm?
13. Find the 4 point DFT of the sequence x (n)={1,1,1,1 } .
14. Find the DFT of x(n)={-2,2}
15. State the advantages of FFT over DFTs.
16. What is Twiddle factor?
17. What do you mean by radix-R FFT algorithm, where R is an integer?
18. Determine the DTFT of a sequence x(n)=an u(n).
19. State complex conjugate DFT property.
20. Differentiate (i) overlap-add method (ii) overlap save method.
21. What are the properties of Twiddle factor (WNK )?
22. Define DFT and IDFT.
23. What is zero padding? What are its uses?
24. Obtain the circular convolution the following sequences x(n)={1, 2, 1 }; h(n)={ 1, -2, 2 }
25. Determine
the
circular
convolution
of the
sequence
x1(n)={1,2,3,1}
and
x2(n)={4,3,2,1}.
26. Find the N-point IDFT of a sequence X(k) ={1 ,0 ,0 ,0}.
27. Find the DFT of the sequence of x(n)= cos (n/4) for 0n 3.
28. Find the IDFT of Y(k)={1,0,1,0}
PART-B
1. Explain the salient features of the 8 point DIT FFT algorithm, Explain the advantages of FFT over
direct computation of DFT.
(8)
2. Explain the salient features of the 8 point DIF FFT algorithm, Explain the advantages of FFT over
direct computation of DFT.
(8)
3. State and prove any four properties of DFT.
(16)
4. Differentiate DFT from DTFT.
(4)
5. Differentiate DIT from DIF algorithm.
(4)
6. With appropriate diagrams describe Overlap add and overlap save method.
(8)
7. What is decimation in frequency algorithm? Write the similarities and differences between DIT
and DIF algorithms.
(8)

KNCET, TRICHY

DIGITAL SIGNAL PROCESSING -QUESTION BANK


8. Determine the N- Point DFT of the following Sequence (i)

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

nn 0
x (n)=

(6)
9. Compute 8 point DFT of the sequences x(n)={0,1,2,3,4,5,6,7} Using radix 2 DIF Algorithm.(8)
10. By means of DFT and IDFT ,Determine the sequence x3(n) corresponding to the circular
convolution of the sequence x1(n)={2,1,2,1}.x2(n)={1,2,3,4}
(8)
11. compute an 8 point DFT of the sequence x(n) (0.5,0.5,0.5,0.5,0,0,0,0)
(12)
12. Determine 8 pt DFT of x (n)=1for -3n3 using DIT-FFT algorithm.
(8)
13. compute an 8 point DFT of the sequence x(n) (1,0,1, -1,1, -1,0,1)
(12)
14. Compute the FFT of the sequence x(n)=n+1 where N=8 using the in place radix 2 decimation in
frequency algorithm.
(8)
15. Find DFT for {1,1,2,0,1,2,0,1} using FFT DIT butterfly algorithm and plot the spectrum. (8)
16. Find 8 Point DFT of x(n)=0.5,0n3 Using DIT FFT
0, 4n7
(8)
17. Use the flow graph of 8 point DIT-FFT to compute the DFT of the sequence
x(n)={1,2,3,4,4,3,2,1}
(8)
18. Use the flow graph of 8 point DIT-FFT to compute the DFT of the sequence
x(n)={1,0,0,0,0,0,0,0}
(8)
19. Compute the eight point DFT of the sequence x(n)={1,0 n7
0, Otherwise
By using the decimation in time and decimation in frequency FFT algorithm.
(16)
20. Find the IDFT of the sequence X(K)={4,1-j2.414,0,1-j0.414,0,1+j0.414,0,1+j2.414}.
(16)
21. Compute the linear convolution of finite duration sequence h(n)={1,2} and x(n)={1,2,-1,2,3,-2,3,-1,1,1,2,-1}by overlap save and overlap add methods.
(16)
22. Compute the linear convolution of finite duration sequence x(n)={1,-1,2,1,2,-1,1,3,1} and
h(n)={1,2,1}by overlap save and overlap add methods.
(16)

Unsolved Problems for Practices


4-Point DFT:
1. Find the 4-point DFT of the Sequence x(n)= {1,1,0,0}
2. Find the 4-point DFT of the Sequence x(n)= {0,1,2,3}
3. Find the 4-point DFT of the Sequence x(n)= {1,0,1,0}
4. Find the 4-point DFT of the Sequence x(n)= {1,0,-1,0}
5. Find the 4-point DFT of the Sequence x(n)= {1,-2,3,4}
6. Find the 4-point DFT of the Sequence x(n)= {1,2,3,4}
7. Find the 4-point DFT of the Sequence x(n)= {1,1,1,1}
8. Find the 4-point DFT of the Sequence x(n)= {1,0,0,0}
9. x(n) = { 1 for 0 n 2
0 for otherwise
10. x(n) = { 1/3 for 0 n 2
0 for otherwise
11. x(n) = (n)
12. x(n) = (n-n0)
13. x(n) = an
14. x(n) = 2n
15. x(n) = 2-n
16. x(n) = (1/4)n
17. x(n) = sin (n/2)
18.
x(n) = e j2k0n/N
4-Point IDFT:
1. Find the 4-point IDFT of the Sequence X(K)= {1,1,0,0}
KNCET, TRICHY

DIGITAL SIGNAL PROCESSING -QUESTION BANK


2. Find the 4-point IDFT of the Sequence X(K)= {1,0,1,0}
3. Find the 4-point IDFT of the Sequence X(K)= {1,1-2j,-1,1+2j}
4. Find the 4-point IDFT of the Sequence X(K)= {1,-2-j,0,-2+j}
8-Point DFT:
1. Find the 8-point DFT of the Sequence x(n)= {1,1,1,1,1,1,0,0}
2. Find the 8-point DFT of the Sequence x(n)= {1,1,1,1,0,0,0,0}
3. Find the 8-point DFT of the Sequence x(n)= {1,0,0,0,0,0,0,0}
4. Find the 8-point DFT of the Sequence x(n)= {2,2,2,2,1,1,1,1}
5. Find the 8-point DFT of the Sequence x(n)= {1,2,1,2}
6. Find the 8-point DFT of the Sequence x(n)= (-1)n
7. Find the 8-point DFT of the Sequence x(n)= cos (n/2).
8-point Butterfly Diagram using radix-2 DIT FFT algorithms:
1. Compute the 8-Point DFT of the sequence x(n) = {1/2,1/2,1/2,1/2,0,0,0,0} by using the in-place
radix-2 DIT FFT algorithm.
2. Compute the 8-Point DFT of the sequence x(n) = {1,0,0,0,0,0,0,0} by using the in-place radix-2
DIT FFT algorithm.
3. Compute the 8-Point DFT of the sequence x(n) = {1,1,1,1,0,0,0,0} by using the in-place radix-2
DIT FFT algorithm.
4. Compute the 8-Point DFT of the sequence x(n) = {1,0,1,0,1,0,1,0} by using the in-place radix-2
DIT FFT algorithm.
5. Compute the 8-Point DFT of the sequence x(n) = {1,2,3,4,4,3,2,1} by using the in-place radix-2
DIT FFT algorithm.
6. Compute the 8-Point DFT of the sequence x(n) = {1,2,1,2,0,0,0,0} by using the in-place radix-2
DIT FFT algorithm.
7. Compute the 8-Point DFT of the sequence x(n) = {2,1,2,1,0,0,0,0} by using the in-place radix-2
DIT FFT algorithm.
8. Compute the 8-Point DFT of the sequence x(n) = { 1 for 0 < n < 7
1 for otherwise
by using DIT algorithm.
9. Compute FFT of the sequence x(n) = {2,2,2,2,1,1,1,1}using DIT signal flow graph method.
10. Compute FFT of the sequence x(n) = {1,-1,-1,-1,1,1,1,-1}using DIT signal flow graph method.
11. Compute FFT of the sequence x(n) = {0,1,2,3,4,5,6,7}using DIT signal flow graph method.
12. Compute FFT of the sequence x(n) = {1,2,2,1,1,2,2,1}using DIT signal flow graph method.
13. Given x(n) = 2n and N=8, find X(k)using DIT FFT algorithm.
14. Find the 8-point DFT of the Sequence x(n)= cos (n/2) using DIT FFT algorithm.
15. Compute the DFT for N=8 using DIT FFT method. x(n)= {1,2,3,2,1,2,3,2}
8-point Butterfly Diagram using radix-2 DIF FFT algorithms:
1. Compute the 8-Point DFT of the sequence x(n) = {1/2,1/2,1/2,1/2,0,0,0,0} by using the in-place
radix-2 DIF FFT algorithm.
2. Compute the 8-Point DFT of the sequence x(n) = {1,0,0,0,0,0,0,0} by using the in-place radix-2
DIF FFT algorithm.
3. Compute the 8-Point DFT of the sequence x(n) = {1,1,1,1,0,0,0,0} by using the in-place radix-2
DIF FFT algorithm.
4. Compute the 8-Point DFT of the sequence x(n) = {1,0,1,0,1,0,1,0} by using the in-place radix-2
DIF FFT algorithm.
5. Compute the 8-Point DFT of the sequence x(n) = {1,2,3,4,4,3,2,1} by using the in-place radix-2
DIF FFT algorithm.
6. Compute the 8-Point DFT of the sequence x(n) = {1,2,1,2,0,0,0,0} by using the in-place radix-2
DIF FFT algorithm.

KNCET, TRICHY

DIGITAL SIGNAL PROCESSING -QUESTION BANK


7. Compute the 8-Point DFT of the sequence x(n) = {2,1,2,1,0,0,0,0} by using the in-place radix-2
DIF FFT algorithm.
8. Compute the 8-Point DFT of the sequence x(n) = { 1 for 0 < n < 7
0 for otherwise by using DIF algorithm.
9. Compute FFT of the sequence x(n) = {2,2,2,2,1,1,1,1}using DIF signal flow graph method.
10. Compute FFT of the sequence x(n) = {1,-1,-1,-1,1,1,1,-1}using DIF signal flow graph method.
11. Compute FFT of the sequence x(n) = {0,1,2,3,4,5,6,7}using DIF signal flow graph method.
12. Compute FFT of the sequence x(n) = {1,2,2,1,1,2,2,1}using DIF signal flow graph method.
13. Given x(n) = 2n and N=8, find X(k)using DIF FFT algorithm.
14. Find the 8-point DFT of the Sequence x(n)= cos (n/2) using DIF FFT algorithm.
15. Compute the DFT for N=8 using DIF FFT method.x(n)= {1,2,3,2,1,2,3,2}
UNIT-2
PART A
1. What is the general form of IIR filter?
2. Compare the Butterworth and Chebyshev Type -1 filter.
3. How will you determine the order N of Chebyshev filter.
4. Mention any two procedures for digitizing the transfer function of an analog filter.
5. What is the relation between digital and analog frequency in impulse invariant transformation?
6. What is Bilinear transformation?
7. What is prewarping? Why it is employed?
8. What are the different types of structures for realization of IIR systems?
9. What are the advantages & disadvantages of bilinear transformation?
10. What are the properties of Chebyshev filter?
11. What is Chebyshev approximation?
12. What are the different types of filters based on the frequency response?
13. Distinguish between FIR and IIR filter.
14. What are the properties of Chebyshev filter?
15. What is the relation between digital and analog frequency in Bilinear transformation?
16. What is frequency warping?
17. State the advantage of direct form structure over direct form structure.
18. Give the bilinear transform equation between S-plane & Z-plane.
19. Distinguish between recursive realization and non-recursive realization.
20. List the Butterworth polynomial for various orders.
21. Why do we go for analog approximation to design a digital filter?
22. Why is the Butterworth response called a maximally flat response?
PART-B
1. Explain in detail Butterworth filter approximation.

(16)

2. Use the Impulse invariance method to design a digital filter from an analog prototype that has a system
function Ha(s)=s+a/((s+a)2 +b2 )
(8)
3. Derive bilinear transformation for an analog filter with system function H(s) =b / (s+a)
(8)

4. Describe the concept of impulse invariance method of designing IIR filters.


5. Explain briefly how cascade and parallel realization of filters are done.

(8)
(8)

6. Explain the method of design of IIR filters using bilinear transform method.
(8)
7. Derive the bilinear transformation and the frequency relationship.
(6)
8. Explain how mapping between S plane and Z plane is done using Bilinear transformation method.(8)

9. Design a low pass Butter worth filter using the Bilinear transform method for satisfying the
following constraints: Pass band :0-400 Hz, Stop band:2.1-4 KHz ,Pass band ripple:2dB Stop
band attenuation:20 Db and sampling frequency :10KHz.
(16)
KNCET, TRICHY

DIGITAL SIGNAL PROCESSING -QUESTION BANK


10. Design a low pass digital filter using bilinear transformation. The filter is to be monotonic in both
stop and pass-bands and has all of the following characteristics(i)an acceptable pass band ripple
of 1 dB,(ii)a pass band edge of 0.3 rad and (iii)stop band attenuation 0f 40 dB or greater beyond
0.6 rad.
(16)
11. Determine the system function H(z) of the chebyshev low pass filter with the specifications
p=1dB ripple in the pass band 00.2
s=15dB ripple in the stop band 0.3 using bilinear transformation(assume T=1 sec). (16)
12. Using bilinear transformation convert to z-domain at T = 0.1sec., H(s) = 2/(s+1)(s+3). (8)
13. Using bilinear transformation convert to z-domain at T = 0.1sec., H(s) = 4/(s+3)(s+4). (8)
14. Design digital filter with H(s) =
using T=1sec.
(8)
15. Convert the analog filter with system function H a(s) = s+0.5/(s+0.5)2+16 into a digital IIR filter
using bilinear transformation .The digital filter should have a resonant frequency of r=/2.
(8)
2
16. Convert the analog filter with system function Ha(s) = s+0.1/(s+0.1) +9 into a digital IIR filter
using bilinear transformation .The digital filter should have a resonant frequency of r=/4.
(8)
17. Design the Butterworth filter using bilinear transformation that satisfies following constraint
0.8 H(ejw) 1, 00.2
H(ejw)0.2, 0.6
(16)
18. Design the Butterworth filter using bilinear transformation that satisfies following constraint
0.9 H(ejw) 1, 0/2
H(ejw)0.2, 3/4
(16)
19. Using the Bilinear transform design a high pass filter, monotonic in Pass band with cut off
frequency of 1000Hz and down 10Db at 350Hz.The sampling frequency is 5000Hz.
(16)
20. Determine H(z) for a Butterworth filter satisfying the following constraints.
0.5 H(ejw) 1, 0/2
H(ejw)0.2, 3/4
With T=1s.Apply impulse invariant transformation.
(16)
21. Determine H(z) for a Chebyshev filter satisfying the following constraints.
0.6H(ejw) 1, 0/2
H(ejw)0.2, 3/2
With T=1s.Apply impulse invariant transformation.
(16)
22. Design a digital Chebyshev filter satisfying the constraints
0.707|H (ej)| 1 for 00.2
|H (ej)| 0.1 for 0.5 with T=1 sec using Bilinear Transformation.
(16)
23. Design a digital butter worth filter satisfying the constraints
jw

0.707 H 1 for 0 w
2

e jw

H 0.20 for 3 4

With T=1 sec using bilinear transformation .realize the same in Direct form II.

(16)

24. Obtain the direct form I ,direct form-II, cascade and parallel form realization for the following
system. y(n) = -0.1y(n-1)+0.2y(n-2)+3x(n)+3.6x(n-1)+0.6x(n-2).
(16)
25. Obtain the direct form I , direct form-II, cascade and parallel form realization for the following
system. y(n) = - 13/12y(n-1) - 9/24y(n-2)- 1/24y(n-3) +x(n)+4x(n-1)+3x(n-2).
(16)
KNCET, TRICHY

DIGITAL SIGNAL PROCESSING -QUESTION BANK


26. Determine the direct form I ,direct form II ,Cascade and parallel structure for the system
Y(n)=-0.1y(n-1)+0.72y(n-2)+0.7x(n)-0.25x(n-2).

(16)

UNIT-3
PART A
1. What is a window and why it is necessary?
2. What is meant by linear phase in FIR filter?
3. Why the FIR filter is called as linear phase filter?
4. What are the design techniques of designing FIR filters?
5. What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
6. Draw the direct form realization of FIR system.
7. What are the properties of FIR filter?
8. What are the possible types of impulse response for linear phase FIR filters?
9. Write the equation specifying Hanning windows.
10. Compare the Rectangular and Hamming window.
11. What is Gibbs phenomenon?
12. What are the different types of filters based on impulse response?
13. What are the advantages and disadvantages of FIR filters?
14. List the steps involved in the design of FIR filters using windows.
15. Why is FIR filter always stable?
16. Write the procedure for FIR filter design by frequency sampling method.
17. Write the equation specifying Blackman windows.
18. Compare the Hamming and Blackman window.
19. Write the features of Blackman window spectrum.
20. Draw the direct form realization of a linear Phase FIR system for N odd
21. Distinguish between FIR filters and IIR filters.
22. Write the magnitude and phase function of FIR filter when impulse response is symmetric and N
is odd.
23. Write the magnitude and phase function of FIR filter when impulse response is anti-symmetric
and N is odd.
24. Write the magnitude and phase function of FIR filter when impulse response is symmetric and N
is even.
25. Write the magnitude and phase function of FIR filter when impulse response is anti-symmetric
and N is even.
PART-B
1. Explain the designing of FIR filters using windows.
(16)
2. Obtain a direct form realization for a linear phase FIR system for N-odd and N-even.
(16)
3. Explain the Type-I and Type-II frequency sampling method of designing an FIR filter. (16)
4. Derive the frequency response of a linear phase FIR filter when impulse responses symmetric
and order N is EVEN and ODD.
(16)
5. Derive the frequency response of a linear phase FIR filter when impulse responses antisymmetric and order N is EVEN and ODD.
(16)
6. Design a FIR LPF using hamming window by taking 9 samples of w(n) and with a cut off
frequency of 1.2 radians/sec.
(16)
7. Design a FIR HPF using hamming window by taking 9 samples of w(n) and with a cut off
frequency of 1.2 radians/sec.
(16)
8. Design a FIR HPF using Rectangular window by taking 8samples of w(n) and with a cut off
frequency of 2 radians/sec.
(16)
9. Design a FIR LPF using Rectangular window by taking 9 samples of w(n) and with a cut off
frequency of 1.2 radians/sec.
(16)
KNCET, TRICHY

DIGITAL SIGNAL PROCESSING -QUESTION BANK


10. Design a BPF to pass frequency in the range of 1 to 2 rad/sec using Hanning window with N=5.
(16)
11. Design a BSF to reject frequency in the range of 1 to 2 rad/sec using rectangular window with
N=7.
(16)
12. What are the advantages and disadvantages with the design of FIR filters using window
function?
(8)
13. Design an ideal low pass filter with N=11 With a frequency response

H d ( e j ) = 1 for /2
/2
6
= 0 for /2
(16).
14. Design a FIR linear phase digital filter approximating the ideal frequency response
H d ( e j ) =e-j5 for /2 /2
= 0 for otherwise
using Hamming window method with N=11.

(16)

15. Design and FIR LPF using frequency sampling method for the following specifications. (16)
Pass band frequency = 400 Hz
Stop band frequency = 800 Hz
Sampling frequency = 2000 Hz
Order of the filter = 11
16. Design a FIR low pass filter having following specifications:

H d ( e j ) = 1 for

6
6
= 0 for otherwise
And given that N=7 using Hanning window and Hamming window.

(16)

17. Design a digital FIR band pass filter with lower cutoff frequency 2000 Hz and upper cut off
frequency 3200 Hz using Hamming window of length N=7,Sampling rate is 10000 Hz.
(16)
18. Design an ideal High pass filter with a frequency response
H d ( e j ) = 1 for /4
= 0 for
/4
Find the values of h(n)=11 using hamming
window, Find H(z) and determine the magnitude response.
(16)
19. Determine the coefficients {h(n)} of a linear phase FIR filter of length M=15 which has a
symmetric unit sample response and a frequency response that satisfies the condition
H[2 k/15]={1,for k=0,1,2,3
0,for k=4,5,6,7
20. Obtain the linear phase realization of the system function
H(Z)=1/2+1/3Z-1+Z-2+1/4 Z-3+Z-4+1/5Z-5+1/2 Z-6
(8)
j
21.The frequency response of a filter is given by H(e )=j;- Design the filter ,using a
rectangular window function. Take N=7.
(8)

22.Design the FIR filter with H d ( e j ) ={ e j 3


4
4

{0
<
4
Using a Hanning window and Hamming window with N=7.
(16)
UNIT-4
KNCET, TRICHY

DIGITAL SIGNAL PROCESSING -QUESTION BANK


PART A
1. List some of the finite word length effects in digital filters.
2. What is truncation?
3. What is meant by fixed point arithmetic? Give example.
4. Define Noise Transfer Function.
5. What is dead band?
6. What is zero input limit cycle?
7. What are the different formats of fixed point representation?
8. What are the two types of quantization employed in digital system?
9. What is quantization noise (or) quantization error?
10. What is meant by limit cycle oscillation?
11. List the advantages of floating point arithmetic.
12. What are limit cycle oscillations?
13. What is product quantization error?
14. Write an account of floating point arithmetic.
15. What is coefficient quantization error? what is its effect?
16. What is overflow oscillations?
17. What is the need for signal scaling?
18. What is meant by quantization of analog signals?
19. What is the possible range of numbers in the fixed point-arithmetic?
20. What is round off noise error?
PART B
1. Discuss in detail the errors resulting from rounding and truncation.
(16)
2. Explain fixed and floating point representation in detail.
(16)
3. Explain the quantization process and errors introduced due to quantization.
(16)
4. Explain the limit cycle oscillations due to product round off and overflow errors.
(8)
5. Explain the various errors that occur in a DSP system.
(16)
6. Derive the expression for quantization noise power.
(6)
7. What is meant by signal scaling? Explain.
(8)
8. Explain coefficient quantization in IIR filter.
(8)
9. Draw the product quantization noise model of second order IIR system.
(8)
10. How to prevent limit cycle oscillation? Explain.
(8)
11. Explain limit cycle oscillations.
(6)
12. With a block diagram, explain the quantization noise model. Derive the expressions for steady
state noise power.
(12)
1
13. Realize the first order transfer function H(z)=
and draw its quantization model. Find
1
1a z
the steady state noise power due to product round off .
(8)
-1
-1
14. Consider a second order IIR filter with H(Z)=1/(1-0.5z )(1-0.45z ).Find the effect on
quantization on pole locations of the given system function in cascade realization. Assume b=3.
(8)
15. For the given transfer function H(Z)=H1(Z)H2(Z)where H1(Z)=1/(1-0.5z-1) &
H2(Z)= 1/(1-0.4z-1).Find the output round off noise power. Calculate the value if b=3. (8)
16. Two first order low pass filter whose system functions are given below are connected in cascade.
Determine the overall output noise power. H1(z) = 1/ (1-0.9z-1) and H2(z) = 1/ (1-0.8z-1) (8)
17. Analyze the limit cycle behavior for the following systems 0.7y(n-1)+x(n) ,
y(n)=0.65 y(n-2)+0.52 y(n-1)+x(n) also determine the dead band of the above systems. (10)
18. Study the limit cycle oscillations of the system y(n)=0.95y(n-1)+x(n) where the product is
quantized by rounding to 4 bits and sign magnitude representations is used. Also determine the
dead band of the system.
(10)
KNCET, TRICHY

DIGITAL SIGNAL PROCESSING -QUESTION BANK


19. The input to the system y(n)=0.999y(n-1)+x(n) is applied to an ADC.What is the output
quantization noise power of the filter if the input is quantized to 8 bits.
(8)
20. An 8 bit ADC feeds a DSP system characterized by the following transfer function H(z)=1/
(z+0.5) estimate the steady state quantization noise power at the output of the system. (8)
UNIT-5
PART A
1. What is multirate signal processing?
2. What are the advantages of multirate signal processing?
3. When do you decimate a signal?
4. What is interpolation?
5. What is decimation?
6. What is meant by aliasing?
7. How can aliasing be avoided?
8. Define up sampling
9. What is an anti-imaging filter?
10. Define sampling Rate conversion.
11. Mention applications of multirate signal processing.
12. If the input to the decimator is x(n)={1,2,-1,4,0,5,3,2}, What is the output?
13. Show that the up sampler and down sampler are time variant systems.
14. What is the need for anti-aliasing filter prior to down sampling?
15. What is meant sub band coding?
16. If the spectrum of a sequence x(n) is X(e (jw)), then what is the spectrum of a signal down sampled
by factor 2?
17. How many types of filter banks are there? What are they?
18. When do you go in for multistage implementation for sampling rate conversion?
19. If x(n)={1,-1,3,4,0,2,5,1,6,9,.},what is y(n)=x(2n),y(n)=x(3n)?
20. If x(n)={1,2,3,7,4,-1,5,.}, Find y(n)=x(n/2),y(n)=x(n/3).
PART B
1. With a neat block diagram explain decimation and interpolation.
(16)
2. Explain the polyphase structure for decimator and interpolator.
(16)
3. Discuss the procedure to implement digital filter bank using multirate signal processing. (16)
4. Explain the efficient transversal structure for decimator and interpolator.
(16)
5. Explain the design of narrow band filter using sampling rate conversion.
(8)
6. Explain the application of sampling rate conversion in sub band coding.
(8)
7. Explain the sampling rate conversion by a rational factor and derive input output relation in both
frequency and time domain.
(10)
8. Explain the multistage implementation of sampling rate conversion.
(6)
9. State the applications of multirate signal processing.
(6)
10. Explain the various sound effects can be generated with help of DSP.
(8)
11. Explain oversampling A/D and D/A conversion with the block diagram.
(10)
12. A multirate system is shown in figure .Find the relation between x(n) and y(n).
(8)
2

x(n)

2
Z-1

Z-1

y(n)
2

KNCET, TRICHY

DIGITAL SIGNAL PROCESSING -QUESTION BANK

KNCET, TRICHY

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