Professional Documents
Culture Documents
DSP Question Bank
DSP Question Bank
(Autonomous)
Elayampalayam – 637 205, Tiruchengode, Namakkal Dt., Tamil
Nadu.
[2013-2017] Batch / III IT - Sixth Semester
U13EC635 – Digital Signal Processing
QUESTION BANK
UNIT –1
19. In linear system for any bounded input the output will be
a) Bounded b) Unbounded
c) Zero d) Unity
20. Signal flow graph of a system represents
a) Causal system b) Static system
c) Dynamic system d) Feedback system
21. Find the z-transform of an u(n)
a) z / z+a b) z / za
c) z / z-a d) z / a
22. Calculate the sampling rate of band pass filter whose band width is 2F
a) 2F samples/sec b) 4F samples/sec
c) 6F samples/sec d) 8F samples/sec
23. Find the nyquist interval of the signal whose frequency value is 250 Hz
a) 0.04 b) 0.4
c) 0.004 d) 4
24. The z-transform of impulse response is-------------------of the system.
a) Transfer function b) System function
c) Unit function d) impulse function
25. The z-transform of impulse function gives the------------- of the system
a) y(n - 1) b) y(n + 1)
c) y(n) d) y(n) + y(n + 1)
12.
The system that does not require memory is called-------------------
a) Causal system b) Static system
c) Dynamic system d) Feedback system
13. Find the nyquist rate of the signal whose frequency value is 250hz
a) Unit response b) Analog response
c) Impulse response d) Feedback response
14. Find the z-transform of an u(n)
a) z / z+a b) z / za
c) z / z-a d) z / a
15. A function having frequency f is to be sampled. The sampling time T should be
a) T=1/2f b) T>1/2f
c) T<1/2f d) T≥1/2f
PART C
3.A discrete time system y(n) = y2(n-1) + x(n) a bounded input x(n) = 2 u(n) is
applied to the system check the stability condition of the system.
4. Define convolution integral and derive the equation.
PART-A
Ans:d
5. If x(n) is a real sequence and X(k) is its N-point DFT, then which of the following is true?
a) X(N-k)=X(-k)
b) X(N-k)=X*(k)
c) X(-k)=X*(k)
d) All of the mentioned
6. If x(n) is real and even, then what is the DFT of x(n)?
d) None of the mentioned
ans:b
7. If x(n) is real and odd, then what is the IDFT of the given sequence?
Ans:c
9. What is the circular convolution of the sequences x1(n)={2,1,2,1} and x2(n)={1,2,3,4}?
a) {14,14,16,16}
b) {16,16,14,14}
c) {2,3,6,4}
d) {14,16,14,16}
10. What is the circular convolution of the sequences x1(n)={2,1,2,1} and x2(n)={1,2,3,4}, find
using the DFT and IDFT concepts?
a) {16,16,14,14}
b) {14,16,14,16}
c) {14,14,16,16}
d) None of the mentioned
11. If X(k) is the N-point DFT of a sequence x(n), then circular time shift property is that N-
point DFT of x((n-l))N is X(k)e-j2πkl/N.
a) True
b) False
12. If X(k) is the N-point DFT of a sequence x(n), then what is the DFT of x*(n)?
a) X(N-k)
b) X*(k)
c) X*(N-k)
d) None of the mentioned
12.By means of the DFT and IDFT, determine the response of the FIR filter with impulse
response h(n)={1,2,3} to the input sequence x(n)={1,2,2,1}?
a) {1,4,11,9,8,3}
b) {1,4,9,11,8,3}
c) {1,4,9,11,3,8}
d) {1,4,9,3,8,11}
13.What is the sequence y(n) that results from the use of four point DFTs if the impulse response
is h(n)={1,2,3} and the input sequence x(n)={1,2,2,1}?
a) {9,9,7,11}
b) {1,4,9,11,8,3}
c) {7,9,7,11}
d) {9,7,9,11}
14.Overlap add and Overlap save are the two methods for linear FIR filtering a long sequence on
a block-by-block basis using DFT.
a) True
b) False
15.In Overlap save method of long sequence filtering, what is the length of the input sequence
block?
a) L+M+1
b) L+M
c) L+M-1
d) None of the mentioned
16.In Overlap save method of long sequence filtering, how many zeros are appended to the
impulse response of the FIR filter?
a) L+M
b) L
c) L+1
d) L-1
17.The first M-1 values of the output sequence in every step of Overlap save method of filtering
of long sequence are discarded.
a) True
b) False
18.In Overlap add method, what is the length of the input data block?
a) L-1
b) L
c) L+1
d) None of the mentioned
18. Which of the following is true in case of Overlap add method?
a) M zeros are appended at last of each data block
b) M zeros are appended at first of each data block
c) M-1 zeros are appended at last of each data block
d) M-1 zeros are appended at first of each data block
19. In which of the following methods, the input sequence is considered as shown in the below
diagram?
a) Overlap save method
b) Overlap add method
20.What is the value of x(n)*h(n), 0≤n≤11 for the sequences x(n)={1,2,0,-3,4,2,-1,1,-2,3,2,1,-3}
and h(n)={1,1,1} if we perform using overlap add fast convolution technique?
a) {1,3,3,1,1,3,5,2,2,2,3,6}
b) {1,2,0,-3,4,2,-1,1,-2,3,2,1,-3}
c) {1,2,0,3,4,2,1,1,2,3,2,1,3}
d) {1,3,3,-1,1,3,5,2,-2,2,3,6}
21.What is the value of x(n)*h(n), 0≤n≤11 for the sequences x(n)={1,2,0,-3,4,2,-1,1,-2,3,2,1,-3}
and h(n)={1,1,1} if we perform using overlap save fast convolution technique?
a) {1,3,3,-1,1,3,5,2,-2,2,3,6}
b) {1,2,0,-3,4,2,-1,1,-2,3,2,1,-3}
c) {1,2,0,3,4,2,1,1,2,3,2,1,3}
d) {1,3,3,1,1,3,5,2,2,2,3,6}
22. Which of the following is true regarding the number of computations required to compute an
N-point DFT?
a) N2 complex multiplications and N(N-1) complex additions
b) N2 complex additions and N(N-1) complex multiplications
c) N2 complex multiplications and N(N+1) complex additions
d) N2 complex additions and N(N+1) complex multiplications
23. Which of the following is true regarding the number of computations required to compute
DFT at any one value of ‘k’?
a) 4N-2 real multiplications and 4N real additions
b) 4N real multiplications and 4N-4 real additions
c) 4N-2 real multiplications and 4N+2 real additions
d) 4N real multiplications and 4N-2 real additions
24.What is the real part of the N point DFT XR(k) of a complex valued sequence x(n)?
PART B
1.Divide-and-conquer approach is based on the decomposition of an N-point DFT into
successively smaller DFTs. This basic approach leads to FFT algorithms.
a) True
b) False
2.If the arrangement is of the form in which the first row consists of the first M elements of x(n),
the second row consists of the next M elements of x(n), and so on, then which of the following
mapping represents the above arrangement?
a) n=l+mL
b) n=Ml+m
c) n=ML+l
d) None of the mentioned
3.How many complex multiplications are performed in computing the N-point DFT of a
sequence using divide-and-conquer method if N=LM?
a) N(L+M+2)
b) N(L+M-2)
c) N(L+M-1)
d) N(L+M+1)
4.How many complex additions are performed in computing the N-point DFT of a sequence
using divide-and-conquer method if N=LM?
a) N(L+M+2)
b) N(L+M-2)
c) N(L+M-1)
d) N(L+M+1)
5.Which is the correct order of the following steps to be done in one of the algorithm of divide
and conquer method?
1) Store the signal column wise
2) Compute the M-point DFT of each row
3) Multiply the resulting array by the phase factors WNlq.
4) Compute the L-point DFT of each column.
5) Read the result array row wise.
a) 1-2-4-3-5
b) 1-3-2-4-5
c) 1-2-3-4-5
d) 1-4-3-2-5
6.If we store the signal row wise then the result must be read column wise.
a) True
b) False
7. If we split the N point data sequence into two N/2 point data sequences f1(n) and f2(n)
corresponding to the even numbered and odd numbered samples of x(n), then such an FFT
algorithm is known as decimation-in-time algorithm.
a) True
b) False
8. If we split the N point data sequence into two N/2 point data sequences f1(n) and f2(n)
corresponding to the even numbered and odd numbered samples of x(n) and F1(k) and F2(k) are
the N/2 point DFTs of f1(k) and f2(k) respectively, then what is the N/2 point DFT X(k) of x(n)?
a) F1(k)+F2(k)
b) F1(k)- WNk F2(k)
c) F1(k)+WNkNk F2(k)
d) None of the mentioned
9.How many complex multiplications are required to compute X(k)?
a) N(N+1)
b) N(N-1)/2
c) N2/2
d) N(N+1)/2
10.The total number of complex multiplications required to compute N point DFT by radix-2
FFT is:
a) (N/2)log2N
b) Nlog2N
c) (N/2)logN
d) None of the mentioned
11.The total number of complex additions required to compute N point DFT by radix-2 FFT is:
a) (N/2)log2N
b) Nlog2N
c) (N/2)logN
d) None of the mentioned
12.The following butterfly diagram is used in the computation of:
a) Decimation-in-time FFT
b) Decimation-in-frequency FFT
13. For a decimation-in-time FFT algorithm, which of the following is true?
a) Both input and output are in order
b) Both input and output are shuffled
c) Input is shuffled and output is in order
d) Input is in order and output is shuffled
14. The following butterfly diagram is used in the computation of:
a) Decimation-in-time FFT
b) Decimation-in-frequency FFT
PART-C
2. Obtain the signal flow graph for computing 8 point DFT using radix-2 DIF FFT algorithm.
6. Calculate the DFT of the sequence x(n)={1,1,-2,-2}.determine the response of LTI system by
radix-2 DIT FFT.
PART-A
b. bilinear transformation
c.frequency sampling
13. The unnormalized transfer function of low pass butter worth filter is obtained from normalized
transfer function by replacing sn by ,
a. sn/Ωc
b. snΩc
c. s/Ωc
d. sΩc
23. For chebyshev filter, when Ap and AS are in dB, cut-off frequency is given by
a. Ωc = Ωp/2
b. Ωc = Ωs/2
c. Ωc = Ωp
d. Ωc = Ωs
24. The two popular techniques used to approximate the ideal frequency response are ………and
……..approximation.
a. butterworth,chebyshev
b.aliasing ,bilinear
c.impulse invariant,bilinear
d.none of these
25. The phenomena of high frequency components acquiring the identity of low frequency components is
called……….
a.aliasing
b.frequency warping
c.stability
d.impulse invariant
PART-B
1. In impulse invariant transformation the analog system with transfer function,H(s)=0.3 /(s+0.7) is
transformed to a digital system with transfer function,
a.H(s) = -0.3/ (1-e-0.7Tz-1)
b. H(s) = 0.3/ (1-e-0.7Tz-1)
c. H(s) = 0.7/ (1-e-0.3Tz-1)
d. H(s) = 0.7/ (1-e0.3Tz-1)
2. In bilinear transformation the analog system with transfer function,H(s) = 0.2/(s+0.9) is trnaformed to a
digital system with transfer function,
a. H(s) = 0.2/[(2/T)((1+z-1)/(1-z-1))+0.9]
b. H(s) = 0.2/[(T/2)((1+z-1)/(1-z-1))+0.9]
c. H(s) = 0.2/[(2/T)((1-z-1)/(1+z-1))+0.9]
d. H(s) = 0.2/[(T/2)((1-z-1)/(1+z-1))+0.9]
3. The poles of butterworth transfer function symmetrically lies on a circlr in s-plane with angular
spacing,
a. π/N
b. π/2N
c. 2π/N
d. π/N2
7. In impulse invariant transformation method ,relationship between digital transformation and analog
transformation is given by
a.1/(S-Pi)→ 1/(1-eTPe-jw)
b. 1/(S-Pi)→ 1/(1-eTPe jw)
c. 1/(S+Pi)→ 1/(1-eTPe-jw)
d. 1/(S+Pi)→ 1/(1-eTPejw)
8. In bilinear transformation method ,the relationship between digital transformation and analog
transformation is given by
a. S=(1/Ts)(1+Z-1/1-Z-1)
b. S=(2/Ts)(1+Z-1/1-Z-1)
c. S=(1/Ts)(1-Z-1/1+Z-1)
d. S=(2/Ts)(1-Z-1/1+Z-1)
9. Magnitude function of butterworth filter is given by
a. │Ha(Ω)│=1/[1+(Ω/Ωc)N] N=1,2,3,…
b. │Ha(Ω)│2 =1/[1+(Ω/Ωc)2N] N=1,2,3,…
c. │Ha(Ω)│=1/[1+(Ωc/Ω)2N] N=1,2,3,…
d. │Ha(Ω)│=1/[1+(Ωc/Ω)N] N=1,2,3,…
10. Magnitude function of chebyshev filter is given by
a. │Ha(Ω)│=1/[1+ε2CN2(Ωc/Ω)]1/2 N=1,2,3,…
b. │Ha(Ω)│=1/[1+εCN2(Ω/Ωc)]1/2 N=1,2,3,…
c. │Ha(Ω)│=1/[1+ε2CN2(Ω/Ωc)]1/2 N=1,2,3,…
d. │Ha(Ω)│=1/[1+CN2(Ωc/Ω)]1/2 N=1,2,3,…
11. For butterworth filter, cut-off frequency is given by
a. Ωc=Ωp/[(1/δp2)-1]1/2N
b. Ωc=Ωp/[(1/δp2)-1]1/N
c. Ωc=Ωp/[(1/δs2)+1]1/2N
d. Ωc=Ωp/[(1/δp2)+1]1/N
12. For chebyshev filter, filter order N is given by
a. N ≥ {cosh-1[(1/ε)((1/δs2)-1)1/2] / cosh-1(Ωs/Ωp)}
b. N ≥ {cosh-1[(1/ε)((1/δs2)+1)1/2] / cosh-1(Ωs/Ωp)}
c. N ≥ {cosh-1[(1/ε)((1/δp2)-1)1/2] / cosh-1(Ωs/Ωp)}
d. N ≥ {cosh-1[(1/ε)((1/δp2)+1)1/2] / cosh-1(Ωs/Ωp)}
13. For the analog transfer function H(s)=3/[(s+1)(s+2)],determine H(z),using bilinear transformation
method. assume Ts=1 sec
a. H(z) = [3(1+z-1)2/8(3+z-1)]
b. H(z) = [3(1-z-1)2/8(3+z-1)]
c. H(z) = [3(1+z-1)2/8(3-z-1)]
d. H(z) = [3(1+z-1)2/8(-3+z-1)]
14. In butterworth filter design, the error function is selected such that the magnitude is maximally flat in
the passband and monotonically decreasing in the ……….
a. stop band
b. wide band
c. frequency band
d. none of these
15. The normalized transfer function of lowpass filter is transformed to highpass filter with cutoff
frequency Ωc, by the transformation …….
a. sn→Ωc/s
b. sn→Ωc/-s
c. sn→s/Ωc
d. none of these
PART C
1. Design third order butterworth digital filter using impulse invariant technique.assume sampling period
T=1 sec.
2. Using the bilinear transform,design a highpass filter , monotonic in passband with cutoff frequency of
1000Hz and down 10dB at 350 Hz.the sampling frequency is 5000Hz.
3.Obtain the direct form I,direct form II realization for the system y(n)= -0.1y(n-1)+0.2y(n-
2)+3x(n)+3.6x(n-1)+0.6x(n-2).
4. Obtain the cascade,parallel realization for the system y(n)= -0.2y(n-1)+0.3y(n-2)+3x(n)+4.6x(n-
1)+1.6x(n-2).
5. Design a chebyshev lowpass filter with specifications αp=1dB ripple in the pass band
0 ≤ ω ≤0.2π, αs=15dB ripple in the stop band 0.3π ≤ ω ≤ π using bilinear transformation.
6.Design a digital butterworth filter satisfying the constraints
0.707 ≤│H(ejω)│≤1 for 0 ≤ ω ≤ π/2
jω
│H(e )│≤ 0.2 for 3π/4 ≤ ω ≤ π
With T=1 sec using impulse invariance.
7. Design a butterworth filter using the impulse variance method for the following specifications
0.8 ≤│H(ejω)│≤1 for 0 ≤ ω ≤ 0.2π
jω
│H(e )│≤ 0.2 for 0.6π ≤ ω ≤ π
8. Design a chebyshev filter for the following specifications using bilinear transformation method.
0.8 ≤│H(ejω)│≤1 for 0 ≤ ω ≤ 0.2π
│H(ejω)│≤ 0.2 for 0.6π ≤ ω ≤ π
9. Explain lattice structure of IIR system.
10.Design a digital chebyshev filter to meet the constraints
1/√2 ≤│H(ejω)│≤1 for 0 ≤ ω ≤ 0.2π
0 ≤│H(ejω)│≤ 0.1 for 0.5π ≤ ω ≤ π
By using bilinear transformation and assume sampling period T=1 sec
UNIT –IV
PART-A
b.hanning
c.bartlett
d.kaiser
14.The width of the main lope should be --------and it should contain as much of the total energy
as possible.
a.large
b.medium
c.very large
d. small
15.Symmetric impulse response having odd number of samples, N=7 with centre of symmetry α
is equal to,
a. 2
b. 5
c. 3.5
d.3
16. FIR filters are
a.non-recursive type
b.recursive type
c.neither recursive nor non-recursive type
d.none of these
22.In FIR filter design using hamming window and hanning window, we assume
a. pass band and stop band ripple are equal
b. only pass band contains ripple
c. only stop band contains ripple
d. pass band and stop band ripple are not equal
𝑁
−1 𝑁−1
2
c.∑𝑛=0 ℎ(𝑛)cos[𝜔( − 𝑛)]
2
𝑁−3
𝑁−1 2 𝑁−1
d.[ℎ ( ) + ∑𝑛=0 ℎ(𝑛)𝑐𝑜𝑠𝜔( − 𝑛)]
2 2
𝑁
−1 𝑁−1
2
c.∑𝑛=0 ℎ(𝑛)cos[𝜔( − 𝑛)]
2
𝑁−3
𝑁−1 2 𝑁−1
d.[ℎ ( ) + ∑𝑛=0 ℎ(𝑛)𝑐𝑜𝑠𝜔( − 𝑛)]
2 2
PART-B
1.Truncation of infinite series results in
a. undesirable oscillations in the pass band and stop band
b. undesirable oscillations in the stop band
c. undesirable oscillations in the pass band
d. undesirable oscillations in the transition band
3.Impulse response of an ideal high pass FIR filter when n=(N-1)/2 is given by
a. 1-ωc /π
b. 1+ωc /π
c. 1-π/ωc
d.1+π/ ωc
4.The filter coefficient H=-0.673 is represented by sign-magnitude fixed point arithmetic.If the
word length is 6 bits, compute the quantization error due to truncation.
a.0.001126
b. 0.001125
c.0.001115
d. 0.001123
7.Impulse response of an ideal band pass FIR filter when n=(N-1)/2 is given by
1
a.𝛱 [Π-ωc2+ ωc1]
1
b.𝛱 [Π-ωc2-ωc1]
𝟏
c. 𝜫 [ωc2- ωc1]
1
d.𝛱 [ωc2+ωc1-Π]
8.the values of h(n) of linear phase FIR are real and symmetrical when
a. magnitude of DFT from 0 to Π is same as Π to 2Π
b.magnitude of DFT from 0 to -Π is same as Π to -Π
c.magnitude of DFT from 0 to -Π is same as Π to -2Π
d. magnitude of DFT from 0 to Π is same as Π to -2Π
9.the characteristics of ideal linear phase FIR filter are
1
a. │H(ejω)│=constant and angle (H(ejω))=𝜔
10.if Ɵ(ω) is the phase function of FIR filter then group delay and phase delay of FIR filters are
defined respectively as,
–𝒅Ɵ(𝝎) −Ɵ(𝝎)
a. ,
𝒅𝝎 𝝎
–𝑑Ɵ(𝜔)
b. ,-ωƟ(ω)
𝑑𝜔
Ɵ(𝜔) 𝑑Ɵ(𝜔)
c. ,
𝜔 𝑑𝜔
𝑑Ɵ(𝜔)
d. -ωƟ(ω) , 𝑑𝜔
11.if ωc1 and ωc2 are the cutoff frequencies of band pass filter ,then the response lies only in the
range of
a. -ωc2 ≤ ω ≤ 0 and ωc2 ≤ ω ≤ Π
b.-Π ≤ ω ≤ -ωc2 and -ωc1 ≤ ω ≤ 0
c. -ωc2 ≤ ω ≤-ωc1 and ωc1 ≤ ω ≤ ωc2
d.ωc1 ≤ ω ≤ ωc2 and ωc2 ≤ ω ≤ Π
12.if ωc1 and ωc2 are the cutoff frequencies of stop band filter ,then the response lies only in the
range of
a. -ωc2 ≤ ω ≤ -ωc1and ωc1 ≤ ω ≤ ωc2andωc2 ≤ ω ≤ Π
b. -Π ≤ ω ≤ -ωc2 and - ωc1 ≤ ω ≤ 0 and0 ≤ ω ≤ ωc1
c. -ωc2 ≤ ω ≤ 0 and ωc1 ≤ ω ≤ ωc2 and ωc2 ≤ ω ≤ Π
d. - Π ≤ ω ≤ -ωc2 and -ωc1 ≤ ω ≤ ωc1 and ωc2 ≤ ω ≤ Π
13.frequency response of LTI system ,with constant phase delay
+
a.H(ω) = │H(ω)│ e-jαω
−
+
b.H(ω) = │H(ω)│ ej(β-αω)
−
+
c.H(ω) = │H(ω)│ ejαω
−
+
d.H(ω) = │H(ω)│ e-j(β-αω)
−
14.The filters designed by using finite number of samples of impulse response are called -------
a.FIR filters
b.IIR filters
c.both a and b
d. none of these
15.in ------------ window spectrum the width of main lope is double that of rectangular window
for same value of N.
a.hamming
b. blackman
c. kaiser
d. hanning
PART C
1.For the given transfer function H(z)=H1(z)H2(z),where H1(z)=1/(1-0.5z-1) and
H2(z)= 1/(1-0.4z-1),find the output roundoff noise power.Calculate the value if b=3
2.A bandpass FIR filter of length 7 is required. It is to have lower and upper cut-off
frequencies of 3kHz respectively and is intended to be used with a sampling frequency of
24kHz.Determine the filter coefficients using Hanning window. Consider the filter to be causal.
3.Using frequency sampling method, design a bandpass filter with the following specifications.
Sampling frequency F=8000Hz, cut-off frequency fc1=1000Hz, cut-off frequency
fc2=3000Hz.determine the filter coefficients for N=7.
4. Design a linear phase FIR low pass filter using rectangular window by taking 7 samples of
window sequence and with a cutoff frequency ,ωc=0.2Π rad /sample.
5. Design a linear phase FIR high pass filter using hamming window with a cutoff frequency
,ωc=0.8Π rad /sample and N=7.
6. Design a linear phase FIR band pass filter to pass frequencies in the range 0.4Π to 0.65 Π rad
/sample by taking 7 samples of hanning window sequence.
7. Design a linear phase FIR bandstop filter to reject frequencies in the range 0.4Π to 0.65 Π
rad /sample using rectangular window ,by taking 7 samples of window sequence.
8.Design a linear phase FIR low pass filter using hamming window by taking 5 samples of
window sequence and with a cutoff frequency ,ωc=0.35Π rad /sample.
9. Design a linear phase FIR high pass filter using rectangular window with a cutoff frequency,
ωc=0.48Π rad /sample and N=5.
10. Design a linear phase FIR band pass filter to pass frequencies in the range 0.35Π to 0.458Π
rad /sample by taking 5 samples of rectangular window sequence.
UNIT –V
PART-A
2. Decimation is used to
a. decrease the sampling rate of a signal
b. increase the sampling rate of a signal
c. decrease the amplitude of a signal
d. increase the amplitude of a signal
3. interpolation is used to
a. decrease the sampling rate of a signal
b. increase the sampling rate of a signal
c. decrease the amplitude of a signal
d.increase the amplitude of a signal
1
4. if x(n)={ ,5,7,3,4,8,7,8} to be decimated by a factor 3, then decimated sequence is given by
↑
𝟏
a. x(n)={ ,3,7}
↑
1
b. x(n)={ ,4,0}
↑
1
c. x(n)={ ,4}
↑
7
d. x(n)={ ,8,0}
↑
1
5. if x(n)= { ,3,7} to be interpolated by a factor 3,then interpolated sequence is given by
↑
1
a. x(n)={ ,5,7,3,4,8,7,8}
↑
𝟏
b. x(n)={ ,0,0,3,0,0,7}
↑
1
c. x(n)={ ,0,0,0,3,0,0,0}
↑
0
d. x(n)={ ,1,0,3,0,7,0}
↑
6. decimation process is generally preceded by a low pass filter
a. to avoid aliasing after decimation
b. to avoid noise
c. to avoid image frequencies
d. none of these
7. if a signal x(n) is decimated by a factor M. the relation between the original and decimated signal is
given by
a. y(n)=x(nM)
𝑛
b. y(n)=x( )
𝑀
c. y(n)=Mx(n)
d. y(n)=nx(M)
8. if a signal x(n) is interpolated by a factor L. the relation between the original and decimated signal is
given by
a. y(n)=x(nL)
𝒏
b. y(n)=x(𝑳 )
c. y(n)=Lx(n)
d. y(n)=nx(L)
𝛱𝑘
b. H0[𝑒 𝑗(𝜔− 𝑀 ) ] Where k=0,1,…..M-1
2𝑘
c. H0[𝑒 𝑗(𝜔− 𝑀 ) ] Where k=0,1,…..M-1
𝟐𝜫𝒌
d. H0[𝒆𝒋(𝝎− 𝑴
)
] Where k=0,1,…..M-1
2𝛱𝑘
13. H0[𝑒 𝑗(𝜔− 𝑀
)
] Where k=0,1,…..M-1 means
2𝛱𝑀
a. frequency response H0[ejω] shifted by 𝑘
𝛱𝑘
b. frequency response H0[ejω] shifted by 𝑀
2𝐾
c. frequency response H0[ejω] shifted by 𝑀
𝟐𝜫𝒌
d. frequency response H0[ejω] shifted by 𝑴
20. if R(n) is covariance matrix and r(n) is cross-correlation vector ,than in RLS algorithm h(n) is
given by
𝑅(𝑛)
a. h(n) = 𝑟(𝑛)
𝑅−1 (𝑛)
b. h(n) =
𝑟(𝑛)
𝑅(𝑛)
c. h(n) = 𝑟−1 (𝑛)
𝑹−𝟏 (𝒏)
d. h(n) = 𝒓−𝟏(𝒏)
21. The processing of signal at different sampling rate is called ---------
a. multirate DSP
b. aliasing
c. interpolation
d. decimation
22. the ----------is the process of increasing the sampling rate.
a. aliasing
b. decimation
c. interpolation
d.none of these
25. to eliminate multiple images in output spectrum of interpolator for interpolation by I ,the output
spectrum is bandlimited to -------------.
𝜫
a.
𝑰
𝐼
b. 𝛱
𝛱
c. 2
d. none of these
PART-B
1. if x(n) and y(n) are input and output of a decimator with sampling rate conversion factor A ,then,
a. y(n) = x(n-A)
b. y(n) = x(n/A)
c. y(n) = x(n+A)
d. y(n) = x(An)
2. if X(ejω) and Y(ejω) are input and output spectrum of decimator then,
𝟏
a. Y(ejω) = X(ejω/D)
𝑫
b. Y(ejω) = D X(ejω/D)
1
c. Y(ejω) = X(ejωD)
𝐷
d. Y(ejω) = D X(ejωD)
3. to avoid aliasing at output during decimation by D ,the input signal of a decimator should be
bandlimited to
𝛱
a. 2𝐷
2𝛱
b.
𝐷
𝜫
c. 𝑫
𝛱
d. 𝐷2
4. if x(n) and y(n) are input and output of an interpolator with sampling rate conversion factor B ,then,
a. y(n)=x(Bn)
b. y(n)=x(n/B)
c. y(n)=x(n)/B
d. y(n)=B x(n)
5. if X(ejω) and Y(ejω) are input and output spectrum of an interpolator then,
a. Y(ejω) =I X(ejωI)
b. Y(ejω) =I X(ejω/I)
c. Y(ejω) = X(ejωI)
d. Y(ejω) =X(ejω/I)
6. In sampling rate conversion by rational factor ,---------- is performed first.
a. aliasing
b. decimation
c. interpolation
d.none of these
7. when the sampling rate conversion factor is very large then ------- sampling rate conversion is
preferred.
a.multistage
b. single stage
c. both a and b
d.none of these
8. The process of dividing a filter into a number of sub-filters is called---------.
a. polyphase decomposition
b. monophase decomposition
c. both a and b
d. none of these
9.the digital filter bank is a set of ---------- filters.
a. lowpass
b. highpass
c. bandpass
d. bandstop
10. the --------- banks are filter banks with complementary frequency response.
a. QMF
b. MF
c. QF
d. none of these
11. To eliminate multiple images at the output , during interpolation by I, the output is filtered to have a
bandwidth of ,
a. Π I
b. Π / I
c. I / Π
d. Π / I2
12. If A and B are integer sampling rate conversion factor for decimation and interpolation respectively,
then sampling rate conversion factor for conversion by rational factor is,
a. A / B
b. B / A
c. A2 / B
d. B / A2
13.In multistage decimation by D, where D=D1D2 ,which of the following is correct implementation?
a. x(n)→ ↓D → ↑D →y(n)
1 2
b . x(n)→ → →y(n)
↑D1 ↑D2
c. x(n)→ → →y(n)
↓D1 ↓D2
d. x(n)→ → →y(n)
↑D1 ↓D2
14.In multistage interpolation by I, where I=I1 I2 which of the following is correct implementation?
a. x(n)→ → →y(n)
↑I1 ↑I2
b. x(n)→ → →y(n)
↑ I1 ↓ I2
c. x(n)→ ↓ I → ↑I →y(n)
1 2
d. x(n)→ → →y(n)
↓ I1 ↓ I2
15.The poly phase decomposition of H(z) into L sections can be represented by the equation ,
a. H(z) = ∑𝐿𝑚=1 𝑍 −𝑚 𝐸𝑚 (𝑍 𝐿 )
b. H(z) = ∑𝑳−𝟏
𝒎=𝟎 𝒁
−𝒎
𝑬𝒎 (𝒁𝑳 )
c. H(z) = ∑𝐿𝑚=1 𝑍 𝑚 𝐸𝑚 (𝑍 𝐿 )
d. H(z) = ∑𝐿−1
𝑚=1 𝑍
−𝑚
𝐸𝑚 (𝑍 𝐿 )
PART C
1. Discuss on efficient transversal structure for decimator and Interpolator.
2. Discuss how image enhancement restoration and coding can be done using
signal processing.