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Nadu.
[2013-2017] Batch / III IT - Sixth Semester
U13EC635 – Digital Signal Processing

QUESTION BANK

UNIT –1

1 A signal with large magnitude and short duration is called-------------

a) Unit signal b) Analog signal
c) Impulse signal d) Step signal
2 The periodic signals will have constant--------------
a) Energy b) Power
c) Amplitude d) Time
3 The ----------------Signals defined for t≥0
a) Causal signal b) Analog signal

c) Impulse signal d) Non causal signal

4 The differentiation of a unit step signal is
a) Unit signal b) Analog signal
c) Impulse signal d) Step signal
5. Sinc signal is a
a) Periodic signal b) Analog signal
c) Impulse signal d) Non periodic signal
6 In ---------------- systems the present output depends on past outputs.
a) Causal system b) Analog system
c) Impulse system d) Feedback system
7 The system that does not require memory is called-------------------
a) Causal system b) Static system
 c) Dynamic system d) Feedback system
8 LTI system supports
a) Causal response b) Analog response
c) Impulse response d) Feedback response
9 In stable system for any bounded input the output will be
a) Bounded b) Unbounded
c) Zero d) Unity
10 Signal flow graph of a system represents
a) Linear b) Graphical
c) Flow chart d) Blocks
11 A signal with infinite magnitude and zero duration is called-------------
a) Unit impulse signal b) Analog signal
c) Impulse signal d) Step signal
12. The non periodic signals will have constant--------------
a) Energy b) Power
c) Amplitude d) Time
13. The ----------------Signals defined for t≤0
a) Causal signal b) Analog signal
c) Impulse signal d) Non causal signal
14. The integral of a unit step signal is
a) Unit signal b) Analog signal
c) Impulse signal d) Step signal
15. et is a
a) Periodic signal b) Analog signal

c) Impulse signal d) Non periodic signal

16. In ---------------- systems depends on past inputs.
a) Causal system b) Analog system
c) Impulse system d) non casual system
17. The system that requires memory is called-------------------
a) Causal system b) Static system
 c) Dynamic system d) Feedback system
18. Causal system supports
a) Unit response b) Analog response
c) Impulse response d) Feedback response

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) Transfer function b) System function

c) Unit function d) impulse function
 Part - B

1. Check causality of the system y(n) = 3x(n-2) + 3x(n+2)

a) Unity causal b) non- causal
c) Causal d) Zero causal
2. Write the expression for Gaussian signal
a) ea2 t2 b) ea1 t1
c) ea3 t3 d) ea4 t4
3. Find the time period of the signal x (t) = 2sin t/4
a) 2π b) 4π
c) 6π d) 8π
4. Determine the odd part of the signal x (t) = 3+2t+5t2
a) 2t b) 4t
c) 6t d) 8t

5. Find the energy signal of the signal x(t) = e-2t u(t)

a) zero b) infinity
c) unity d) equality
6.
What about the stability of system in
a) system is stable b) unstable
c) stable at 0.4 d) cant say
7. An energy signal has G(f) = 10. Its energy density spectrum is
a) 10 b) 100
c) 50 d) 20
8. Find the fundamental period of the signal x (t) = 2cos t/4
a) 2π b) 4π
c) 6π d) 8π

9. Which one is a linear system?

a) y[n] = x[n] x x[n - 1] b) y[n] = x[n] + x[n - 10]
 c) y[n] = x2[n] d) (a) and (c)
10. Find the energy signal of the signal x(t) = t u(t)
a) zero b) infinity
c) Unity d) equality
11. Which of following is recursive 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

1.Verify the periodicity properties of the continuous time signals

(i) x(t) = 2 cos t/4 (ii) x(t) = 3cos(5t+ π/6)

2.Determine the power and energy of the continuous time signals

(i) x(t) = e-2t u(t) (ii) x(t) = 3cos 5Ωot

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.

5. Explain z transform properties.

6. Explain circular and linear convolution.

7.Compute circular convolution of following sequences x(n)={1,2,4}and h(n)={4,5,6,7}

8. Explain properties of discrete time systems.

9. Explain the block diagram of DSP systems.

10. Discuss sampling theorem.

 UNIT –II

PART-A

1. If x(n) and X(k) are an N-point DFT pair, then x(n+N)=x(n).

a) True
b) False
2. If x(n) and X(k) are an N-point DFT pair, then X(k+N)=?
a) X(-k)
b) -X(k)
c) X(k)
d) None of the mentioned
3. If X1(k) and X2(k) are the N-point DFTs of x1(n) and x2(n) respectively, then what is the N-
point DFT of x(n)=ax1(n)+bx2(n)?
a) X1(ak)+X2(bk)
b) aX1(k)+bX2(k)
c) eakX1(k)+ebkX2(k)
d) None of the mentioned
4. If x(n) is a complex valued sequence given by x(n)=xR(n)+jxI(n), then what is the DFT of
xR(n)?

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?

d) None of the mentioned

ans:a
8. If x1(n),x2(n) and x3(m) are three sequences each of length N whose DFTs are given as
X1(k),X2(k) and X3(k) respectively and X3(k)=X1(k).X2(k), then what is the expression for
x3(m)?

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)?

d) None of the mentioned

ans:c
25.The computation of XR(k) for a complex valued x(n) of N points requires:
a) 2N2 evaluations of trigonometric functions
b) 4N2 real multiplications
c) 4N(N-1) real additions
d) All of the mentioned

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

15.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

PART-C

1. Derive the equation for decimation in time algorithm for FFT.

2. Obtain the signal flow graph for computing 8 point DFT using radix-2 DIF FFT algorithm.

3. Using above flow graph compute DFT of x(n)=cos(nπ/4),0≤n≤7.

4. Discuss in detail the important properties of the DFT.

5. compute 8 point DFT using radix-2 DIF FFT algorithm x(n)={1,2,3,4,2,3,1,5}

6. Calculate the DFT of the sequence x(n)={1,1,-2,-2}.determine the response of LTI system by
radix-2 DIT FFT.

7. Derive the equation for decimation in frequency algorithm for FFT.

8. compute 8 point DFT using radix-2 DIT FFT algorithm x(n)={9,2,3,4,2,5,1,5}

9. Compare circular and linear convolution.

10. Compare decimation in time algorithm and decimation in frequency algorithm.

 UNIT –III

PART-A

1. IIR filters are designed by considering all

a. infinite samples of frequency response
b. finite samples of impulse response
c. infinite samples of impulse response
d. None of these
2. For the analog and digital IIR filters to be casual, the number of zeros should be
a. ≥number of poles
b. ≤number of poles
c. =number of poles
d. zero
3. An analog filter has poles at s=0,s= -2,s= -1.if impulse invariant transformation is employed then the
corresponding poles of digital filters are respectively,
a. 0,e-T/2,eT
b. 1,e-2T,eT
c. 1,e2T,e-T
d. 0,e-2T,e-T
4. An analog filter transfer function is given by, H(s)= 3/(s+1).when the filter is transformed to digital
filter using impulse invariant transformation, what are the poles and zeros of the filter?
a. zeros at z=0,poles at z=0.368
b. zeros at z=1,poles at z=0
c. zeros at z=0.368,poles at z=0
d. zeros at z=0,poles at z=1
5. The digital lowpass chebyshev filter with following specification is realized using impulse invariant
transformation. what should be the attenuation constant and order N of the filter?
o.75≤H(w)≤1.0 ; 0≤w≤0.4π
H(w)≤0.05 ; 0.5π≤w≤π
a. 0.9, N≥10
b. 0.1,N≤20
 c. 0.882,N≥6
d. 0.7,N≤5
6. In impulse invariant transformation the digital frequency ‘ω’ for a given analog frequency, Ω is given
by
a. ω= ΩT
b. ω= Ω/T
c. ω= T/Ω
d. ω=tan ΩT
7. The transfer function of a normalized low pass filter can be transformed to a high pass filter with
cutoff frequency, Ωc by the transformation,
a. s→ 1/s
b. s→ Ωc /s
c. s→s/ Ωc
d.s→Ωc
8. The zeros of the butter worth filters exist at
a. left half of s-plane
b. origin
c. infinity
d. right half of s-plane
9. The poles of butterworth transfer function lie,
a. symmetrically on a circle in s-plane
b. symmetrically on an ellipse in s-plane
c. antisymmetrically on a circle in s-plane
d.antisymmetrically on an ellipse in s-plane
10. In butter worth and chebyshev transfer function ,when N is even, the nature of poles are,
A.complex and exist as conjugate pair
b. complex but not conjugate pairs
c.one pole is complex and other poles are real
d. one pole is real and other poles are complex
11. In butter worth and chebyshev transfer function ,when N is odd, the nature of poles are,
A.complex and exist as conjugate pair
b. complex but not conjugate pairs
C.one pole is complex and other poles are real
d. One pole is real and other poles are complex
12. The relation between analog and digital frequency is nonlinear in case of
a. Impulse invariant transformation

b. bilinear transformation

c.frequency sampling

d. all the above

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

14. Which of the following is true for a chebyshev analog filter?

a. In type-1,the magnitude response is monotonic in pass band and equiripple in stop band.
b. in type-1,the magnitude response is monotonic in pass band and stop band.
c. in type-2,the magnitude response is equiripple in pass band and stop band.
d. in type-2,the magnitude response is monotonic in pass band and equiripple in stop band.

15. The poles of chebyshev transfer function lie,

a. symmetrically on a circle in s-plane
b. symmetrically on an ellipse in s-plane
c. antisymmetrically on a circle in s-plane
d.antisymmetrically on an ellipse in s-plane
16. IIR filters are
a. non-recursive type
b. recursive type
c. neither recursive nor non-recursive type
d. none of these
17. Non-linearity in the relationship between ω and Ω is known
a.Frequency warping
b. frequency non -warping
c. frequency mixing
d.aliasing

18. In bilinear transformation method, relationship between ω and Ω is given by

a. Ω = 2/Ts tan (ω/2)
b. Ω = tan (ω/2)
c. Ω = 1/Ts tan (ω/2)
d. Ω = 1/Ts tan (ω/4)

19. Butterworth filer have

a.wide band transition region
b.sharp transition region
c. oscillation in the transition region
d. none of these

20. Chebyshev filter have

a. wideband transition region
b.sharp transition region
c. oscillation in the transition region
d. none of these

21. Chebyshev filter contains

a. oscillations in the pass band
b. oscillations in the stop band
c.oscillations in the pass band and stop band
d.oscillations in the transition band

22. For chebyshev filter, cut-off frequency is given by

a. Ωc = Ωp/2
b. Ωc = Ωs/2
c. Ωc = Ωp
d. Ωc = Ωs

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

4. Consider the digital lowpass butterworth filter with following specification.

o.9≤H(w)≤1.0 ; 0≤w≤0.2π
H(w)≤0.1 ; 0.4π≤w≤π
what should be the order of the filter to realize the above specifications using bilinear transformation?
a.N≥3
b. N≥20
c. N≥4
d. N≥5

5. The normalized transfer function of 3rd order lowpass butterworth filter is

a. 1/(s3+1.414sn2+sn+1)
b. 1/[(sn+1)(sn2+sn+1)
c.1/s2(sn+1)
d.1/(sn3+sn2+sn+1)
6. The condition for a digital filter to be casual and stable is
a.h(n)=0 for n<0 and ∑∞
𝒌=−∞ 𝒉(𝒌)<∞

b. h(n)=0 for n<0 and ∑∞

𝑘=−∞ ℎ(𝑘)>∞

c. h(n)=0 for n>0 and ∑∞

𝑘=−∞ ℎ(𝑘)<∞

d. h(n)=0 for n>0 and ∑∞

𝑘=−∞ ℎ(𝑘)>∞

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

1. The frequency response of a digital filter is periodic in the range

a. 0<ω<2Π
b. -Π<ω<Π
c.0<ω<Π
d. 0≤ω≤2Π
2. The frequency response of FIR filter with constant group delay will be in the form,
a. H(ejω)= C e-jαω
b.H(ejω)= C ejαω
c. H(ejω)= C e-j(β-αω)
d. H(ejω)= C ejαω
3. In FIR filters the gibbs oscillations are due to
a. non-linear magnitude characteristics
b. non-linear phase characteristics
c. sharp transition from pass-band to stop-band
d. gradual transition from pass-band to stop- band
4.If ωc is the cutoff frequency of lowpass filter, then the response lies only in the range of ,
a. –ωc≤ω≤Π
b.–ωc ≤ ω ≤ ωc
c. –Π≤ ω ≤ -ωc
d. –ωc ≤ ω ≤ Π
5.If ωc is the cutoff frequency of high pass filter,then the response lies only in the range of ,
a.ωc ≤ ω ≤ Π and –Π≤ ω ≤ 0
b. – Π≤ ω ≤ -ωc and ωc ≤ ω ≤ Π
c. –ωc ≤ ω ≤ -Π and–ω≤ ω ≤ ωc
d. –ωc ≤ ω ≤ 0 and 0≤ ω ≤ ωc
6. Symmetric impulse response having even number of samples can be used to design,
a. low pass and high pass filters
b. low pass and high pass filters
c. low pass and bandstop filters
d. only low pass filters
7. Raised cosine windows also called generalized
a.hamming window
b. hanning window
c. rectangular window
d. blackman window
8. The symmetric impulse response having odd number of samples has
a. symmetric magnitude function
b. antisymmetric magnitude function
c. both a and b
d. none of these
9. The symmetric impulse response having even number of samples cannot be used to design,
a. low pass filter
b. bandstop filter
c. high pass filter
d.band pass filter
10.The width of the main –lope in rectangular window spectrum is ,
a. 4Π/N
b.16Π/N
c.8Π/N
d. 2Π/N
11. In hamming window spectrum the side-lope magnitude remains constant with,
a. decreasing ω
b. constant ω
c. increasing ω
d. none of these
12. In which window sequence, the width of the main lope can be adjusted by varying the length
N of
the window?
a.hamming

b.hanning

c.bartlett

d.kaiser

13.The condition for the impulse response to be antisymmetric is,

a. h(n)= -h(N-1-n)
b.h(n)= h(-n)
c.h(n)= h(N-1-n)
d.all the above

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

17.Band pass filter

a.Allows the signal between ωc1 to ωc2 to pass and reject the other signal
b, Allows the signal between 0 to ωc1 to pass and reject the other signal
c. Allows the signal between 0 to ωc2 to pass and reject the other signal
d. Rejects the signal between ωc1 to ωc2 and allows the other signal to pass

18.Band reject filter

a.Allows the signal between ωc1 to ωc2 to pass and reject the other signal
b. Allows the signal between 0 to ωc1 to pass and reject the other signal
c. Allows the signal between 0 to ωc2 to pass and reject the other signal
d. Rejects the signal between ωc1 to ωc2 and allows the other signal to pass

19.Stop band attenuation in dB is given by

a.-20 log10(δs)
b.20 log10(δs)
c.-10 log10(δs)
d.10 log10(δs)

20.Equation specification for hamming window is given by

𝟐𝜫𝒏
𝟎. 𝟓𝟒 − 𝟎. 𝟒𝟔𝒄𝒐𝒔 (𝑵−𝟏)
a. wHam(n)={ 𝟎≤𝒏≤𝑵−𝟏
𝟎 𝑶𝒕𝒉𝒆𝒓𝒘𝒊𝒔𝒆
2𝛱𝑛
0.5 − 0.5𝑐𝑜𝑠 (𝑁−1)
b.wHam(n)={ 0≤𝑛 ≤𝑁−1
0 𝑂𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒
2𝛱𝑛
0.46 − 0.54𝑐𝑜𝑠 (𝑁−1)
c. wHam(n)={ 0≤𝑛 ≤𝑁−1
0 𝑂𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒
𝛱𝑛
0.54 − 0.46𝑐𝑜𝑠 (𝑁−1)
d. wHam(n)={ 0≤𝑛 ≤𝑁−1
0 𝑂𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒
21.In Bartlett window
𝟖𝜫
a. for transition width desired attenuation is -27 dB
𝑵
8𝛱
b. for transition width desired attenuation is -43 dB
𝑁
8𝛱
c. for transition width desired attenuation is -32 dB
𝑁
8𝛱
d.for transition width desired attenuation is -25 dB
𝑁

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

23. In kaiser window filter length N is given by

𝑨−𝟕.𝟗𝟓
a. N ≥ 𝟐.𝟐𝟖𝟓∆𝝎
𝐴−7.95
b.N ≥ 14.36∆𝜔
𝐴−14.36
c.N ≥ 2.285∆𝜔
𝐴+7.95
d.N ≥ 2.285∆𝜔

24.In symmetric FIR filter ,when N is even ,|H(ejω)| is given by

𝑵
−𝟏 𝑵−𝟏
𝟐
a.∑𝒏=𝟎 𝟐𝒉(𝒏)𝐜𝐨𝐬[𝝎( − 𝒏)]
𝟐
𝑁−3
𝑁−1 2 𝑁−1
b. [ℎ ( ) + ∑𝑛=0 2ℎ(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

25.In symmetric FIR filter ,when N is odd ,|H(ejω)| is given by

 𝑁
−1 𝑁−1
2
a.∑𝑛=0 2ℎ(𝑛)cos[𝜔( − 𝑛)]
2
𝑵−𝟑
𝑵−𝟏 𝟐 𝑵−𝟏
b.[𝒉 ( ) + ∑𝒏=𝟎 𝟐𝒉(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

2.Values of h(n) of linear phase FIR filter are

a. Real and symmetrical
b.Imaginary and symmetrical
c. Complex and symmetrical
d.Complex and anti-symmetrical

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

5.2’s complement representation of (-0.5625)10 is

a.(1.011100)2
b. (1.011000)2
c.(1.001100)2
d.(1.011110)2
6.Impulse response of an ideal band reject 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]

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

b. │H(ejω)│=constant and angle (H(ejω))=-αω

c. │H(ejω)│=-αω and angle (H(ejω))=constant
 1
d. │H(ejω)│=𝜔 and angle (H(ejω))=constant

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

1.Sampling rate conversion process is used for

a. changing the sampling frequency of a signal
b. changing the amplitude of a signal
c. changing the phase of a signal
d. none of the above

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)

9. analysis filter banks are used

a. to decompose signal x(n) into set of M sub band signals
b. to combine M sub band signals to get y(n)
c. to remove noise in the signal
d. to remove image frequency in the signal
10.synthesis filter banks are used
a. to decompose signal x(n) into set of M sub band signals
b. to combine M sub band signals to get y(n)
c. to remove noise in the signal
d. to remove image frequency in the signal
11. DFT filter banks decomposes the signal with
a. M components with equally spaced frequency bands
b. M components with unequally spaced frequency bands
c. M components with equally and unequally spaced frequency bands
d. none of these

12. in DFT filter bank transfer function HK(ejω) is given by

4𝛱𝑘
a.H0[𝑒 𝑗(𝜔− 𝑀
)
] Where k=0,1,…..M-1

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

14. In Quadrature Mirror Filter Bank

a. the signals are down sampled before procesing
b.the signals are up sampled before
processing
c. the signals are processed directly
d. none of the above

15.in QMF bank, decimation and interpolation factor is

a. lesser than the number of bands of the filter
b. equal to or greater than the number of bands of the filter
c. equal to or lesser than the number of bands of the filter
d. none of the above
16. adaptive filters are based on
a. prior knowledge of signal and noise
b. on the knowledge of characteristics of the incoming signal and noise
c. on the knowledge of characteristics of the incoming signal
d. on the knowledge of characteristics of the incoming noise

17. in adaptive filters noise of secondary input is always

a.Correlated with desired signal of primary
b. Uncorrelated with noise of primary.
c.correlated with noise of primary
d.Uncorrelated with desired signal and noise of primary

18.in LMS algorithm, filter coefficients updating equation is given by

a. h(n+1)=h(n)-2μe(n)x(n)
b. h(n+1)=h(n)+2μe(n)x(n)
c. h(n+1)=h(n)+2e(n)x(n)
d. h(n+1)=h(n)+2μe(n

19. LMS algorithm is based on

a. steepest-descent method
b. gradient method
c. inversion method
d. none of these

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

23. the ----------is the process of decreasing the sampling rate.

a. aliasing
b. decimation
c. interpolation
d.none of these
24.to avoid aliasing of output spectrum of decimator for decimation by D, the input spectrum is -------
to Π/D.
a. bandlimited
b. changed
c. decreased
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.

3. Discuss the sub-band coding of speech signal with a suitable diagram.

4. Draw and explain the polyphase structure of a decimator.

5. Draw and explain the polyphase structure of an interpolator.
6. Compare the FIR and IIR filters.
7. Explain any two applications where adaptive filters are used.
8. Explain in detail about LMS adaptive algorithm.
9. Discuss the concept of musical sound processing.
10.Discuss the sampling rate conversion by a rational factor I / D.