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MCQ for DSP

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Mewar University, Gangrar, Chaittorgarh

Electronics and Communication Engineering Department
Multiple Choice Question Bank
Subject: DSP
(5th Semester ECE)
Unit-1
1.
a)
b)
c)
d)

DSP stands?
Digital signal processing
Discrete signal processing
Double signal processor
None of the above
i

2. Given that W
a) 0
b) 1
c) -1
d) e

3. Given that W
a) 0
b) 1
c) -1
d) e

2
N

2
N

, where N

3 . Then F

, where N

3. F

W N can be computed as F

N
2

can be computed as F

4. Determine the convolution sum of two sequences x(n) = {3, 2, 1, 2} and h(n) = {1, 2, 1, 2}
a) y(n) = {3,8,8,12,9,4,4}
b) y(n) = {3,8,3,12,9,4,4}
c) y(n) = {3,8,8,12,9,1,4}
d) y(n) = {3,8,8,1,9,4,4}
5. Sampling theorem:
a) fm<fs
b) fs>fm

c) fs>=2fm
d) fs=2fm
6. Application of Convolution:
a) FIR Filtering
b) Addition
c) Manipulation
d) None of these
7. Condition for aliasing problem:
a) fs<fm
b) fs<2fm
c)fs=fm
d) all of these
8. DFT stands as:
a. Discrete Fourier transform
b. digital function transform
c. digital frequency transform
d. none
9. FFT stands as:
a. Fast Fourier transform
b. Fourier function transform
c. Fast frequency transform
d. none
10. Twiddle factor:
i

a. W

e
i

b. W

2
N

2
N

c. W

2
N

d. none

11. Phase factor:

i

a. W

2
N

b. W

e
i

c. W

2
N

2
N

d. none

12. Calculate DFT of x (n) = {1, 0, 1, 0}.

a. x (k) = {2, 0, 2, 0}
b. x (k) = {1, 0, 1, 0}
c. x (k) = {2, 0, 1, 0}
d. none
13. Calculate DFT of x (n)= (n) .
a. 1
b. 0
i

c. W

2
N

d. none
14. Calculate DFT of x (n)= (n n0) where 0<n0<N.
a. e

jwn 0

b. e jwn0
c. 1

d. none
15. The FFT algorithms:
a. eliminate the redundant calculation and enable to analyze the spectral properties of a signal.
b. enable the redundant calculation and redundant to analyze the spectral properties of a signal.
c. a & b
d. none
16. The relation between DFT and Fourier series coefficients of a periodic sequence is
a. X(K) = Ck/N
b. X(K)= Ck
c. X(K) = NCk
d. X(K)=1/Ck
17. If x(n) ------N pt DFT------ X(K) Then x*(-n, (mod N)) -------------N pt DFT--------
___________
a. X*(-K)
b. X*(K)
c. X(-k)
d. X(K)
18. If the Nyquist rate for xa(t) is s , what is the Nyquist rate for d xa(t)/dt
a. ds/df
b. s
c. s/2
d. 2s
19. If the Nyquist rate for xa(t) is s , what is the Nyquist rate for xa(2t)
a. 2s
b. s/2
c. s
d. s/4
20. If the Nyquist rate for xa(t) is s , what is the Nyquist rate for xa2(t)
a. 2s
b. s/2
c. s
d. s/4
21. If the Nyquist rate for xa(t) is s , what is the Nyquist rate for xa(t)Cos( 0 t)
a. s + 20
b. s * 20
c. s /20

d. s - 20
22. The minimum sampling frequency for xa(t) is real with Xa(f) non-zero only for 9 KHz < |f| <
12 KHz is
a. 4.5 KHz
b. 6 KHz
c. 9 KHz
d. 12 KHz
23. The minimum sampling frequency for xa(t) is real with Xa(f) non-zero only for 18 KHz < |f| <
22 KHz is
a. 8.8 KHz
b. 9 KHz
c. 11 KHz
d. 17.6 KHz
24. The minimum sampling frequency for xa(t) is complex with X a(f) non-zero only for 30 KHz <
|f| < 35 KHz is
a. 6 KHz
b. 5 KHz
c. 15 KHz
d. 17.5 KHz
25. Find two different continuous-time signals that will produce the sequence x(n) = cos( 0.15
n) when sampled with a sampling frequency of 8 KHz.
a. sine(1200t) and Cos(17200t)
b. Cos(1200t) and Sine(17200t)
c. Cos(1200t) and Cos(17200t)
d. Sine(1200t) and Sine(17200t)
26. A continuous-time signal xa(t) is known to be uniquely recoverable from its samples xa(nTs)
when Ts = 1 ms. What is the highest frequency in Xa( f )?
a. 500 Hz
b. 1000 Hz
c. 700 Hz
d. 5 KHz
27. Suppose that xa(t) is bandlimited to 8 kHz (that is, Xa( f ) = 0 for |f| > 8000), then what is
the Nyquist rate for xa(t)?
a. 16 KHz
b. 4 KHz
c. 8 KHz
d. 12 KHz
28. Suppose that xa(t) is bandlimited to 8 kHz (that is, Xa( f ) = 0 for |f| > 8000), then what is
the Nyquist rate for xa(t)cos(2 . 1000t)?
a. 16 KHz
b. 4 KHz
c. 18 KHz
d. 5 KHz

28. If a continuous-time filter with an impulse response ha(t) is sampled with a sampling
frequency of fs , what happens to the cutoff frequency wc of the discrete-time filter as fs is
increased?
a. wc increases
b. wc decreases
c. wc remains constant
d. wc depends upon fs
29. A complex bandpass signal xa(t) with Xa(f) nonzero for 10 kHz < f < 12 kHz is sampled at a
sampling rate of 2 kHz. The resulting sequence is x(n) = (n), then x a(t) will be
a. xa(t) = (1/2000) (Sine(2000t)/(t))ej2(11000)t
b. xa(t) = (1/2000) (Sine(2000t)/(t))e-j2(11000)t
c. xa(t) = (1/2000) (Cos(2000t)/(t))e j2(11000)t
d. xa(t) = (1/2000) (Cos(2000t)/(t))e-j2(11000)t
30. If the highest frequency in xa(t) is f = 8 kHz, then the minimum sampling frequency for the
bandpass signal ya(t) = xa(t) Cos(0t) if 0 = 2.20.103 will be
a. 56 KHz
b. 64 KHz
c. 16 KHz
d. 32 KHz
31. Drawbacks of DSP is
a. Digital processing needs pre and post processing devices
b. high cost
c. No memory storage
d. none of above
32. Drawbacks of DSP is
a. Digital processing needs A/D and D/A converters and associated reconstruction filters
b. high cost
c. No reliable
d. none of above
33. Advantages of DSP are:
a. low cost
b. stable
c. reliable
d. all of above
34. Advantages of DSP are:
a. predictable
b. repeatable
c. Sharing a single processor among a number of signals by time sharing
d. all of above

35. Advantages of DSP are:

a. low cost
b. repeatable
c. storage of data is very easy
d. all of above
36. Application of DSP:
a. Military
b. telecommunication
c. consumer electronics
d. all of above
37. Application of DSP:
a. medicine
b. seismology
c. signal filtering
d. all of above
38. Fast convolution techniques:
a. overlap save
b. overlap add
c. a & b
d. none of above
39. Correlation
a. It gives a measure of similarity between two data sequences.
b. It gives a measure of dis-similarity between two data sequences
c. a & b
d. none of above
40. Find the response of an FIR filter with impulse response h(n)= {1,2,4} to the input sequence
x(n)={1,2}.
a. y(n)={1,4,8,8}
b. y(n)={1,4,6,6}
c. y(n)={1,2,8,8}
d. none of above
Answer Key Unit-1:
1

10 11 12 13 14 15 16 17 18 19 20

21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40

Unit-2
1. IIR filters
a) Use feedback
b) Are sometimes called recursive filters
c) Can oscillate if not properly designed
d) all of the above
2. A Blackman window can eliminate ripple in FIR filters. The tradeoff is
a) larger transition bandwidth
b) smaller transition bandwidth
c) a non-linear phase response
d) possible instability
3. The output of two digital filters can be added. Or, the same effect can be achieved by
a) adding their coefficients
b) subtracting their coefficients
c) convolving their coefficients
d) averaging their coefficients and then using a Blackman window
4. The letter A below indicates the filter

a) stopband
b) transition band
c) passband

d) ripple
5. A DSP convolves each discrete sample with four coefficients and they are all equal to 0.25. This
must be a
a) low-pass filter
b) high-pass filter
c) band-pass filter
d) band-stop filter
6. The inverse Fourier transform
a) converts from the frequency domain to the time domain
b) converts from the time domain to the frequency domain
c) converts from the phasor domain to the magnitude domain
d) is used to make real-time spectrum analyzers
7. This is the impulse response for

a) an IIR highpass filter

b) an FIR bandpass filter
c) an IIR lowpass filter
d) an FIR lowpass filter
8. Coefficient symmetry is important in FIR filters because it provides
a) a smaller transition bandwidth
b) less passband ripple

c) less stopband ripple

d) a linear phase response
10. This time graph shows the

a) frequency response of an IIR filter

b) amplitude response of an IIR filter
c) impulse response of an IIR filter
d) none of the above
11. Curve A is the

a) phase response of a lowpass filter

b) amplitude response of a lowpass filter
c) both of the above
d) none of the above
12. This windowed sinc FIR filter has ripple caused by

a) non-symmetrical coefficients
b) Gibb's phenomenon
c) too few taps
d) a defective accumulator
13. Two digital filters can be operated in cascade. Or, the same effect can be achieved by
a) adding their coefficients
b) subtracting their coefficients
c) convolving their coefficients
d) averaging their coefficients and then using a Blackman window
14. A DSP convolves each discrete sample with four coefficients and they are all equal to 0.25. This
must be an

a) IIR filter
b) FIR filter
c) RRR filter
d) All of the above
15. This frequency response graph is for a

a) lowpass filter
b) highpass filter
c) bandpass filter
d) bandstop filter
16. The letter C below indicates the filter

a) stopband
b) passband
c) transition band
d) ripple
17. A quantizer operates at a sampling frequency of 16 kHz. What is its Nyquist limit?
a) 4 kHz

b) 8 kHz
c) 16 kHz
d) 32 kHz
18. Curve B is the

a) phase response of a lowpass filter

b) amplitude response of a lowpass filter
c) both of the above
d) none of the above
19. After point D (as frequency is increasing)

a) the phase response is linear

b) the phase response is non-linear
c) the stopband is infinite
d) the Nyquist limit has been exceeded
20. The letter B below indicates the filter

a) stopband
b) passband
c) transition band
d) ripple
21. If a linear phase filter has a phase response of 40 degrees at 200 Hz, what will its phase response
be at a frequency of 400 Hz (assuming that both frequencies are in the passband of the filter)?
a) 35 degrees
b) 40 degrees
c) 45 degrees
d) 80 degrees
22. Which of the following is used to alter FIR filter coefficients so they smoothly approach zero at
both ends?
a) rectangular window
b) Blackman window
c) Laplace window
d) Hilbert window
23. Point C is called

a) a phase reversal
b) the half-power point
c) a phase discontinuity
d) a phase wrap
24. A DSP convolves each discrete sample with these coefficients: -0.25, -0.25, 1.0, -0.25, and -0.25.
This must be a
a) low-pass filter
b) high-pass filter
c) band-pass filter
d) band-stop filter
25. The basic process that's going on inside a DSP chip is
a) quantization
b) MAC
c) logarithmic transformation
d) vector calculations

26. For the rectangular window function, the transition width of the main lobe is approximately
(here M is the length of the filter)
a)
4*pi*M
b)
pi/4M
c)
pi*M/4
d)
4*pi/M

27. For the rectangular window function, the first sidelobe will be __________ dB down the
peak of the main lobe.
a)
12 dB
b)
11 dB
c)
13 dB
d)
14 dB
28. For the rectangular window function, the roll-off will be __________ dB/decade.
a)
25 dB
b)
20 dB
c)
15 dB
d)
10 dB
29. For the hamming window function, the width of the main lobe is approximately (here M is
the length of the filter)
a)
8*pi*M
b)
pi/8M
c)
pi*M/8
d)
8*pi/M
30. For the hamming window function, the peak of the first sidelobe will be at __________ dB.
a)
-40 dB
b)
-48 dB
c)
-43 dB
d)
-45 dB
31. For the hamming window function, the side lobe roll-off will be __________ dB/decade.
a)
25 dB
b)
20 dB
c)
15 dB
d)
10 dB
32. For the hanning window function, the width of the main lobe is approximately (here M is the
length of the filter)
a)
8*pi*M
b)
pi/8M
c)
pi*M/8
d)
8*pi/M
33. For the hamming window function, the peak of the first sidelobe will be at __________ dB.

a)
b)
c)
d)

-35 dB
-32 dB
-40 dB
-43 dB

34. For the Blackmann window function, the width of the main lobe is approximately (here M is
the length of the filter)
a)
12*pi*M
b)
pi/8M
c)
pi*M/8
d)
12*pi/M
35. For the hamming window function, the peak of the first sidelobe will be at __________ dB.
a)
-58 dB
b)
-48 dB
c)
-45 dB
d)
-43 dB

36. What is a delay?

a. Delay a copy of the output signal (by x number of samples), and combine it with the new input
signal.
b. Delay a copy of the input signal (by x number of samples), and combine it with the new output
signal.
c. a & b
d. none of above
37. What is FIR filter?
a. FIR filters are finite there is a specific limit to the number of times that any delayed sample
is added to a new input sample.
b. FIR filters are finite there is a specific limit to the number of times that any delayed sample
is added to a new output sample.
c. a & b
d. none of above
38. The output of a filter is a function not only of the input at the present time:

a. but also of previous events.

b. but also of future events.
c. a & b
d. none of above
39. FIR filters have ., and IIR filters have .
a. Zeros, poles & zeros
b. poles & zeros, Zeros
c. Zeros, zeros
d. none of above
40. How to define IIR filters term as infinite:
a. As with any feedback device, create a loop, hence the term infinite.
b. As with any non-feedback device, create a loop, hence the term infinite.
c. As with any feedback device, create a open loop, hence the term infinite.
d. None of above
Answers-Key Unit-2:
1

10 11 12 13 14 15 16 17 18 19 20

21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40

Unit-3
1. The filter coefficients are stored in:
a. binary registers
b. digital system
c. binary memory
d. none
2. Truncation or rounding of the data results in
a. degradation of system performance
b. increase system performance
c. grow power
d. none
3. the process of quantization is introduce
a. error
b. noise
c. power
d. none
4. Issue connected with finite word length effects:
a. quantization effects in A/D conversion
b. product quantization and coefficient quantization errors in digital filters
c. a & b
d. none.
5. Issue connected with finite word length effects:
a. limit cycles in IIR filters
b. product quantization and coefficient quantization errors in digital filters
c. a & b
d. none.
6. Issue connected with finite word length effects:
a. finite word length effects in FFTs
b. product quantization and coefficient quantization errors in digital filters
c. limit cycles in IIR filters
d. all of above
7. Rounding or truncation introduces an error whose magnitude depends
a. On the number of bits truncated or rounded bits.
b. On the number of bits rounded bits.

c. On the number of bits truncated bits.

d. all of above.
8. The range for negative truncation error for sign magnitude representation is
0
a. (2 B 2 L )
T
b. 0
c.

(2

(2

2 L)

2 L)
T

d. none of above
9. The range for positive truncation error for sign magnitude representation is
0
a. (2 B 2 L )
T
b. 0
c.

(2

(2

2 L)

2 L)
T

d. none of above
10. The range for truncation error for twos complement representation is
0
a. (2 B 2 L )
T
b. 0
c.

(2

(2

2 L)

2 L)
T

d. none of above
11. The range for round off error for sign magnitude representation is
(2 B 2 L ) / 2
a. (2 B 2 L ) / 2
R
b. 0
c.

(2

(2

2 L)

2 L)
T

d. none of above
12. The range for round off error for twos complement representation is
(2 B 2 L ) / 2
a. (2 B 2 L ) / 2
R
b. 0
c.

(2

(2

2 L)

2 L)
T

d. none of above
13. The dynamic range is
a. DR=6B + 10.8
b. DR=3B + 10.8

c. DR=6B + 1.8
d. none of above
14. The dynamic range is
a. DR =-2*logPe(n)
b. DR =-logPe(n)
c. DR =-10*logPe(n)
d. none of above

x 2 (n )

15.
n 0

a.

1
X(z)X(z 1 )z 1dz
2 jc

b.

1
X ( z)z 1dz
2 jc

c.

1
2

X(z)X(z 1 )z 1dz
c

d. none of above

16. Coefficient quantization effects in Direct form realization of IIR filters is

a. Y(z) = [Hideal(z) X(z) + E(z)]
b. Y (z) = [H ideal(z) + E(z)]
c. Y(z) = [Hideal(z) X(z)]
d. none of above
17. Stray filter
a. coefficient quantization error in Direct form realization of IIR filters
b. coefficient quantization error in cascaded-Direct form realization of IIR filters
c. coefficient quantization error in ladder form realization of IIR filters
d. none of above
18. Limit cycle is
a. zero input limit cycle
b. overflow limit cycle

c. a & b
d. none of above
19. The effects of limit cycles in first and second order systems were studied by
a. Hendy using an effective value model
b. Thomson using an effective value model
c. Jackson using an effective value model
d. none of above
20. If a is positive the limit cycle will have
a. variable magnitude by alternating sign.
b. constant magnitude by alternating sign.
c. constant phase by alternating sign.
d. none of above
21. What is scaling?
a. Scaling must be done in such a way that no overflow occurs at the summing point.
b. Scaling must be done in such a way that overflow occurs at the summing point.
c. Scaling must be done in such a way that no underflow occurs at the summing point.
d. none of above
22. The necessary and sufficient condition for preventing overflow in a IIR digital filter.

a. X

h i (k )
k

b. X

h i (k )
k

c. X

h i (k )
k

d. none of above
23. The necessary and sufficient condition for preventing overflow in a FIR digital filter.

a. X

h i (k )
k

b. X

h i (k )
k

c. X

1
M 1

h i (k )
k 0

d. none of above

24. The band of integers is known as the "deadband".

a. true
b. false
c. either true or false
d. none of above
25. In the second order system, under rounding, the output assumes a cyclic set of values of the
deadband.
a. limit-cycle.
b. band-cycle.
c. dead-cycle.
d. none of above

26. With finite precision the reponse does not converge to the origin but assumes cyclically a set
of values:
a. the limit-cycle.
b. band-cycle.
c. dead-cycle.
d. none of above
27. With infinite precision the response converges to the ..........
a. origin.
b. center.
c. mid.
d. none of above
28. below figure shows:

a. Quantisation error in rounding.

b. Quantisation error in truncation in 2s complement.
c. Quantisation error in truncation in sign magnitude.
d. none of above
29. below figure shows:

a. Quantisation error in rounding.

b. Quantisation error in truncation in 2s complement.
c. Quantisation error in truncation in sign magnitude.
d. none of above
30. Below figure shows:

a. probabilistic characteristics of Quantisation error in round-off.

b. probabilistic characteristics of Quantisation error in truncation in 2s complement.
c. probabilistic characteristics of Quantisation error in truncation in sign magnitude.
d. none of above

31. Below figure shows:

a. probabilistic characteristics of Quantisation error in round-off.

b. probabilistic characteristics of Quantisation error in truncation in 2s complement.
c. probabilistic characteristics of Quantisation error in truncation in sign magnitude.
d. none of above
32. This is a deterministic frequency response error is referred to as...
a. coefficient quantization error
b. product quantization error
c. a & b
d. none of above
33. A digital system is characterized by the difference equation y(n)=0.9 y(n-1) + x(n) with
x(n)=0 and initial condition y(-1)=12. Determine deadband of the system.
a. [-5,5]
b. [-3 , 3]
c. [-1,1]

d. none of above
34. FIR filters are .. generally as sensitive to coefficient roundoff.
a. not
b. less
c. most
d. none of above
35. FIR filters often require more computation, because you must do for each term in
the impulse response.
a. a multiply-add
b. add
c. multiply
d. all of above
36. FIR filters can be . delay, IIR filters can.
a. constant, not
b. not, constant
c. not, not
d. none of above
37. Linear Phase (constant delay), If a filter has a .delay, the phase shift of the filter
will be t*w, where t is the time delay, and w the natural frequency
(2*pi*f). a.
a. constant
b. variable
c. equal
d. none of above

38. Non-linear delay, This is the part of the phase shift (in and around the filters passband) that
is not modeled by a ...
a. straight line
b. circle
c. square
d. none of above
39. Power of quantization noise
a. p e ( n )

b. p e ( n )

c. p e ( n )

2 2B
12
2B

2
2 B
12

d. none of above
40. If you dont want a zero at pi, you cant use a symmetric -length filter. You can use an
antisymmetric even length filter if you want a highpass filter, but then youll have a zero at DC.
This means that symmetric high pass filters are of length.
a. even , odd
b. odd, even
c. even, even
d. none of above
Answers-Key Unit-3:
1

10 11 12 13 14 15 16 17 18 19 20

21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40

Unit-4
1. Used to increase the sampling rate by an integer factor
a. Up-sampler
b. down sampler
c. a & b
d. none of above
2. Used to decrease the sampling rate by an integer factor
a. Up-sampler
b. down sampler
c. a & b
d. none of above
3. This block represents

a. Up-sampler
b. down sampler
c. a & b
d. none of above
4. Up-sampling operation is implemented by inserting L-1 equidistant ..-valued samples
between two consecutive samples of x[n].
a. zero
b. one
c. two
d. none of above

5. Input-output relation for ..

x u [n ]

x[n /L],
0,

0, L, 2L,
otherwise

a. Up-sampler
b. down sampler
c. a & b
d. none of above
6. In practice, the zero-valued samples inserted by the up-sampler are replaced with appropriate
nonzero values using some type of filtering process, Process is called.
a. interpolation
b. decimation
c. a & b
d. none of above
7. operation is implemented by keeping every M-th sample of x[n] and removing M-1 inbetween samples to generate y[n].
a. Up-sampling
b. Down-sampling
c. a & b
d. none of above
8. Input-output relation for y[n] = x[nM]
a. Up-sampler
b. down sampler
c. a & b
d. none of above
9. The up-sampler and the down-sampler are ..but time-varying discrete-time systems:

a. linear
b. none linear
c. a & b
d. none of above
j
10. A factor-of-2 sampling rate expansion leads to a compression of X(e ) by a factor of 2 and
a 2-fold repetition in the baseband [0, 2 ]. This process is called

a. imaging
b. sampling
c. decimation
d. none of above
11. A ..is formed by an interconnection of the up-sampler, the down-sampler, and the
components of an LTI digital filter.
a. complex multirate system
b. complex single-rate system
c. a & b
d. none of above
12. An interchange of the positions of the branches in a cascade often can lead to a
computationally .realization.
a. efficient
b. non-efficient
c. neither efficient nor non- efficient
d. none of above
13. To implement a ..in the sampling rate we need to employ a cascade of an up-sampler
and a down-sampler.
a. fractional change

b. constant change
c. variable change
d. none of above
14. A cascade of a factor-of-M down-sampler and a factor-of-L up-sampler is interchangeable
with no change in the input-output relation: y1[n] y 2 [n] if and only if M and L are relatively
..
a. prime
b. non prime
c. natural number
d. none of above
15. From the sampling theorem it is known that a the sampling rate of a critically sampled
discrete-time signal with a spectrum occupying the full Nyquist range cannot be reduced any
further since such a reduction will introduce..
a. aliasing
b. quantization
c. error
d. none of above
16. The bandwidth of a critically sampled signal must be reduced by filtering before its
sampling rate is reduced by a down-sampler.
a. lowpass
b. highpass
c. a& b
d. none of above
17. The zero-valued samples introduced by an up-sampler must be interpolated to more
appropriate values for an effective sampling rate...
a. decrease

b. increase
c. a & b
d. none of above
18. Since up-sampling causes periodic repetition of the basic spectrum, the unwanted images in
the spectra of the up-sampled signal x u [n ] must be removed by using a lowpass filter H(z),
called..
a. the interpolation filter
b. the decimation filter
c. the Low pass filter
d. all of above
19. Down-sampling, the signal v[n] should be bandlimited to
filter, called ...

/ M by means of a lowpass

a. the interpolation filter

b. the decimation filter
c. the Low pass filter
d. all of above
20. In the case of single-rate digital signal processing, IIR digital filters are, in general,
computationally more efficient than equivalent FIR digital filters, and are therefore preferred
where computational .needs to be minimized.
a. cost
b. memory
c. speed
d. none of above
21. Below figure shows:

Decimation
a. Aliasing Step
b. Anti-Aliasing Step
c. a & b
d. all of above
22. Below figure shows:

Interpolation
a. Imaging Step
b. Anti-Imaging Step
c. a & b
d. all of above
23. To prevent, the down-sampled signal should be band-width limited to fs/2N by
low-pass filtering prior to sample removal.
a. aliasing
b. quantization
c. a & b
d. all of above
24. A signal can be restored to a higher sampling frequency by the processes of.
a. up sampling and interpolation

b. down sampling and decimation

c. a & b
d. all of above
25. .have the property of noise-shaping, which allows the elimination of quantization
noise by low-pass filtering.
a. Delta-Delta quantizers
b. Delta-sigma quantizers
c. a & b
d. all of above
26. Care has to be taken with any feedback system. Feedback coefficients have to remain
below..
a. 1.0
b. 2.0
c. 1.5
d. all of above
27. Drawbacks of IIR filters are:
a. phase distortion and ringing.
b. prevent phase distortion
c. more computation
d. all of above
28. Drawbacks FIR filters are:
a. more computation than an IIR with similar effect.
b. prevent phase distortion
c. less computation

d. all of above
29. Which of the Following is not a Filter?
a. Graphic Equalizer on a stereo system
b. Tone control on a stereo system
c. Ear
d. all of above
30. Finite Impulse Response (FIR) is a
a. feedforward filter
b. feedback filter
c. a & b
d. all of above
31. The direct form creates a number of difficulties:
a. It increases the size (in terms of bit depth) of numerical coefficients
b. It increases the depth required for accumulators (mantissa for floating point)
c. a & b
d none of above
32. A .is nothing but a way to implement a set of filters, generally strongly
mathematically related, in one operation.
a. filterbank
b. summing point bank
c. node bank
d. none of above
33. A can always be decomposed into a set of individual filters; this is usually a lot more
work that its worth, but not always.

a. filterbank
b. summing point bank
c. node bank
d. none of above
34. The MDCT, or Modified Discrete Cosine Transform, its a
a. filter-bank.
b. transform
c. a & b
d. none of above
35. Programmable DSP with .can be used to implement digital filters.
a. MAC
b. MAA
c. ADD
d. none of above
36. For high-bandwidth signal processing applicationscan provide multiple MACs to
achieve the desired thoughput.
a. FPGA technology
b. Nanotechnology
c. MEMS technology
d. none of above
37. The roots of polynomial F(z) define the zeros of the filter. FIRs are also called...........
a. all zero filters
b. all poles filters
c. all poles and zero filters

d. none of above
38. A variation of the direct FIR model is called the transposed FIR filter. It can be constructed
from the direct form FIR filter by
a. Exchanging the input and output
b. Inverting the direction of signal flow
c. Substituting an adder by a fork, and vice versa
d. all of above
39. The direct form FIR filter needs ................ between the adders to reduce the delay of the
adder tree and to achieve high throughput.
a. extra pipeline registers
b. less pipeline registers
c. Not pipeline registers
d. all of above
40. The FIR filter with transposed structure has registers between the adders and can achieve
high ........without adding any extra pineline registers.
a. throughput
b. speed
c. memory
d. all of above
Answers-Key Unit-4:
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