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DSP objective questions

The document consists of a series of multiple-choice questions related to signal processing, covering topics such as analog and digital conversion, filter design, DFT, FFT algorithms, and system stability. Each question presents a specific concept or theorem, with options for answers that test the reader's understanding of the material. The questions are structured to assess knowledge in areas such as the characteristics of signals, the properties of filters, and the mathematical transformations used in signal processing.
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0% found this document useful (0 votes)
5 views9 pages

DSP objective questions

The document consists of a series of multiple-choice questions related to signal processing, covering topics such as analog and digital conversion, filter design, DFT, FFT algorithms, and system stability. Each question presents a specific concept or theorem, with options for answers that test the reader's understanding of the material. The questions are structured to assess knowledge in areas such as the characteristics of signals, the properties of filters, and the mathematical transformations used in signal processing.
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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1) The interface between an analog signal and a digital processor is [b]

a. D/A converter b. A/D converter c. Modulator d. Demodulator

2) The speech signal is obtained after [b]


a. Analog to digital conversion b. Digital to analog conversion c. Modulation
d. Quantization

3) Telegraph signals are examples of[a]


a. Digital signals b. Analog signals c. Impulse signals d. Pulse train

4) As compared to the analog systems, the digital processing of signals allow[d]


1) Programmable operations 2) Flexibility in the system design 3) Cheaper systems
4) More reliability
a. 1, 2 and 3 are correct b. 1 and 2 are correct c. 1, 2 and 4 are correct d. All the four
are correct

5) The Nyquist theorem for sampling[c]


1) Relates the conditions in time domain and frequency domain 2) Helps in quantization
3) Limits the bandwidth requirement 4) Gives the spectrum of the signal
a. 1, 2 and 3 are correct b. 1 and 2 are correct c. 1 and 3 are correct d. All the four are
correct

6) Roll-off factor is [a]


a. The bandwidth occupied beyond the Nyquist Bandwidth of the filter
b. The performance of the filter or device c. Aliasing effect d. None of the above

7) A discrete time signal may be[a]


1) Samples of a continuous signal 2) A time series which is a domain of integers
3) Time series of sequence of quantities 4) Amplitude modulated wave
a. 1, 2 and 3 are correct b. 1 and 2 are correct
c. 1 and 3 are correct d. All the four are correct

8) The discrete impulse function is defined by[b]


a. δ(n) = 1, n ≥ 0,δ(n) = 0, n ≠ 1 b. δ(n) = 1, n = 0,δ(n) = 0, n ≠ 0
c. δ(n) = 1, n ≤ 0, δ(n) = 0, n ≠ 1 d. δ(n) = 1, n ≤ 0,δ(n) = 0, n ≥ 1

9) DTFT is the representation of [b]


a. Periodic Discrete time signals b. Aperiodic Discrete time signals
c. Aperiodic continuous signals d. Periodic continuous signals

10) The transforming relations performed by DTFT are[d]


1) Linearity 2) Modulation 3) Shifting 4) Convolution
a. 1, 2 and 3 are correct b. 1 and 2 are correct
c. 1 and 3 are correct d. All the four are correct
11) The DFT is preferred for[c]
1) Its ability to determine the frequency component of the signal 2) Removal of noise
3) Filter design 4) Quantization of signal
a. 1, 2 and 3 are correct b. 1 and 2 are correct c. 1 and 3 are correct d. All the four are correct

12) Frequency selectivity characteristics of DFT refers to[a]


a. Ability to resolve different frequency components from input signal
b. Ability to translate into frequency domain
c. Ability to convert into discrete signal d. None of the above

13) The Cooley–Tukey algorithm of FFT is a [a]


a. Divide and conquer algorithm b. Divide and rule algorithm
c. Split and rule algorithm d. Split and combine algorithm

14) FFT may be used to calculate[b]


1) DFT 2) IDFT 3) Direct Z transform 4) In direct Z transform
a. 1, 2 and 3 are correct b. 1 and 2 are correct c. 1 and 3 are correct d. All the four are correct

15) DIT algorithm divides the sequence into[b]


a. Positive and negative values b. Even and odd samples
c. Upper higher and lower spectrum d. Small and large samples

16) The computational procedure for Decimation in frequency algorithm takes [a]
a. Log2 N stages b. 2Log2 N stages c. Log2 N2 stages d. Log2 N/2 stages

17) The transformations are required for[c]


1) Analysis in time or frequency domain 2) Quantization
3) Easier operations 4) Modulation
a. 1, 2 and 3 are correct b. 1 and 2 are correct c. 1 and 3 are correct d. All the four are correct

18) The s plane and z plane are related as[a]


a. z = esT b. z = e2sT c. z = 2esT d. z = esT/2

19) The similarity between the Fourier transform and the z transform is that[b]
a. Both convert frequency spectrum domain to discrete time domain
b. Both convert discrete time domain to frequency spectrum domain
c. Both convert analog signal to digital signal d. Both convert digital signal to analog signal

20) The ROC of a system is the [a]


a. range of z for which the z transform converges
b. range of frequency for which the z transform exists
c. range of frequency for which the signal gets transmitted
d. range in which the signal is free of noise
1) The several ways to perform an inverse Z transform are[d]
1) Direct computation 2) Long division 3) Partial fraction expansion with table lookup
4) Direct inversion
a. 1, 2 and 3 are correct b. 1 and 2 are correct c. 2 and 3 are correct d. All the four are correct

2) The anti causal sequences have ______ components in the left hand sequences.[a]
a. Positive b. Negative c. Both a and b d. None of the above

3) For an expanded power series method, the coefficients represent[a]


a. Inverse sequence values b. Original sequence values
c. Negative values only d. Positive values only

4) The region of convergence of x/ (1+2x+x2) is[b]


a. 0 b. 1 c. Negative d. Positive

5) The IIR filter designing involves[b]


a. Designing of analog filter in analog domain and transforming into digital domain
b. Designing of digital filter in analog domain and transforming into digital domain
c. Designing of analog filter in digital domain and transforming into analog domain
d. Designing of digital filter in digital domain and transforming into analog domain

6) For a system function H(s) to be stable[c]


a. The zeros lie in left half of the s plane b. The zeros lie in right half of the s plane
c. The poles lie in left half of the s plane d. The poles lie in right half of the s plane

7) IIR filter design by approximation of derivatives has the limitations[c]


1) Used only for transforming analog high pass filters
2) Used for band pass filters having smaller resonant frequencies
3) Used only for transforming analog low pass filters
4) Used for band pass filters having high resonant frequencies
a. 1, 2 and 3 are correct b. 1 and 2 are correct c. 2 and 3 are correct d. All the four are correct

8) The filter that may not be realized by approximation of derivatives techniques are[b]
1) Band pass filters 2) High pass filters 3) Low pass filters 4) Band reject filters
a. 1, 2 and 3 are correct b. 2 and 4 are correct c. 2 and 3 are correct d. All the four are correct

9) In direct form for realization of IIR filters,[c]


1) Denominator coefficients are the multipliers in the feed forward paths
2) Multipliers in the feedback paths are the positives of the denominator coefficients
3) Numerator coefficients are the multipliers in the feed forward paths
4) Multipliers in the feedback paths are the negatives of the denominator coefficients
a. 1, 2 and 3 are correct b. 1 and 2 are correct c. 3 and 4 are correct d. All are correct

10) The direct form II for realization involves[b]


1) The realisation of transfer function into two parts 2) Realisation after fraction
3) Product of two transfer functions 4) Addition of two transfer functions
a. 1, 2 and 3 are correct b. 1 and 3 are correct c. 3 and 4 are correct d. All are correct

11) The cascade realisation of IIR systems involves[b]


1) The transfer function broken into product of transfer functions
2) The transfer function divided into addition of transfer functions
3) Factoring the numerator and denominator polynomials
4) Derivatives of the transfer functions
a. 1, 2 and 3 are correct b. 1 and 3 are correct c. 3 and 4 are correct d. All the four are correct

12) The advantage of using the cascade form of realization is[c]


1) It has same number of poles and zeros as that of individual components
2) The number of poles is the product of poles of individual components
3) The number of zeros is the product of poles of individual components
4) Over all transfer function may be determined
a. 1, 2 and 3 are correct b. 1 and 3 are correct c. 1 and 4 are correct
d. All the four are correct

13) Which among the following represent/s the characteristic/s of an ideal filter?[d]
a. Constant gain in passband b. Zero gain in stop band c. Linear Phase Response
d. All of the above

14_________ filters exhibit their dependency upon the system design for the stability
purpose?[a]
a. IIR b. FIR c. FIR &IIR d. None

15. One dimensional signal is a function of___________[a]


a.single independent variable b. single dependent variable c.both d none

16. If g(n) is a real valued sequence of 2N points and x1(n)=g(2n) and x2(n)=g(2n+1), then
what is the value of G(k), k=0,1,2…N-1?[b]
a) X1(k)-W2kNX2(k) b) X1(k)+W2kNX2(k) c) X1(k)+W2kX2(k) d) X1(k)-W2kX2(k)

17. If g(n) is a real valued sequence of 2N points and x1(n)=g(2n) and x2(n)=g(2n+1), then
what is the value of G(k), k=N,N-1,…2N-1?[d]
a) X1(k)-W2kX2(k) b) X1(k)+W2kNX2(k) c) X1(k)+W2kX2(k) d) X1(k)-W2kNX2(k)

18.. How many complex multiplications are need to be performed for each FFT
algorithm?[c]
a) (N/2)logN b) Nlog2N c) (N/2)log2N d) None of the mentioned

19.. If x1(n) and x2(n) are two real valued sequences of length N, and let x(n) be a complex
valued sequence defined as x(n)=x1(n)+jx2(n), 0≤ n≤ N-1, then what is the value of x2(n)?[d]
a) x(n)−x∗(n)/2 b) x(n)+x∗(n)/2 c) x(n)+x∗(n)/2j d) x(n)−x∗(n)/2j

20.. If X(k) is the DFT of x(n) which is defined as x(n)=x1(n)+jx2(n), 0≤ n≤ N-1, then what is
the DFT of x1(n)?[a]
a) 1/2[X∗(k)+X∗(N−k)] b) 1/2[X∗(k)−X∗(N−k)] c) 1/2j[X∗(k)−X∗(N−k)] d) 1/2j[X∗(k)+X∗(N−k)]

1. If x1(n) and x2(n) are two real valued sequences of length N, and let x(n) be a complex
valued sequence defined as x(n)=x1(n)+jx2(n), 0≤n≤N-1, then what is the value of x1(n)?[b]
a) x(n)−x∗(n)/2 b) x(n)+x∗(n)/2 c) x(n)−x∗(n)/2j d) x(n)+x∗(n)/2j

2. If X(k) is the DFT of x(n) which is defined as x(n)=x1(n)+jx2(n), 0≤ n≤ N-1, then what is
the DFT of x1(n)?[c]
a) 1/2[X∗(k)+X∗(N−k)] b) 1/2[X∗(k)−X∗(N−k)] c) 1/2j[X∗(k)−X∗(N−k)] d)
1/2j[X∗(k)+X∗(N−k)]

3. How many complex additions are required to be performed in linear filtering of a


sequence using FFT algorithm?[b]
a) (N/2)logN b) 2Nlog2N c) (N/2)log2N d) Nlog2N

4. How many complex multiplication are required per output data point?[b]
a) [(N/2)logN]/L b) [Nlog22N]/L c) [(N/2)log2N]/L d) None of the
mentioned

5) Two vectors a, b are orthogonal if[a]


a. <a,b> = 0 b. <a,b> = <a,b> c. <a,b> = 1 d. <a,b> = - <a,b>
6) Superposition of signals in a linear system refers to the [b]
a. Output that is product of all the signals b. Output that is sum of all the signals
c. Output that is of highest amplitude of all the signals d. Output that is of largest spectrum of
all the signals
7) The scaling of a sequence x[n] by a factor α is given by[c]

a. y[n] = α [x[n]]2 b. y[n] = α x[n2] c. y[n] = α x[n] d. y[n] = x[n]x[-n]

8) DFT is applied to [b]

a. Infinite sequences b. Finite discrete sequences c. Continuous infinite signals


d. Continuous finite sequences
9) The filtering is performed using DFT using[c]

1) Limited size or blocks of data 2) Small memory size 3) Large memory size
4) Large segments of data

a. 1, 2 and 3 are correct b. 3 and 4 are correct c. 1 and 2 are correct d. All the four are correct
10) In Overlap-Add Method with linear convolution of a discrete-time signal of length L
and a discrete-time signal of length M, for a length N, zero padding should be of length[c]

a. L, M > N b. L, M = N c. L, M < N d. L, M < N2


11) Discrete cosine transforms (DCTs) express a function or a signal in terms of [a]

a. Sum of cosine functions oscillating at different frequencies


b. Sum of cosine functions oscillating at same frequencies
c. Sum of cosine functions at different sampling intervals
d. Sum of cosine functions oscillating at same sampling intervals
12) A system is said to be unstable if[c]

a. None of the poles of its transfer function is shifted to the right half of s-plane
b. At least one zero of its transfer function is shifted to the right half of s-plane
c. At least one pole of its transfer function is shifted to the right half of s-plane
d. At least one pole of its transfer function is shifted to the left half of s-plane
13) The nonlinear difference equations are solved using[c]

a. Iterative method b. Cobweb model c. Phase diagram d. Power series method


14) Correlation is used for[a]

1) Computation of average power in waveforms 2) Climatography


3) Identification of binary code word in PCM systems 4) Quantization

a. 1, 2 and 3 are correct b. 1 and 2 are correct c. 2 and 3 are correct d. All are correct
15. If x(n) is a discrete-time signal, then the value of x(n) at non integer value of ‘n’ is:[d]
a) Zero b) Positive c) Negative d) Not defined

16. The discrete time function defined as u(n)=n for n=0;=0 for n<0 is an:[c]
a) Unit sample signal b) Unit step signal c) Unit ramp signal d) None of the mentioned

17. Time scaling operation is also known as:[a]


a) Down-sampling b) Up-sampling c) Sampling d) None of the mentioned

18. Which of the following should be done in order to convert a continuous-time signal to a
discrete-time signal?[a]
a) Sampling b) Differentiating c) Integrating d) None of the mentioned

19. The process of converting discrete-time continuous valued signal into discrete-time
discrete valued(digital) signal is known as:[b]
a) Sampling b) Quantization c) Coding d) None of the mentioned

20. The difference between the un quantized x(n) and quantized xq(n) is known as:[d]
a) Quantization coefficient b) Quantization ratio c) Quantization factor
d) Quantization error

1. In the Frequency Transformations of the analog domain the transformation is [b]


a. Low Pass to Low pass b. Low pass to High pass c. Low pass to Band pass
d. Low pass to Band reject

2. In the Frequency Transformations of the analog domain the transformation is [d]


a. Low Pass to Low pass b. Low pass to High pass
c. Low pass to Band pass d. Low pass to Band reject

3. What is the convolution of the sequences of x1(n)=x2(n)={1,1,1}?[a]


a) {1,2,3,2,1} b) {1,2,3,2,1} c) {1,1,1,1,1} d) {1,1,1,1,1}

4. What is the z-transform of the signal x(n)=[3(2n)-4(3n)]u(n)?[a]


a) 3/(1-2z-1)-4/(1-3z-1) b) 3/(1+2z-1)-4/(1+3z-1) c) 3/(1-2z)-4/(1-3z)
d) None of the mentioned

5. The even part of a signal x(t) is:[c]


a) x(t)+x(-t) b) x(t)-x(-t) c) (1/2)*(x(t)+x(-t)) d) (1/2)*(x(t)-x(-t))

6. I I R digital filters are of the following nature[a]


a. Recursive b. Non Recursive c. Reversive d. Non Reversive

7. In I I R digital filter the present output depends on[d]


a. Present and previous Inputs only b. Present input and previous outputs only
c. Present input only d. Present Input, Previous input and output

8. Which of the following is best suited for I I R filter when compared with the FIR filter[a]
a. Lower side lobes in stop band b. Higher Side lobes in stop band
c. Lower side lobes in Pass band d. No side lobes in stop band

9. In the case of I I R filter which of the following is true if the phase distortion is
tolerable[c]
a. More parameters for design b. More memory requirement
c. Lower computational Complexity d. Higher computational complexity

10. A causal and stable I I R filter has[b]


a. Linear phase b. No Linear phase c. Linear amplitude d. No Amplitude

11. Neither the Impulse response nor the phase response of the analog filter is Preserved in
the digital filter in the following method[c]
a. The method of mapping of differentials b. Impulse invariant method
c. Bilinear transformation d. Matched Z - transformation technique

12. Out of the given I I R filters the following filter is the efficient one[b]
a. Circular filter b. Elliptical filter c. Rectangular filter d. Chebyshev filter

13. What is the disadvantage of impulse invariant method[a]


a. Aliasing b. one to one mapping c. anti aliasing d. warping

14. Which of the I I R Filter design method is anti aliasing method?[c]


a. The method of mapping of differentials b. Impulse invariant method
c. Bilinear transformation d. Matched Z - transformation technique

15. The nonlinear relation between the analog and digital frequencies is called [b]
a. aliasing b. warping c. pre warping d. anti aliasing

16. The most common technique for the design of I I R Digital filter is[b]
a. Direct Method b. In direct method c. Recursive method d. non recursive method

17. In the design a IIR Digital filter for the conversion of analog filter in to Digital domain
the desirable property is[b]
a.The axis in the s - plane should map outside the unit circle in the z - Plane
b.The Left Half Plane(LHP) of the s - plane should map in to the unit circle in the Z -
Plane
c.The Left Half Plane(LHP) of the s-plane should map outside the unit circle in the z-
Plane
d.The Right Half Plane(RHP) of the s-plane should map in to the unit circle in the Z -
Plane

18. Which of the following is a digital-to-analog conversion process?[d]


a) Staircase approximation b) Linear interpolation
c) Quadratic interpolation d) All of the mentioned

19. The quality of output signal from a A/D converter is measured in terms of:[c]
a) Quantization error b) Quantization to signal noise ratio
c) Signal to quantization noise ratio d) Conversion constant

20.The I I R filter design method that overcomes the limitation of applicability to only
Low pass filter and a limited class of band pass filters is[b]
a. Approximation of derivatives b. Impulse Invariance
c. Bilinear Transformation d. Frequency sampling

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