Digital Electronic MCQ Set 1
1. Which of the following is the magnitude frequency response of High pass filter?
d) None of the mentioned
Answer
Answer: b [Reason:] The property of a high pass filter is to pass the signals with high frequency and stop low frequency signal, which is as shown in the magnitude frequency response of ‘b’.
2. Which filter has a magnitude frequency response as shown in the plot given below?
a) Low pass Filter
b) High pass Filter
c) Band pass Filter
d) Band stop Filter
Answer
Answer: d [Reason:] In the magnitude response shown in the question, the system is stopping a particular band of signals. Hence the filter is called as Band stop filter.
3. An ideal filter should have zero gain in their stop band.
a) True
b) False
Answer
Answer: a [Reason:] For an ideal filter, in the magnitude response plot at the stop band it should have a sudden fall which means an ideal filter should have a zero gain at stop band.
4. The ‘Envelope delay’ or ‘Group delay’ is the time delay that the signal component of frequency ω undergoes as it passes from the input to the output of the system.
a) True
b) False
Answer
Answer: a [Reason:] The time delay taken to reach the output of the system from the input by a signal component is called as envelope delay or group delay.
5. If the phase ϴ(ω) of the system is linear, then the group delay of the system:
a) Increases with frequency of signal
b) Constant
c) Decreases with frequency of signal
d) Independent of frequency of signal
Answer
Answer: b [Reason:] We know that the group delay of the system with phase ϴ(ω) is defined as
Tg(ω)=(dϴ(ω))/dω
Given the phase is linear=> the group delay of the system is constant.
6. A two pole low pass filter has a system function 
What is the value of ‘p’ such that the frequency response H(ω) satisfies the condition |H(π/4)|2=1/2 and H(0)=1?
a) 0.46
b) 0.38
c) 0.32
d) 0.36
Answer
Answer: c [Reason:] Given
Upon solving the above quadratic equation, we get the value of p as 0.32.
7. A two pole low pass filter has a system function What is the value of ‘b0’ such that the frequency response H(ω) satisfies the condition |H(π/4)|2=1/2 and H(0)=1?
a) 0.36
b) 0.38
c) 0.32
d) 0.46
Answer
Answer: d [Reason:] Given
8. What is the system function for a two pole band pass filter that has the centre of its pass band at ω=π/2, zero its frequency response characteristic at ω=0 and at ω=π, and its magnitude response is 1/√2 at ω=4π/9?
a) 0.15(1-z-2)/(1+0.7z-2 )
b) 0.15(1+z-2)/(1-0.7z-2 )
c) 0.15(1-z-2)/(1-0.7z-2 )
d) 0.15(1+z-2)/(1+0.7z-2 )
Answer
Answer: a [Reason:] Clearly, the filter must have poles at P1,2=re±jπ/2 and zeros at z=1 and z=-1. Consequently the system function is
9. If hlp(n) denotes the impulse response of a low pass filter with frequency response Hlp(ω), then what is the frequency response of the high pass filter in terms of Hlp(ω)?
a) Hlp(ω-π/2)
b) Hlp(ω+π/2)
c) Hlp(ω-π)
d) Hlp(ω+π)
Answer
Answer: c [Reason:] The impulse response of a high pass filter is simply obtained from the impulse response of the low pass filter by changing the signs of the odd numbered samples in hlp(n). Thus
hhp(n)=(-1)n hlp(n)=(ejπ)n hlp(n)
Thus the frequency response of the high pass filter is obtained as Hlp(ω-π).
10. If the low pass filter described by the difference equation y(n)=0.9y(n-1)+0.1x(n) is converted into a high pass filter, then what is the frequency response of the high pass filter?
a) 0.1/(1+0.9ejω )
b) 0.1/(1+0.9e-jω)
c) 0.1/(1-0.9ejω )
d) None of the mentioned
Answer
Answer: b [Reason:] The difference equation for the high pass filter is
y(n)=-0.9y(n-1)+0.1x(n)
and its frequency response is given as
H(ω)= 0.1/(1+0.9e-jω).
11. A digital resonator is a special two pole band pass filter with the pair of complex conjugate poles located near the unit circle.
a) True
b) False
Answer
Answer: a [Reason:] The magnitude response of a band pass filter with two complex poles located near the unit circle is as shown below.
The filter gas a large magnitude response at the poles and hence it is called as digital resonator.
12. Which of the following filters have a frequency response as shown below?
a) Band pass filter
b) Band stop filter
c) All pass filter
d) Notch filter
Answer
Answer: d [Reason:] The given figure represents the frequency response characteristic of a notch filter with nulls at frequencies at ω0 and ω1.
13. A comb filter is a special case of notch filter in which the nulls occur periodically across the frequency band.
a) True
b) False
Answer
Answer: a [Reason:] A comb filter can be viewed as a notch filter in which the nulls occur periodically across the frequency band, hence the analogy to an ordinary comb that has periodically spaced teeth.
14. The filter with the system function H(z)=z -k is a:
a) Notch filter
b) Band pass filter
c) All pass filter
d) None of the mentioned
Answer
Answer: c [Reason:] The system with the system function given as H(z)=z -k is a pure delay system . It has a constant gain for all frequencies and hence called as All pass filter.
15. If the system has a impulse response as h(n)=Asin(n+1)ω0u(n), then the system is known as Digital frequency synthesizer.
a) True
b) False
Answer
Answer: a [Reason:] The given impulse response is h(n)=Asin(n+1)ω0u(n).
According to the above equation, the second order system with complex conjugate poles on the unit circle is a sinusoid and the system is called a digital sinusoidal oscillator or a Digital frequency synthesizer.
Digital Electronic MCQ Set 2
1. In which of the following transformations, poles and zeros of H(s) are mapped directly into poles and zeros in the z-plane?
a) Impulse invariance
b) Bilinear transformation
c) Approximation of derivatives
d) Matched Z-transform
Answer
Answer: d [Reason:] In this method of transforming analog filter into an equivalent digital filter is to map the poles and zeros of H(s) directly into poles and zeros in the z-plane.
2. Which of the following is true in matched z-transform?
a) Poles of H(s) are directly mapped to poles in z-plane
b) Zeros of H(s) are directly mapped to poles in z-plane
c) Both of the mentioned
d) None of the mentioned
Answer
Answer: c [Reason:] In the transformation of analog filter into digital filter by matched z-transform method, the poles and zeros of H(s) directly into poles and zeros in the z-plane.
3. In matched z-transform, the poles and zeros of H(s) are directly mapped into poles and zeros in z-plane.
a) True
b) False
Answer
Answer: a [Reason:] In this method of transforming analog filter into an equivalent digital filter is to map the poles and zeros of H(s) directly into poles and zeros in the z-plane.
4. The factor of the form (s-a) in H(s) is mapped into which of the following factors in z-domain?
a) 1-eaTz
b) 1-eaTz-1
c) 1-e-aTz-1
d) 1+eaTz-1
Answer
Answer: b [Reason:] If T is the sampling interval, then each factor of the form (s-a) in H(s) is mapped into the factor (1-eaTz-1) in the z-domain.
5. The factor of the form (s+a) in H(s) is mapped into which of the following factors in z-domain?
a) 1-eaTz
b) 1-eaTz-1
c) 1-e-aTz-1
d) 1+eaTz-1
Answer
Answer: c [Reason:] If T is the sampling interval, then each factor of the form (s+a) in H(s) is mapped into the factor (1-e-aTz-1) in the z-domain.
6. If the factor of the form (s-a) in H(s) is mapped into 1-eaTz-1 in the z-domain, the that kind of transformation is called as:
a) Impulse invariance
b) Bilinear transformation
c) Approximation of derivatives
d) Matched Z-transform
Answer
Answer: d [Reason:] If T is the sampling interval, then each factor of the form (s-a) in H(s) is mapped into the factor (1-eaTz-1) in the z-domain. This mapping is called the matched z-transform.
7. The poles obtained from matched z-transform are identical to poles obtained from which of the following transformations?
a) Bilinear transformation
b) Impulse invariance
c) Approximation of derivatives
d) None of the mentioned
Answer
Answer: b [Reason:] We observe that the poles obtained from the matched z-transform are identical to the poles obtained with the impulse invariance method.
8. The zero positions obtained from matched z-transform and impulse invariance method are not same.
a) True
b) False
Answer
Answer: a [Reason:] We observe that the poles obtained from the matched z-transform are identical to the poles obtained with the impulse invariance method. However, the two techniques result in different zero positions.
9. The sampling interval in the matched z-transform must be properly selected to yield the pole and zero locations at the equivalent position in the z-plane.
a) True
b) False
Answer
Answer: a [Reason:] To preserve the frequency response characteristic of the analog filter, the sampling interval in the matched z-transformation must be properly selected to yield the pole and zero locations at the equivalent position in the z-plane.
10. What should be value of sampling interval T, to avoid aliasing?
a) Zero
b) Sufficiently large
c) Sufficiently small
d) None of the mentioned
Answer
Answer: c [Reason:] Aliasing in this matched z-transformation can be avoided by selecting the sampling interval T sufficiently small.
Digital Electronic MCQ Set 3
1. There is no requirement to process the various signals at different rates commensurate with the corresponding bandwidths of the signals.
a) True
b) False
Answer
Answer: b [Reason:] In telecommunication systems that transmit and receive different types of signals, there is a requirement to process the various signals at different rates commensurate with the corresponding bandwidths of the signals.
2. What is the process of converting a signal from a given rate to a different rate?
a) Sampling
b) Normalizing
c) Sampling rate conversion
d) None of the mentioned
Answer
Answer: c [Reason:] The process of converting a signal from a given rate to a different rate is known as sampling rate conversion.
3. The systems that employ multiple sampling rates are called multi-rate DSP systems.
a) True
b) False
Answer
Answer: a [Reason:] Systems that employ multiple sampling rates in the processing of digital signals are called multi rate digital signal processing systems.
4. Which of the following methods are used in sampling rate conversion of a digital signal?
a) D/A convertor and A/D convertor
b) Performing entirely in digital domain
c) None of the mentioned
d) Both of the mentioned
Answer
Answer: d [Reason:] Sampling rate conversion of a digital signal can be accomplished in one of the two general methods. One method is to pass the signal through D/A converter, filter it if necessary, and then to resample the resulting analog signal at the desired rate. The second method is to perform the sampling rate conversion entirely in the digital domain.
5. Which of the following is the advantage of sampling rate conversion by converting the signal into analog signal?
a) Less signal distortion
b) Quantization effects
c) New sampling rate can be arbitrarily selected
d) None of the mentioned
Answer
Answer: c [Reason:] One apparent advantage of the given method is that the new sampling rate can be arbitrarily selected and need not have any special relationship with the old sampling rate.
6. Which of the following is the disadvantage of sampling rate conversion by converting the signal into analog signal?
a) Signal distortion
b) Quantization effects
c) New sampling rate can be arbitrarily selected
d) Signal distortion & Quantization effects
Answer
Answer: d [Reason:] The major disadvantage by the given type of conversion is the signal distortion introduced by the D/A converter in the signal reconstruction and by the quantization effects in the A/D conversion.
7. In which of the following, sampling rate conversion are used?
a) Narrow band filters
b) Digital filter banks
c) Quadrature mirror filters
d) All of the mentioned
Answer
Answer: d [Reason:] There are several applications of sampling rate conversion in multi rate digital signal processing systems, which include the implementation of narrow band filters, quadrature mirror filters and digital filter banks.
8. Which of the following use quadrature mirror filters?
a) Sub band coding
b) Trans-multiplexer
c) Both of the mentioned
d) None of the mentioned
Answer
Answer: c [Reason:] There are many applications where quadrature mirror filters can be used. Some of these applications are sub-band coding, trans-multiplexers and many other applications.
9. The sampling rate conversion can be as shown in the figure below.
a) True
b) False
Answer
Answer: a [Reason:] The process of sampling rate conversion in the digital domain can be viewed as a linear filtering operation as illustrated in the given figure.
10. If Fx and Fy are the sampling rates of the input and output signals respectively, then what is the value of Fy/Fx?
a) D/I
b) I/D
c) I.D
d) None of the mentioned
Answer
Answer: b [Reason:] The input signal x(n) is characterized by the sampling rate Fx and he output signal y(m) is characterized by the sampling rate Fy, then
Fy/Fx= I/D
where I and D are relatively prime integers.
11. What is the process of reducing the sampling rate by a factor D?
a) Sampling rate conversion
b) Interpolation
c) Decimation
d) None of the mentioned
Answer
Answer: c [Reason:] The process of reducing the sampling rate by a factor D, i.e., down-sampling by D is called as decimation.
12. What is the process of increasing the sampling rate by a factor I?
a) Sampling rate conversion
b) Interpolation
c) Decimation
d) None of the mentioned
Answer
Answer: b [Reason:] The process of increasing the sampling rate by a integer factor I, i.e., up-sampling by I is called as interpolation.
Digital Electronic MCQ Set 4
1. The z-transform of a signal x(n) whose definition is given by 
is known as:
a) Unilateral z-transform
b) Bilateral z-transform
c) Rational z-transform
d) None of the mentioned
Answer
Answer: a [Reason:] The z-transform of the x(n) whose definition exists in the range n=-∞ to +∞ is known as bilateral or two sided z-transform. But in the given question the value of n=0 to +∞. So, such a z-transform is known as Unilateral or one sided z-transform.
2. For what kind of signals one sided z-transform is unique?
a) All signals
b) Anti-causal signal
c) Causal signal
d) None of the mentioned
Answer
Answer: c [Reason:] One sided z-transform is unique only for causal signals, because only these signals are zero for n<0.
3. What is the one sided z-transform X+(z) of the signal x(n)={1,2,5,7,0,1}?
a) z2+2z+5+7z-1+z-3
b) 5+7z+z3
c) z-2+2z-1+5+7z+z3
d) 5+7z-1+z-3
Answer
Answer: d [Reason:] Since the one sided z-transform is valid only for n>=0, the z-transform of the given signal will be X+(z)= 5+7z-1+z-3.
4. What is the one sided z-transform of x(n)=δ(n-k)?
a) z-k
b) zk
c) 0
d) 1
Answer
Answer: a [Reason:] Since the signal x(n)= δ(n-k) is a causal signal i.e., it is defined for n>0 and x(n)=1 at z=k
So, from the definition of one sided z-transform X+(z)=z-k.
5. What is the one sided z-transform of x(n)=δ(n+k)?
a) z-k
b) 0
c) zk
d) 1
Answer
Answer: b [Reason:] Since the signal x(n)= δ(n+k) is an anti causal signal i.e., it is defined for n<0 and x(n)=1 at z= -k. Since the one sided z-transform is defined only for causal signal, in this case X+(z)=0.
6. If X+(z) is the one sided z-transform of x(n), then what is the one sided z-transform of x(n-k)?
Answer
Answer: c [Reason:] From the definition of one sided z-transform we have,
7. If x(n)=an, then what is one sided z-transform of x(n-2)?
Answer
Answer: a [Reason:]
8. If x(n)=an, then what is one sided z-transform of x(n+2)?
Answer
Answer: d [Reason:] We will apply the time advance theorem with the value of k=2.We obtain,
9. If X+(z) is the one sided z-transform of the signal x(n), then lim┬(n→∞)x(n)=lim┬(z→1)(z-1) X+ (z) is called Final value theorem.
a) True
b) False
Answer
Answer: a [Reason:] In the above theorem, we are calculating the value of x(n) at infinity, so it is called as final value theorem.
10. The impulse response of a relaxed LTI system is h(n)=anu(n),|a|<1. What is the value of the step response of the system as n→∞?
a) 1/(1+a)
b) 1/(1-a)
c) a/(1+a)
d) a/(1-a)
Answer
Answer: b [Reason:] The step response of the system is y(n)=x(n)*h(n) where x(n)=u(n)
On applying z-transform on both sides, we get
11. What is the step response of the system y(n)=ay(n-1)+x(n) -1<a<1, when the initial condition is y(-1)=1?
Answer
Answer: c [Reason:] By taking the one sided z-transform of the given equation, we obtain
Digital Electronic MCQ Set 5
1. Which of the following techniques of designing IIR filters do not involve the conversion of an analog filter into digital filter?
a) Bilinear transformation
b) Impulse invariance
c) Approximation of derivatives
d) None of the mentioned
Answer
Answer: b [Reason:] Except for the impulse invariance method, the design techniques for IIR filters involve the conversion of an analog filter into a digital filter by some mapping from the s-plane to the z-plane.
2. Using which of the following methods, a digital IIR filter can be directly designed?
a) Pade approximation
b) Least square design in time domain
c) Least square design in frequency domain
d) All of the mentioned
Answer
Answer: d [Reason:] There are several methods for designing digital filters directly. The three techniques are Pade approximation and least square method, the specifications are given in the time domain and the design is carried in time domain. The other one is least squares technique in which the design is carried out in frequency domain.
3. What is the number of parameters that a filter consists of?
a) M+N+1
b) M+N
c) M+N-1
d) M+N-2
Answer
Answer: a [Reason:] The filter has L=M+N+1 parameters, namely, the coefficients {ak} and {bk}, which can be selected to minimize some error criterion.
4. The minimization of ε involves the solution of a set of non-linear equations.
a) True
b) False
Answer
Answer: a [Reason:] In general, h(n) is a non-linear function of the filter parameters and hence the minimization of ε involves the solution of a set of non-linear equations.
5. What should be the upper limit of the solution to match h(n) perfectly to the desired response hd(n)?
a) L
b) L+1
c) L-1
d) L+2
Answer
Answer: c [Reason:] If we select the upper limit as U=L-1, it is possible to match h(n) perfectly to the desired response hd(n) for 0 < n < M+N.
6. For how many values of the impulse response, a perfect match is present between h(n) and hd(n)?
a) L
b) M+N+1
c) 2L-M-N-1
d) All of the mentioned
Answer
Answer: d [Reason:] We obtain a perfect match between h(n) and the desired response hd(n) for the first L values of the impulse response and we also know that L=M+N+1.
7. The degree to which the design technique produces acceptable filter designs depends in part on the number of filter coefficients selected.
a) True
b) False
Answer
Answer: a [Reason:] The degree to which the design technique produces acceptable filter designs depends in part on the number of filter coefficients selected. Since the design method matches hd(n) only up to the number of filter parameters, the more complex the filter, the better the approximation to hd(n).
8. According to this method of designing, the filter should have which of the following in large number?
a) Only poles
b) Both poles and zeros
c) Only zeros
d) None of the mentioned
Answer
Answer: b [Reason:] The major limitation of Pade approximation method, namely, the resulting filter must contain a large number of poles and zeros.
9. Which of the following conditions are in the favor of Pade approximation method?
a) Desired system function is rational
b) Prior knowledge of the number of poles and zeros
c) Both of the mentioned
d) None of the mentioned
Answer
Answer: c [Reason:] The Pade approximation method results in a perfect match to Hd(z) when the desired system function is rational and we have prior knowledge of the number of poles and zeros in the system.
10. Which of the following filters will have an impulse response as shown in the below figure?
a) Butterworth filters
b) Type-I chebyshev filter
c) Type-II chebyshev filter
d) None of the mentioned
Answer
Answer: a [Reason:] The diagram that is given in the question is the impulse response of Butterworth filter.
11. For what number of zeros, the approximation is poor?
a) 3
b) 4
c) 5
d) 6
Answer
Answer: a [Reason:] We observe that when the number of zeros in minimum, that is when M=3, the resulting frequency response is a relatively poor approximation to the desired response.
12. Which of the following pairs of M and N will give a perfect match?
a) 3,6
b) 3,4
c) 3,5
d) 4,5
Answer
Answer: d [Reason:] When M is increased from three to four, we obtain a perfect match with the desired Butterworth filter not only for N=4 but for N=5, and in fact, for larger values of N.
13. Which of the following filters will have an impulse response as shown in the below figure?
a) Butterworth filters
b) Type-I chebyshev filter
c) Type-II chebyshev filter
d) None of the mentioned
Answer
Answer: c [Reason:] The diagram that is given in the question is the impulse response of type-II chebyshev filter.
