Explore topic-wise InterviewSolutions in .

This section includes InterviewSolutions, each offering curated multiple-choice questions to sharpen your knowledge and support exam preparation. Choose a topic below to get started.

101.

The following frequency characteristic is for which of the following filter?(a) Type-2 Chebyshev filter(b) Type-1 Chebyshev filter(c) Butterworth filter(d) Bessel filterI have been asked this question in homework.My question is taken from Frequency Transformations in the Analog Domain in chapter Digital Filters Design of Digital Signal Processing

Answer»

The CORRECT option is (c) Butterworth filter

Explanation: The FREQUENCY characteristic given in the FIGURE is the magnitude RESPONSE of a 37-order Butterworth filter.

102.

The frequency transformation in the digital domain involves replacing the variable z^-1 by a rational function g(z^-1).(a) True(b) FalseThe question was posed to me in an interview for job.Enquiry is from Frequency Transformations in the Digital Domain topic in division Digital Filters Design of Digital Signal Processing

Answer»

Right answer is (a) True

Best explanation: As in the analog DOMAIN, frequency transformations can be performed on a digital LOW pass filter to CONVERT it to either a band pass, band STOP or HIGH pass filter. The transformation involves the replacing of the variable z^-1 by a rational function g(z^-1).

103.

Which of the following is a low pass-to-high pass transformation?(a) s→ s / Ωu(b) s→ Ωu / s(c) s→Ωu.s(d) none of the mentionedThis question was addressed to me in exam.I need to ask this question from Frequency Transformations in the Analog Domain topic in division Digital Filters Design of Digital Signal Processing

Answer»

Correct choice is (b) s→ Ωu / s

Explanation: The LOW pass-to-high pass transformation is simply achieved by replacing s by 1/s. If the desired high pass filter has the pass BAND EDGE frequency Ωu, then the transformation is

s→ Ωu / s

104.

If A=\(\frac{Ω_1 (Ω_u-Ω_l)}{-Ω_1^2+Ω_u Ω_l}\) and B=\(\frac{Ω_2 (Ω_u-Ω_l)}{-Ω_2^2+Ω_u Ω_l}\), then which of the following is the backward design equation for a low pass-to-band stop transformation?(a) ΩS=Max{|A|,|B|}(b) ΩS=Min{|A|,|B|}(c) ΩS=|B|(d) ΩS=|A|This question was addressed to me by my college professor while I was bunking the class.This key question is from Frequency Transformations in the Analog Domain topic in division Digital Filters Design of Digital Signal Processing

Answer»

The correct option is (B) ΩS=Min{|A|,|B|}

The explanation: If Ωu and Ωl are the upper and lower cutoffpass BAND frequencies of the desired band stop filter and Ω1 and Ω2 are the lower and upper cutoffstop band frequencies of the desired band stop filter, then the BACKWARD design equation is

ΩS= Min{|A|,|B|}

where, =\(\FRAC{Ω_1 (Ω_u-Ω_l)}{-Ω_1^2+Ω_u Ω_l}\) and B=\(\frac{Ω_2 (Ω_u-Ω_l)}{-Ω_2^2+Ω_u Ω_l}\).

105.

If A=\(\frac{-Ω_1^2+Ω_u Ω_l}{Ω_1 (Ω_u-Ω_l)}\) and B=\(\frac{Ω_2^2-Ω_u Ω_l}{Ω_2 (Ω_u-Ω_l)}\), then which of the following is the backward design equation for a low pass-to-band pass transformation?(a) ΩS=|B|(b) ΩS=|A|(c) ΩS=Max{|A|,|B|}(d) ΩS=Min{|A|,|B|}I have been asked this question in quiz.I'm obligated to ask this question of Frequency Transformations in the Analog Domain topic in section Digital Filters Design of Digital Signal Processing

Answer»

Right answer is (d) ΩS=Min{|A|,|B|}

Explanation: If Ωu and Ωl are the upper and LOWER cutoffpass band frequencies of the desired band pass FILTER and Ω1 and Ω2 are the lower and upper cutoffstop band frequencies of the desired band pass filter, then the BACKWARD design EQUATION is

ΩS=Min{|A|,|B|}

where, A=\(\frac{-Ω_1^2+Ω_u Ω_l}{Ω_1 (Ω_u-Ω_l)}\) and B=\(\frac{Ω_2^2-Ω_u Ω_l}{Ω_2 (Ω_u-Ω_l)}\).

106.

Which of the following filter has a phase spectrum as shown in figure?(a) Chebyshev filter(b) Butterworth filter(c) Bessel filter(d) Elliptical filterI had been asked this question during an interview for a job.The origin of the question is Frequency Transformations in the Analog Domain in chapter Digital Filters Design of Digital Signal Processing

Answer»

Correct answer is (a) CHEBYSHEV filter

Best explanation: The PHASE response given in the FIGURE belongs to the FREQUENCY characteristic of a 13-order type-1 Chebyshev filter.

107.

If H(s) is the transfer function of a analog low pass normalized filter and Ωu is the desired pass band edge frequency of new low pass filter, then which of the following transformation has to be performed?(a) s → s / Ωu(b) s → s.Ωu(c) s → Ωu/s(d) None of the mentionedI have been asked this question in an internship interview.My query is from Frequency Transformations in the Analog Domain topic in portion Digital Filters Design of Digital Signal Processing

Answer»

Correct option is (a) s → s / Ωu

The explanation is: If Ωu is the DESIRED pass BAND EDGE frequency of NEW low pass filter, then the transfer function of this new low pass filter is obtained by using the TRANSFORMATION s → s / Ωu.

108.

Which of the following is the backward design equation for a low pass-to-low pass transformation?(a) ΩS=\(\frac{Ω_S}{Ω_u}\)(b) ΩS=\(\frac{Ω_u}{Ω’_S}\)(c) Ω’S=\(\frac{Ω_S}{Ω_u}\)(d) ΩS=\(\frac{Ω’_S}{Ω_u}\)The question was posed to me in an interview.The origin of the question is Frequency Transformations in the Analog Domain topic in chapter Digital Filters Design of Digital Signal Processing

Answer»

Right answer is (d) ΩS=\(\frac{Ω’_S}{Ω_u}\)

For EXPLANATION: If Ωu is the DESIRED pass band EDGE frequency of new low pass filter, then the transfer function of this new low pass filter is obtained by using the transformation s → s / Ωu. If ΩS and Ω’S are the stop band frequencies of PROTOTYPE and transformed filters respectively, then the backward DESIGN equation is given by

ΩS=\(\frac{Ω’_S}{Ω_u}\) .

109.

Which of the following is a low pass-to-high pass transformation?(a) s → s / Ωu(b) s → Ωu / s(c) s → Ωu.s(d) none of the mentionedThis question was addressed to me in semester exam.I need to ask this question from Frequency Transformations in the Analog Domain in chapter Digital Filters Design of Digital Signal Processing

Answer» RIGHT option is (B) s → Ωu / s

Explanation: The low pass-to-high pass transformation is simply ACHIEVED by replacing s by 1/s. If the DESIRED high pass filter has the pass band edge frequency Ωu, then the transformation is

s → Ωu / s.
110.

What is the pass band edge frequency of an analog low pass normalized filter?(a) 0 rad/sec(b) 0.5 rad/sec(c) 1 rad/sec(d) 1.5 rad/secI have been asked this question during a job interview.This question is from Frequency Transformations in the Analog Domain topic in section Digital Filters Design of Digital Signal Processing

Answer»

The correct choice is (c) 1 rad/sec

To elaborate: LET H(s) denote the TRANSFER function of a low pass analog filter with a pass BAND edge frequency ΩP EQUAL to 1 rad/sec. This filter is known as analog low pass normalized PROTOTYPE.

111.

Which of the following is the backward design equation for a low pass-to-high pass transformation?(a) ΩS=\(\frac{Ω_S}{Ω_u}\)(b) ΩS=\(\frac{Ω_u}{Ω’_S}\)(c) Ω’S=\(\frac{Ω_S}{Ω_u}\)(d) ΩS=\(\frac{Ω’_S}{Ω_u}\)I got this question during a job interview.My question comes from Frequency Transformations in the Analog Domain topic in chapter Digital Filters Design of Digital Signal Processing

Answer»

Correct option is (b) ΩS=\(\frac{Ω_u}{Ω’_S}\)

To elaborate: If Ωu is the DESIRED pass band edge FREQUENCY of new high pass filter, then the TRANSFER function of this new high pass filter is obtained by using the TRANSFORMATION s→Ωu/s. If ΩS and Ω’S are the STOP band frequencies of prototype and transformed filters respectively, then the backward design equation is given by

ΩS=\(\frac{Ω_u}{Ω’_S}\) .

112.

Which of the following is true about the magnitude square response of an elliptical filter?(a) Equi-ripple in pass band(b) Equi-ripple in stop band(c) Equi-ripple in pass band and stop band(d) None of the mentionedI had been asked this question in examination.The question is from Characteristics of Commonly Used Analog Filters topic in portion Digital Filters Design of Digital Signal Processing

Answer»

Right CHOICE is (c) Equi-ripple in pass band and stop band

The explanation: An elliptical filter is a filter which exhibit equi-ripple BEHAVIOR in both pass band and stop band of the magnitude SQUARE RESPONSE.

113.

Bessel filters exhibit a linear phase response over the pass band of the filter.(a) True(b) FalseI had been asked this question in an interview for internship.The origin of the question is Characteristics of Commonly Used Analog Filters topic in chapter Digital Filters Design of Digital Signal Processing

Answer»

Correct choice is (a) True

Explanation: An IMPORTANT characteristic of the Bessel filter is the LINEAR phase RESPONSE over the pass BAND of the filter. As a CONSEQUENCE, Bessel filters has a larger transition bandwidth, but its phase is linear within the pass band.

114.

What is the order of the type-2 chebyshev filter whose magnitude square response is as shown in the following figure?(a) 2(b) 4(c) 6(d) 3This question was posed to me in class test.My question comes from Characteristics of Commonly Used Analog Filters in section Digital Filters Design of Digital Signal Processing

Answer»

Right choice is (d) 3

To ELABORATE: SINCE from the magnitude square response of the type-2 chebyshev filter, it has odd number of maxima and minima in the STOP band, the ORDER of the filter is odd i.e., 3.

115.

The frequency response shown in the figure below belongs to which of the following filters?(a) Type-1 chebyshev(b) Type-2 chebyshev(c) Butterworth(d) EllipticalI have been asked this question in quiz.I'm obligated to ask this question of Characteristics of Commonly Used Analog Filters topic in chapter Digital Filters Design of Digital Signal Processing

Answer»
116.

The zeros of type-2 class of chebyshev filters lies on ___________(a) Imaginary axis(b) Real axis(c) Zero(d) Cannot be determinedThis question was posed to me in a job interview.Question is from Characteristics of Commonly Used Analog Filters in portion Digital Filters Design of Digital Signal Processing

Answer»

The correct choice is (a) Imaginary AXIS

To EXPLAIN I WOULD SAY: The ZEROS of this class of filters lie on the imaginary axis in the s-plane.

117.

Which of the following is false about the type-2 chebyshev filters?(a) Monotonic behavior in the pass band(b) Equi-ripple behavior in the stop band(c) Zero behavior(d) Monotonic behavior in the stop bandI got this question during an online interview.I'm obligated to ask this question of Characteristics of Commonly Used Analog Filters topic in section Digital Filters Design of Digital Signal Processing

Answer»

The CORRECT answer is (d) MONOTONIC BEHAVIOR in the STOP band

The best explanation: Type-2 chebyshev filters exhibit equi-ripple behavior in stop band and a monotonic characteristic in the pass band.

118.

Which of the following is true about type-1 chebyshev filter?(a) Equi-ripple behavior in pass band(b) Monotonic characteristic in stop band(c) Equi-ripple behavior in pass band & Monotonic characteristic in stop band(d) None of the mentionedI had been asked this question in an interview for internship.I want to ask this question from Characteristics of Commonly Used Analog Filters topic in section Digital Filters Design of Digital Signal Processing

Answer»
119.

Type-2 chebyshev filters consists of ______________(a) Only poles(b) Both poles and zeros(c) Only zeros(d) Cannot be determinedThis question was posed to me in a job interview.Query is from Characteristics of Commonly Used Analog Filters in chapter Digital Filters Design of Digital Signal Processing

Answer»

The CORRECT option is (B) Both poles and ZEROS

To explain: Type-1 chebyshev filters are all-pole filters where as the family of type-2 chebyshev filters CONTAINS both poles and zeros.

120.

What is the order of a low pass Butterworth filter that has a -3dB bandwidth of 500Hz and an attenuation of 40dB at 1000Hz?(a) 4(b) 5(c) 6(d) 7This question was posed to me in an interview for internship.Asked question is from Characteristics of Commonly Used Analog Filters in portion Digital Filters Design of Digital Signal Processing

Answer»

The CORRECT option is (d) 7

To explain: Given Ωc=1000π and Ωs=2000π

For an attenuation of 40DB, δ2=0.01. We know that

N=\(\frac{log⁡[(\frac{1}{δ_2^2})-1]}{2log⁡[\frac{Ω_s}{Ω_s}]}\)

THUS by substituting the corresponding values in the above equation, we get N=6.64

To MEET the desired SPECIFICATIONS, we select N=7.

121.

The magnitude square response shown in the below figure is for which of the following given filters?(a) Butterworth(b) Chebyshev(c) Elliptical(d) None of the mentionedI have been asked this question during a job interview.Query is from Characteristics of Commonly Used Analog Filters in chapter Digital Filters Design of Digital Signal Processing

Answer»

The CORRECT option is (a) BUTTERWORTH

To explain I would say: The MAGNITUDE square response shown in the given FIGURE is for Butterworth FILTER.

122.

Which of the following is the band edge value of |H(Ω)|^2?(a) (1+ε^2)(b) (1-ε^2)(c) 1/(1+ε^2)(d) 1/(1-ε^2)I have been asked this question during an online exam.This is a very interesting question from Characteristics of Commonly Used Analog Filters in chapter Digital Filters Design of Digital Signal Processing

Answer»

The correct option is (c) 1/(1+ε^2)

To explain: 1/(1+ε^2) gives the BAND edge VALUE of the magnitude square RESPONSE |H(Ω)|^2.

123.

What is the transfer function of magnitude squared frequency response of the normalized low pass Butterworth filter?(a) \(\frac{1}{1+(s/j)^{-2N}}\)(b) \(1+(\frac{s}{j})^{-2N}\)(c) \(1+(\frac{s}{j})^{2N}\)(d) \(\frac{1}{1+(\frac{s}{j})^{2N}}\)I have been asked this question by my college professor while I was bunking the class.My enquiry is from Characteristics of Commonly Used Analog Filters topic in portion Digital Filters Design of Digital Signal Processing

Answer»

Correct CHOICE is (d) \(\frac{1}{1+(\frac{s}{j})^{2N}}\)

For explanation I would say: We know that the magnitude squared frequency response of a normalized low PASS Butterworth filter is given as

|H(jΩ)|^2=\(\frac{1}{1+Ω^{2N}}\) => HN(jΩ).HN(-jΩ)=\(\frac{1}{1+Ω^{2N}}\)

REPLACING jΩ by ‘s’ and hence Ω by s/j in the above equation, we get

HN(s).HN(-s)=\(\frac{1}{1+(\frac{s}{j})^{2N}}\) which is called the transfer function.

124.

Low pass Butterworth filters are also called as _____________(a) All-zero filter(b) All-pole filter(c) Pole-zero filter(d) None of the mentionedThe question was asked during an interview.The query is from Characteristics of Commonly Used Analog Filters topic in chapter Digital Filters Design of Digital Signal Processing

Answer»

Right option is (b) All-pole filter

Easiest explanation: LOW pass BUTTERWORTH filters are also CALLED as all-pole filters because it has only non-zero POLES.

125.

What is the equation for magnitude square response of a low pass Butterworth filter?(a) \(\frac{1}{\sqrt{1+(\frac{Ω}{Ω_C})^{2N}}}\)(b) \(1+(\frac{Ω}{Ω_C})^{2N}\)(c) \(\sqrt{1+(\frac{Ω}{Ω_C})^{2N}}\)(d) None of the mentionedThis question was posed to me in an international level competition.My question is from Characteristics of Commonly Used Analog Filters topic in section Digital Filters Design of Digital Signal Processing

Answer»

Right ANSWER is (a) \(\FRAC{1}{\sqrt{1+(\frac{Ω}{Ω_C})^{2N}}}\)

The explanation is: A Butterworth is CHARACTERIZED by the magnitude FREQUENCY response

|H(jΩ)| = \(\frac{1}{\sqrt{1+(\frac{Ω}{Ω_C})^{2N}}}\)

where N is the order of the filter and ΩC is defined as the cutoff frequency.

126.

What should be value of sampling interval T, to avoid aliasing?(a) Zero(b) Sufficiently large(c) Sufficiently small(d) None of the mentionedThis question was posed to me during an interview.The above asked question is from Matched Z Transformation in division Digital Filters Design of Digital Signal Processing

Answer»

Correct ANSWER is (c) Sufficiently small

For EXPLANATION: Aliasing in this MATCHED z-transformation can be avoided by selecting the sampling INTERVAL T sufficiently small.

127.

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 mentionedI had been asked this question in a job interview.I'd like to ask this question from Matched Z Transformation topic in portion Digital Filters Design of Digital Signal Processing

Answer»

The correct choice is (b) IMPULSE INVARIANCE

The best I can EXPLAIN: We observe that the poles OBTAINED from the matched z-transform are identical to the poles obtained with the impulse invariance method.

128.

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) FalseI have been asked this question in an international level competition.This is a very interesting question from Matched Z Transformation topic in chapter Digital Filters Design of Digital Signal Processing

Answer»

Right option is (a) True

Explanation: 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.

129.

The zero positions obtained from matched z-transform and impulse invariance method are not same.(a) True(b) FalseThe question was posed to me in my homework.The doubt is from Matched Z Transformation topic in portion Digital Filters Design of Digital Signal Processing

Answer»
130.

If the factor of the form (s-a) in H(s) is mapped into 1-e^aTz^-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-transformI have been asked this question in an interview for internship.This interesting question is from Matched Z Transformation in division Digital Filters Design of Digital Signal Processing

Answer» RIGHT ANSWER is (d) Matched Z-transform

The explanation: 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. This mapping is CALLED the matched z-transform.
131.

The factor of the form (s+a) in H(s) is mapped into which of the following factors in z-domain?(a) 1-e^aTz(b) 1-e^aTz^-1(c) 1-e^-aTz^-1(d) 1+e^aTz^-1The question was posed to me in a national level competition.I would like to ask this question from Matched Z Transformation in division Digital Filters Design of Digital Signal Processing

Answer» RIGHT answer is (C) 1-e^-aTz^-1

Easiest explanation: 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.
132.

The factor of the form (s-a) in H(s) is mapped into which of the following factors in z-domain?(a) 1-e^aTz(b) 1-e^aTz^-1(c) 1-e^-aTz^-1(d) 1+e^aTz^-1This question was posed to me by my college professor while I was bunking the class.I would like to ask this question from Matched Z Transformation topic in portion Digital Filters Design of Digital Signal Processing

Answer»

The correct CHOICE is (b) 1-e^aTz^-1

To elaborate: 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.

133.

In matched z-transform, the poles and zeros of H(s) are directly mapped into poles and zeros in z-plane.(a) True(b) FalseI had been asked this question by my college director while I was bunking the class.My doubt is from Matched Z Transformation in portion Digital Filters Design of Digital Signal Processing

Answer»

Right answer is (a) True

To ELABORATE: 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.

134.

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) Poles & Zeros of H(s) are directly mapped to poles in z-plane(d) None of the mentionedThe question was posed to me during an online exam.My question is taken from Matched Z Transformation topic in section Digital Filters Design of Digital Signal Processing

Answer» CORRECT option is (c) Poles & Zeros of H(s) are DIRECTLY MAPPED to poles in z-plane

To explain: 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.
135.

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-transformI got this question at a job interview.I need to ask this question from Matched Z Transformation topic in division Digital Filters Design of Digital Signal Processing

Answer»

The correct ANSWER is (d) Matched Z-transform

For explanation: 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.

136.

When σ>0, then what is the condition on ‘r’?(a) 0

Answer»

The correct choice is (c) R>1

Best explanation: We know that z=e^sT

Now substitute s=σ+jΩ and z=r.e^jω, that is represent ‘z’ in the POLAR form

On EQUATING both SIDES, we get

r=e^σT

Thus when σ>0, the value of ‘r’ varies from r>1.

137.

What is the period of the scaled spectrum Fs.X(F)?(a) 2Fs(b) Fs/2(c) 4Fs(d) FsThe question was asked during an interview.My query is from IIR Filter Design by Impulse Invariance in division Digital Filters Design of Digital Signal Processing

Answer»

The correct ANSWER is (d) Fs

For explanation: When a continuous time signal x(t) with spectrum X(F) is SAMPLED at a rate Fs=1/T samples PER SECOND, the spectrum of the sampled signal is periodic repetition of the scaled spectrum Fs.X(F) with period Fs.

138.

When σ

Answer»

The correct choice is (a) 0
Explanation: We know that z=e^sT

Now SUBSTITUTE s=σ+jΩ and z=r.e^jω, that is represent ‘z’ in the polar form

On EQUATING both sides, we get

r=e^σT

Thus when σ<0, the value of ‘r’ VARIES from 0

139.

Which of the following is the correct relation between ω and Ω?(a) Ω=ωT(b) T=Ωω(c) ω=ΩT(d) None of the mentionedThe question was asked during an internship interview.Enquiry is from IIR Filter Design by Impulse Invariance in chapter Digital Filters Design of Digital Signal Processing

Answer»

The CORRECT option is (C) ω=ΩT

The best I can explain: We know that z=e^sT

Now SUBSTITUTE s=σ+jΩ and z=r.e^jω, that is REPRESENT ‘z’ in the polar form

On equating both SIDES, we get

ω=ΩT.

140.

Which of the following filters cannot be designed using impulse invariance method?(a) Low pass(b) Band pass(c) Low and band pass(d) High passI got this question in homework.This question is from IIR Filter Design by Impulse Invariance topic in portion Digital Filters Design of Digital Signal Processing

Answer»

The correct answer is (d) High pass

To EXPLAIN: It is clear that the impulse INVARIANCE method is in -appropriate for DESIGNING high pass filter due to the SPECTRUM ALIASING that results from the sampling process.

141.

Sampling interval T is selected sufficiently large to completely avoid or at least minimize the effects of aliasing.(a) True(b) FalseI got this question in an interview.This question is from IIR Filter Design by Impulse Invariance topic in division Digital Filters Design of Digital Signal Processing

Answer»

Correct choice is (B) False

Easy explanation: The digital filter with FREQUENCY response H(ω) has the frequency response CHARACTERISTICS of the corresponding analog filter if the sampling interval T is SELECTED SUFFICIENTLY small to completely avoid or at least minimize the effects of aliasing.

142.

The frequency response given in the above question is for a low pass digital filter.(a) True(b) FalseThe question was asked in exam.I'm obligated to ask this question of IIR Filter Design by Impulse Invariance topic in portion Digital Filters Design of Digital Signal Processing

Answer»

Correct answer is (a) True

Best explanation: The above given frequency RESPONSE depicts the frequency response of a LOW pass ANALOG filter and the frequency response of the corresponding DIGITAL filter.

143.

Which of the filters have a frequency response as shown in the figure below?(a) Analog filter(b) Digital filter without aliasing(c) Digital filter with aliasing(d) None of the mentionedThe question was asked during an interview.The doubt is from IIR Filter Design by Impulse Invariance topic in division Digital Filters Design of Digital Signal Processing

Answer»

The correct choice is (c) Digital filter with aliasing

To explain: In the given diagram, the CONTINUOUS line is the FREQUENCY response of ANALOG filter and dotted line is the frequency response of the CORRESPONDING digital filter with aliasing.

144.

Aliasing occurs if the sampling rate Fs is more than twice the highest frequency contained in X(F).(a) True(b) FalseThe question was asked by my college director while I was bunking the class.I'd like to ask this question from IIR Filter Design by Impulse Invariance topic in portion Digital Filters Design of Digital Signal Processing

Answer»

The correct choice is (b) False

The EXPLANATION is: Aliasing occurs if the sampling rate FS is less than twice the highest FREQUENCY contained in X(F).

145.

What is the equation for normalized frequency?(a) F/Fs(b) F.Fs(c) Fs/F(d) None of the mentionedI had been asked this question during an online interview.Question is taken from IIR Filter Design by Impulse Invariance in portion Digital Filters Design of Digital Signal Processing

Answer»

The CORRECT option is (a) F/Fs

Explanation: In the impulse invariance method, the normalized FREQUENCY f is GIVEN by

f= F/Fs.

146.

If a continuous time signal x(t) with spectrum X(F) is sampled at a rate Fs=1/T samples per second, then what is the scaled spectrum?(a) X(F)(b) Fs.X(F)(c) X(F)/Fs(d) None of the mentionedThis question was posed to me in unit test.Question is from IIR Filter Design by Impulse Invariance in chapter Digital Filters Design of Digital Signal Processing

Answer»

Correct option is (b) Fs.X(F)

For explanation: When a continuous time signal x(t) with spectrum X(F) is sampled at a rate Fs=1/T samples PER second, the spectrum of the sampled signal is PERIODIC REPETITION of the scaled spectrum Fs.X(F).

147.

When σ=0, then what is the condition on ‘r’?(a) 0

Answer» RIGHT answer is (b) R=1

To EXPLAIN I would say: We know that z=e^sT

Now substitute s=σ+jΩ and z=r.e^jω, that is REPRESENT ‘z’ in the polar form

On EQUATING both sides, we get

r=e^σT

Thus when σ=0, the value of ‘r’ varies from r=1.
148.

If a continuous time signal x(t) with spectrum X(F) is sampled at a rate Fs=1/T samples per second, the spectrum of the sampled signal is _____________(a) Non periodic repetition(b) Non periodic non-repetition(c) Periodic repetition(d) None of the mentionedThis question was addressed to me in a national level competition.This question is from IIR Filter Design by Impulse Invariance in chapter Digital Filters Design of Digital Signal Processing

Answer»

Right ANSWER is (C) Periodic repetition

Best EXPLANATION: When a continuous TIME signal x(t) with SPECTRUM X(F) is sampled at a rate Fs=1/T samples per second, the spectrum of the sampled signal is periodic repetition.

149.

By impulse invariance method, the IIR filter will have a unit sample response h(n) that is the sampled version of the analog filter.(a) True(b) FalseI have been asked this question in unit test.The doubt is from IIR Filter Design by Impulse Invariance in portion Digital Filters Design of Digital Signal Processing

Answer»

The CORRECT choice is (a) True

The best I can explain: In the impulse invariance method, our objective is to design an IIR filter having a unit sample response h(n) that is the sampled VERSION of the impulse response of the ANALOG filter. That is

h(n)=h(nT); n=0,1,2…

where T is the SAMPLING interval.

150.

It is possible to map the jΩ-axis into the unit circle.(a) True(b) FalseThis question was posed to me during an online interview.This intriguing question originated from IIR Filter Design by Approximation of Derivatives in portion Digital Filters Design of Digital Signal Processing

Answer»

Correct choice is (a) True

The explanation is: By proper choice of the COEFFICIENTS of {αk}, it is POSSIBLE to MAP the jΩ-axis into the unit CIRCLE.