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51.

Which of the following is the right way to reduce distortion in the DM?(a) By setting up an integrator in front of DM(b) By setting up an integrator behind the DM(c) By setting up an integrator in the middle of DM(d) None of the mentionedThis question was posed to me during an online interview.This interesting question is from Sample and Hold in chapter Sampling and Reconstruction of Signals of Digital Signal Processing

Answer» CORRECT choice is (a) By SETTING up an integrator in FRONT of DM

To explain: We note that increasing Δ REDUCES overload distortion but increases the granular noise, and vice versa. One way to reduce these two types of distortion is to USE an integrator in front of the DM.
52.

In DM, By increasing Δ, reduces the overload distortion but increases the granular noise, and vice versa.(a) True(b) FalseI had been asked this question in a job interview.Origin of the question is Sample and Hold topic in chapter Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

The correct choice is (a) True

Easiest explanation: The GRANULAR noise occurs when the DM tracks a relatively flat (SLOWLY changing) input SIGNAL. We note that increasing Δ reduces overload DISTORTION but increases the granular noise, and vice VERSA.

53.

The slope-overload distortion is avoided, if which of the following conditions satisfy?(a) Min|dx(t)/d(t)| ≤ Δ/T(b) Max|dx(t)/d(t)| ≤ Δ/T(c) |dx(t)/d(t)| ≤Δ/T(d) None of the mentionedI have been asked this question during a job interview.This is a very interesting question from Sample and Hold topic in section Sampling and Reconstruction of Signals of Digital Signal Processing

Answer» CORRECT OPTION is (b) Max|dx(t)/d(t)| ≤ Δ/T

Easy explanation: The crosshatched areas illustrate two types of quantization error in DM, SLOPE-overload distortion and granular noise. types of quantization error in DM, slope-overload distortion and granular noise. Since the maximum slope A (T in x (n) is limited by the step size, slope-overload distortion can be AVOIDED if max|dx(t)/d(t)|≤Δ/T).
54.

The crosshatched areas gives two types of Quantization error in DM, they are?(a) Slope-overload distortion(b) Granular noise(c) Slope-overload distortion & Granular noise(d) None of the mentionedI got this question during an online exam.My question is based upon Sample and Hold topic in portion Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

The correct ANSWER is (c) Slope-overload distortion & Granular noise

The best EXPLANATION: The crosshatched areas ILLUSTRATE two types of quantization error in DM, slope-overload distortion and granular noise.

55.

If the interpolation factor is I = 256, the A/D converter output can be obtained by averaging successive non-overlapping blocks of 128 bits.(a) True(b) FalseThis question was posed to me in an international level competition.My doubt is from Sample and Hold topic in division Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

Right choice is (a) True

For explanation: If the INTERPOLATION factor is I = 256, the A/D converter output can be obtained by averaging SUCCESSIVE non-overlapping blocks of 128 bits. This averaging WOULD RESULT in a digital signal with a range of values from zero to 256 (b as 8 bits) at the Nyquist RATE. The averaging process also provides the required anti-aliasing filtering.

56.

The noise power σn^2 can be reduced by increasing the sampling rate to spread the quantization noise power over a larger frequency band (-Fs/2, Fs/2).(a) True(b) FalseI got this question by my school teacher while I was bunking the class.I want to ask this question from Sample and Hold topic in portion Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

Correct option is (a) True

To EXPLAIN: The noise power σn^2 can be REDUCED by increasing the sampling rate to SPREAD the quantization noise power over a larger frequency band (-Fs/2, Fs/2), and then shaping the noise power spectral density by MEANS o F an appropriate filter.

57.

In the absence of an S/H, the input signal must change by more than one-half of the quantization step during the conversion, which may be an impractical constraint.(a) True(b) FalseThe question was posed to me in an interview.My doubt stems from Sample and Hold in portion Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

Correct answer is (b) False

To elaborate: The use of an S/H allows the A /D converter to OPERATE more slowly compared to the TIME actually used to acquire the sample. In the absence of an S/H, the input signal must not change by more than one-half of the QUANTIZATION STEP during the conversion, which may be an IMPRACTICAL constraint.

58.

What is the process of down sampling called?(a) Decimation(b) Fornication(c) Both Decimation & Fornication(d) None of the mentionedI have been asked this question during an online interview.This question is from Sample and Hold topic in chapter Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»
59.

In the practical A/D converters, what are the distortions and time-related degradations occur during the conversion process?(a) Jitter errors(b) Droops(c) Nonlinear variations in the duration of the sampling aperture(d) All of the mentionedI have been asked this question in an international level competition.I'm obligated to ask this question of Sample and Hold topic in section Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

The correct CHOICE is (d) All of the mentioned

For explanation I would say: An ideal S/H introduces no distortion in the conversion process and is accurately modeled as an ideal SAMPLER. However, time-related degradations such as ERRORS in the PERIODICITY of the sampling process (“jitter”), nonlinear variations in the duration of the sampling aperture, and changes in the voltage held during conversion (“droop”) do occur in PRACTICAL devices.

60.

In A/D converter, what is the time relation between sampling period T and the duration of the sample mode and the hold mode?(a) Should be larger than the duration of sample mode and hold mode(b) Should be smaller than the duration of sample mode and hold mode(c) Should be equal to the duration of sample mode and hold mode(d) Should be larger than or equals to the duration of sample mode and hold modeThis question was posed to me in an interview for internship.The above asked question is from Sample and Hold in chapter Sampling and Reconstruction of Signals of Digital Signal Processing

Answer» RIGHT OPTION is (a) Should be larger than the duration of sample MODE and hold mode

To explain: The A/D converter begins the conversion after it RECEIVES a convert command. The SAMPLING period T should be larger than the duration of the sample mode and the hold mode.
61.

The time required to complete the conversion of Analog to Digital is ________ the duration of the hold mode of S/H.(a) Greater than(b) Equals to(c) Less than(d) Greater than or Equals toThis question was addressed to me in an online quiz.My doubt is from Sample and Hold in division Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

Right answer is (C) LESS than

Best explanation: The A/D converter begins the conversion after it receives a convert command. The TIME required to complete the conversion should be less than the DURATION of the HOLD mode of S/H.

62.

The S/H is a digitally controlled analog circuit that tracks the analog input signal during the sample mode and then holds it fixed during the hold mode to the instantaneous value of the signal at the time the system is switched from the sample to the hold mode.(a) True(b) FalseI had been asked this question during an online exam.My doubt stems from Sample and Hold topic in portion Sampling and Reconstruction of Signals of Digital Signal Processing

Answer» RIGHT choice is (a) True

Easiest explanation: The sampling of an analog signal is performed by a sample-and-HOLD (S/H) circuit. The sampled signal is then quantized and converted to digital form. Usually, the S/H is INTEGRATED into the (A/D) converter. The S/H is a digitally controlled analog circuit that tracks the analog INPUT signal during the sample MODE and then holds it fixed during the hold mode to the instantaneous value of the signal at the time the system is switched from the sample mode to the hold mode.
63.

In the equation xq(n)=axq(n-1)+dq(n), if a < 1 then integrator is called?(a) Leaky integrator(b) Ideal integrator(c) Ideal accumulator(d) Both Ideal integrator & accumulatorThe question was posed to me during an interview for a job.This key question is from Oversampling A/D Converters topic in chapter Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

The correct answer is (a) LEAKY INTEGRATOR

Easy explanation: In the equation xq(N)=axq(n-1)+ dq(n), a < 1 results in a ”leaky integrator”.

64.

What is the main function of (D/A) or DAC converter?(a) Converts Digital to Analog Signal(b) Converts Analog to Digital signal(c) All of the mentioned(d) None of the mentionedThe question was asked in my homework.The above asked question is from Sample and Hold topic in division Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

The correct choice is (a) Converts Digital to Analog Signal

To EXPLAIN I would SAY: A digital-to-analog (D/A) converter (DAC) TAKES a digital sequence and PRODUCES at its output a voltage or current proportional to the size of the digital word applied to its input.

65.

What is the main function of (A/D) or ADC converter?(a) Converts Digital to Analog Signal(b) Converts Analog to Digital signal(c) All of the mentioned(d) None of the mentionedI have been asked this question at a job interview.Query is from Sample and Hold topic in chapter Sampling and Reconstruction of Signals of Digital Signal Processing

Answer» RIGHT CHOICE is (b) Converts Analog to DIGITAL signal

Explanation: The electronic DEVICE that performs this conversion from an analog signal to a digital sequence is called an analog-to-digital (A/D) converter (ADC).
66.

In the equation xq(n)=axq(n-1)+dq(n), if a = 1 then integrator is called?(a) Leaky integrator(b) Ideal integrator(c) Ideal accumulator(d) Both Ideal integrator & accumulatorThe question was posed to me at a job interview.My doubt stems from Oversampling A/D Converters topic in division Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

The CORRECT OPTION is (d) Both Ideal integrator & accumulator

Best explanation: In the equation xq(n)=axq(n-1)+dq(n), if a = 1, we have an ideal accumulator (integrator).

67.

In DM, What is the order of predictor is used?(a) Zero-order predictor(b) Second-order predictor(c) First-order predictor(d) Third-order predictorI had been asked this question in unit test.The above asked question is from Oversampling A/D Converters in portion Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

Right CHOICE is (c) First-order PREDICTOR

Explanation: In DM, the QUANTIZER is a simple 1-bit (two-level) quantizer and the predictor is a first-order predictor.

68.

What is the abbreviation of DM?(a) Diameter Modulation(b) Distance Modulation(c) Delta Modulation(d) None of the mentionedThis question was posed to me in an internship interview.I'd like to ask this question from Oversampling A/D Converters topic in portion Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

Right answer is (c) DELTA Modulation

The explanation is: The simplest FORM of differential PREDICTIVE QUANTIZATION is called delta modulation (DM).

69.

In DM, the quantizer is a simple ________ bit and______ level quantizer.(a) 2-bit, one-level(b) 1-bit, two-level(c) 2-bit, two level(d) 1-bit, one levelThis question was addressed to me in unit test.This intriguing question comes from Oversampling A/D Converters in portion Sampling and Reconstruction of Signals of Digital Signal Processing

Answer» CORRECT option is (B) 1-bit, two-level

Explanation: The simplest form of differential predictive quantization is CALLED delta modulation (DM). In DM, the quantizer is a SIMPLE 1-bit (two-level) quantizer.
70.

The simplest form of differential predictive quantization is called?(a) AM(b) BM(c) DM(d) None of the mentionedThe question was posed to me in exam.I'd like to ask this question from Oversampling A/D Converters in chapter Sampling and Reconstruction of Signals of Digital Signal Processing

Answer» CORRECT answer is (C) DM

The explanation: The SIMPLEST form of differential predictive quantization is called delta modulation (DM).
71.

To reduce the dynamic range of the difference signal d(n) = x(n) – \(\hat{x}(n)\), thus a predictor of order p has the form?(a) \(\hat{x}(n)=\sum_{k=1}^pa_k x(n+k)\)(b) \(\hat{x}(n)=\sum_{k=1}^pa_k x(n-k)\)(c) \(\hat{x}(n)=\sum_{k=0}^pa_k x(n+k)\)(d) \(\hat{x}(n)=\sum_{k=0}^pa_k x(n-k)\)This question was addressed to me in an international level competition.My enquiry is from Oversampling A/D Converters in section Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

The CORRECT answer is (b) \(\hat{x}(n)=\sum_{k=1}^pa_k x(n-k)\)

Best explanation: The goal of the predictor is to provide an estimate \(\hat{x}(n)\) of x(n) from a linear COMBINATION of past values of x(n), so as to reduce the dynamic RANGE of the DIFFERENCE signal d(n) = x(n)-\(\hat{x}(n)\). Thus a predictor of ORDER p has the form\(\hat{x}(n)=\sum_{k=1}^pa_k x(n-k)\).

72.

What is the quantity ax(n-1) is called?(a) Second-order predictor of x(n)(b) Zero-order predictor of x(n)(c) First-order predictor of x(n)(d) Third-order predictor of x(n)The question was asked in exam.My question comes from Oversampling A/D Converters in portion Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

Correct answer is (C) First-order predictor of x(N)

The EXPLANATION: In the equation d(n) = x(n)–AX(n-1), the quantity ax(n-1) is called a First-order predictor of x(n).

73.

What are the main uses of DPCM?(a) Speech Decoding and Transmission over mobiles(b) Speech Encoding and Transmission over mobiles(c) Speech Decoding and Transmission over telephone channels(d) Speech Encoding and Transmission over telephone channelsI got this question in semester exam.Origin of the question is Oversampling A/D Converters topic in chapter Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

Correct choice is (d) SPEECH Encoding and TRANSMISSION over telephone channels

The best explanation: A differential PREDICTIVE signal quantizer system. This system is used in speech encoding and transmission over telephone channels and is known as differential pulse CODE modulation (DPCM ).

74.

What is the expansion of DPCM?(a) Differential Pulse Code Modulation(b) Differential Plus Code Modulation(c) Different Pulse Code Modulation(d) None of the mentionedI got this question during an online interview.The query is from Oversampling A/D Converters in division Sampling and Reconstruction of Signals of Digital Signal Processing

Answer» RIGHT option is (a) Differential Pulse CODE Modulation

To elaborate: A differential predictive SIGNAL quantizer system. This system is USED in speech ENCODING and transmission over telephone channels and is known as differential pulse code modulation (DPCM ).
75.

The differential predictive signal quantizer system is known as?(a) DCPM(b) DMPC(c) DPCM(d) None of the mentionedThe question was asked in an interview for job.This key question is from Oversampling A/D Converters topic in division Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

Correct OPTION is (c) DPCM

To explain: A DIFFERENTIAL PREDICTIVE SIGNAL quantizer system. This system is used in speech encoding and transmission over telephone channels and is known as differential pulse code modulation (DPCM).

76.

If the difference d(n) = x(n)–ax(n-1), then what is the optimum choice for a = ?(a) \({γ_{xx} (1)}{σ_x^2}\)(b) \({γ_{xx} (0)}{σ_x^2}\)(c) \({γ_{xx} (0)}{σ_d^2}\)(d) \({γ_{xx} (1)}{σ_d^2}\)I have been asked this question in a job interview.Query is from Oversampling A/D Converters in division Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

The correct option is (a) \({γ_{XX} (1)}{σ_x^2}\)

EASIEST explanation: An even better approach is to quantize the difference, d(n) = x(n)–ax(n-1), W here a is a parameter selected to minimize the variance in d(n). This leads to the result that the OPTIMUM choice of a is \({γ_{xx} (1)}{γ_{xx} (0)} = {γ_{xx} (1)}{σ_x^2}\).

77.

What is the variance of the difference between two successive signal samples, d(n) = x(n)–ax(n-1)?(a) \(σ_d^2=2σ_x^2 [1-a^2]\)(b) \(σ_d^2=σ_x^2 [1+a^2]\)(c) \(σ_d^2=σ_x^2 [1-a^2]\)(d) \(σ_d^2=2σ_x^2 [1+a^2]\)The question was posed to me during an online interview.Enquiry is from Oversampling A/D Converters in chapter Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»
78.

What is the variance of the difference between two successive signal samples, d(n) = x(n) – x(n-1)?(a) \(σ_d^2=2σ_x^2 [1+γ_{xx} (1)]\)(b) \(σ_d^2=2σ_x^2 [1-γ_{xx} (1)]\)(c) \(σ_d^2=4σ_x^2 [1-γ_{xx} (1)]\)(d) \(σ_d^2=3σ_x^2 [1-γ_{xx} (1)]\)This question was posed to me in an interview for internship.This is a very interesting question from Oversampling A/D Converters in section Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

The correct answer is (b) \(σ_d^2=2σ_x^2 [1-γ_{XX} (1)]\)

Explanation: \(σ_d^2=E[d^2 (n)] = E{[X(n)- x(n-1)]^2}\)

= \(E [x^2 (n)]-2E{x(n)x(n-1)}+E[x^2 (n-1)]\)

= \(2σ_x^2 [1+γ_{xx} (1)]\).

79.

For a given number of bits, the power of quantization noise is proportional to the variance of the signal to be quantized.(a) True(b) FalseI have been asked this question during an online exam.I would like to ask this question from Oversampling A/D Converters topic in section Sampling and Reconstruction of Signals of Digital Signal Processing

Answer»

The CORRECT CHOICE is (a) True

Easy explanation: The dynamic range of the signal, which is proportional to its standard DEVIATION σx, should match the range R of the quantizer, it follows that ∆ is proportional to σx. Hence for a given number of bits, the POWER of the quantization NOISE is proportional to the variance of the signal to be quantized.