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51.

Which method is preferred in magnetic tape recording?(a) NRZ-L(b) NRZ-M(c) NRZ-S(d) None of the mentionedI had been asked this question during an online interview.I'd like to ask this question from Baseband Transmission in division Formatting and Baseband Modulation of Digital Communications

Answer»

The correct CHOICE is (b) NRZ-M

Easy explanation: NRZ-M is also called as DIFFERENTIAL ENCODING and it is most preferred in magentic TAPE RECORDING.

52.

Which method is called as differential encoding?(a) NRZ-L(b) NRZ-M(c) NRZ-S(d) None of the mentionedI got this question in an online quiz.The query is from Baseband Transmission topic in portion Formatting and Baseband Modulation of Digital Communications

Answer»

Right choice is (b) NRZ-M

For EXPLANATION: In NRZ-M, logic 1 is represented by a change in VOLTAGE level and logic 0 is represented by no change in level. This is called as differential encoding.

53.

Which type is used and preferred in digital logic circuits?(a) NRZ-L(b) NRZ-M(c) NRZ-S(d) None of the mentionedThis question was posed to me in an interview for internship.Question is taken from Baseband Transmission topic in chapter Formatting and Baseband Modulation of Digital Communications

Answer»

Right option is (a) NRZ-L

For explanation: NRZ-L is EXTENSIVELY used in DIGITAL LOGIC circuits. In this method, logic 1 is REPRESENTED by ONE voltage level and logic 0 is represented by another voltage level.

54.

When pulse code modulation is applied to non binary symbols we obtain waveform called as(a) PCM(b) PAM(c) M-ary(d) line codesThe question was asked in my homework.The query is from Baseband Transmission topic in chapter Formatting and Baseband Modulation of Digital Communications

Answer»

Right OPTION is (c) M-ary

The EXPLANATION: When pulse code modulation is applied to binary symbols we get PCM waveforms and when it is applied to non binary symbols we obtain M-ary waveforms.

55.

Examples of PCM waveforms are(a) Non return to zero(b) Phase encoded(c) Multilevel binary(d) All of the mentionedThe question was posed to me in final exam.This intriguing question originated from Baseband Transmission topic in section Formatting and Baseband Modulation of Digital Communications

Answer»

The correct answer is (d) All of the mentioned

For explanation: Some of the examples or classification of pulse code modulated SIGNALS are non RETURN to ZERO, return to zero, phase encoded, multilevel binary ETC.

56.

Which waveforms are also called as line codes?(a) PCM(b) PAM(c) FM(d) AMThe question was posed to me in homework.Query is from Baseband Transmission in portion Formatting and Baseband Modulation of Digital Communications

Answer»

Correct ANSWER is (a) PCM

Best explanation: When pulse modulation is applied to binary symbol we OBTAIN pulse CODE modulated waveforms. These waveforms are also called as line codes.

57.

The standard value of A in A-law is(a) 87(b) 88(c) 86.7(d) 87.6The question was posed to me during an interview.I need to ask this question from Uniform and Non Uniform Quantization in chapter Formatting and Baseband Modulation of Digital Communications

Answer»

The correct choice is (d) 87.6

The best I can EXPLAIN: Another FAMOUS compression characteristic used is A-LAW. In this law, the standard value of A is 87.6.

58.

Which type of quantization is most preferable for audio signals for a human ear?(a) Uniform quantization(b) Non uniform quantization(c) Uniform & Non uniform quantization(d) None of the mentionedThis question was addressed to me during an online exam.I would like to ask this question from Uniform and Non Uniform Quantization in division Formatting and Baseband Modulation of Digital Communications

Answer»

Correct option is (b) NON uniform quantization

The explanation is: The HUMAN ear is sensitive to quantization error in SMALL VALUES so non uniform quantization is more preferable than uniform quantization.

59.

What is the standard value of μ in μ-law ?(a) 128(b) 255(c) 256(d) 0This question was addressed to me by my college professor while I was bunking the class.The origin of the question is Uniform and Non Uniform Quantization topic in division Formatting and Baseband Modulation of Digital Communications

Answer»

The correct answer is (B) 255

To explain I WOULD SAY: The standard value of μ in μ-law is 255.

60.

Which value of μ corresponds to linear amplification?(a) μ=0(b) μ=1(c) μ>0(d) μ

Answer»

The correct option is (a) μ=0

Easy explanation: In μ-law compression CHARACTERISTICS, we get linear amplification or UNIFORM QUANTIZATION when μ=0.

61.

Companding is the process of(a) Compression(b) Expansion(c) Compression & Expansion(d) None of the mentionedThe question was asked in an online quiz.My query is from Uniform and Non Uniform Quantization topic in section Formatting and Baseband Modulation of Digital Communications

Answer»

The CORRECT answer is (c) Compression & Expansion

The best I can EXPLAIN: The given signal is first compressed USING a logarithmic compressor and then it is given as input to the uniform QUANTIZER. Both these steps together is called as companding.

62.

The size of the quantile interval is called as(a) Inter level(b) Step size(c) Quantile size(d) Level widthThis question was posed to me during an online interview.Asked question is from Uniform and Non Uniform Quantization in portion Formatting and Baseband Modulation of Digital Communications

Answer»

Correct ANSWER is (b) Step size

To ELABORATE: The interval between the quantization levels is CALLED as step size.

63.

The output SNR can be made independent of input signal level by using(a) Uniform quantizer(b) Non uniform quantizer(c) Uniform &Non uniform quantizer(d) None of the mentionedI have been asked this question by my college professor while I was bunking the class.The above asked question is from Uniform and Non Uniform Quantization in portion Formatting and Baseband Modulation of Digital Communications

Answer»

Right answer is (b) Non uniform quantizer

Explanation: The WEAK signal experiences POORER SNR compared to high level SIGNALS. So if non uniform quantizer like logarithmic compressor is USED the SNR ratio can be MADE independent of input signal level.

64.

In non uniform quantization, the quantization noise is _______ to signal size.(a) Inversely proportional(b) Directly proportional(c) Equal(d) DoubleThe question was asked during an internship interview.I want to ask this question from Uniform and Non Uniform Quantization in portion Formatting and Baseband Modulation of Digital Communications

Answer»

Correct CHOICE is (B) Directly proportional

For EXPLANATION I would say: In sampling and quantization, the quantization NOISE is directly dependent on signal SIZE.

65.

Uniform quantization provides better quantization for(a) Weak signals(b) Strong signals(c) Weak & Strong signals(d) None of the mentionedI have been asked this question in semester exam.This intriguing question originated from Uniform and Non Uniform Quantization in section Formatting and Baseband Modulation of Digital Communications

Answer»

The correct choice is (b) Strong SIGNALS

To ELABORATE: Signal to noise RATIO is worse for weak level signals.so it provides better quantization for HIGH level signals.

66.

Non uniform quantization provides better quantization for(a) Weak signals(b) Coarse signals(c) Weak & Coarse signals(d) None of the mentionedI got this question in an international level competition.Question is from Uniform and Non Uniform Quantization topic in portion Formatting and Baseband Modulation of Digital Communications

Answer»

Correct choice is (a) Weak SIGNALS

Easy explanation: ACCORDING to signal to NOISE level ratio NON uniform quantization PROVIDES better quantization for weak signals.

67.

Adaptive DPCM is used to(a) Increase bandwidth(b) Decrease bandwidth(c) Increase SNR(d) None of the mentionedI got this question during an interview for a job.My enquiry is from Pulse Code Modulation in portion Formatting and Baseband Modulation of Digital Communications

Answer» RIGHT choice is (B) Decrease BANDWIDTH

Best explanation: ADAPTIVE DPCM is used to decrease REQUIRED bandwidth for the given SNR.
68.

Sample resolution for LPCM ____ bits per sample.(a) 8(b) 16(c) 24(d) All of the mentionedThe question was posed to me during a job interview.Query is from Pulse Code Modulation topic in portion Formatting and Baseband Modulation of Digital Communications

Answer»

The CORRECT option is (d) All of the mentioned

The best I can explain: COMMON sampling resolution for LPCM are 8, 16, 20, 24 bits per SAMPLE.

69.

Delta modulation uses _____ bits per sample.(a) One(b) Two(c) Four(d) EightThis question was addressed to me at a job interview.My enquiry is from Pulse Code Modulation in chapter Formatting and Baseband Modulation of Digital Communications

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Correct choice is (a) One

For explanation: Delta modulation is used for analog to DIGITAL conversion and VICE VERSA. It is a simple form of DPCM. Its uses 1 bit per SAMPLE. It also depends on the difference between the current and previous sample values.

70.

DPCM encodes the PCM values based on(a) Quantization level(b) Difference between the current and predicted value(c) Interval between levels(d) None of the mentionedI had been asked this question in an online interview.Enquiry is from Pulse Code Modulation topic in portion Formatting and Baseband Modulation of Digital Communications

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Correct option is (b) Difference between the current and predicted value

The explanation: Differential PCM ENCODES the PCM value BASED on the difference between the previous sample and the present sample value.

71.

Choosing a discrete value that is near but not exactly at the analog signal level leads to(a) PCM error(b) Quantization error(c) PAM error(d) Sampling errorThe question was posed to me in an internship interview.This is a very interesting question from Pulse Code Modulation in division Formatting and Baseband Modulation of Digital Communications

Answer»

Right choice is (B) Quantization error

To EXPLAIN: One of the limitations of PCM is quantization error which OCCURS when we choose a discrete value at some near by value and not at the analog signal level.

72.

In PCM the samples are dependent on ________(a) Time(b) Frequency(c) Quanization leavel(d) Interval between quantization levelI have been asked this question in an internship interview.The question is from Pulse Code Modulation in section Formatting and Baseband Modulation of Digital Communications

Answer»

The correct choice is (a) Time

The best explanation: The samples DEPEND on time,an ACCURATE clock is REQUIRED for accurate reproduction.

73.

In PCM encoding, quantization level varies as a function of ________(a) Frequency(b) Amplitude(c) Square of frequency(d) Square of amplitudeThis question was posed to me in an interview for job.Enquiry is from Pulse Code Modulation in portion Formatting and Baseband Modulation of Digital Communications

Answer»

Right CHOICE is (b) Amplitude

For explanation I would say: In linear PCM the quantization levels are uniform. But in normal PCM ENCODING the quantization level vary according to the amplitude, BASED of A-law of Myu-law.

74.

What is bit depth?(a) Number of quantization level(b) Interval between two quantization levels(c) Number of possible digital values to represent each sample(d) None of the mentionedThis question was posed to me during an interview.Asked question is from Pulse Code Modulation topic in section Formatting and Baseband Modulation of Digital Communications

Answer» CORRECT answer is (c) Number of possible digital values to represent each SAMPLE

To elaborate: ONE of the PROPERTIES of PCM SIGNAL which determines its stream fidelity is bit depth which is the number of possible digital values that can be used to represent each sample.
75.

Quantization noise can be reduced by ________ the number of levels.(a) Decreasing(b) Increasing(c) Doubling(d) SquaringI had been asked this question during an internship interview.This interesting question is from Pulse Code Modulation topic in section Formatting and Baseband Modulation of Digital Communications

Answer» RIGHT answer is (b) Increasing

The explanation: The process of QUANTIZATION replaces the true signal with the APPROXIMATION(quantization noise). By increasing the number of quantization LEVEL the quantization noise can be reduced.
76.

The length of the code-word obtained by encoding quantized sample is equal to(a) l=log(to the base 2)L(b) l=log(to the base 10)L(c) l=2log(to the base 2)L(d) l=log(to the base 2)L/2The question was posed to me in an internship interview.My enquiry is from Pulse Code Modulation in section Formatting and Baseband Modulation of Digital Communications

Answer»

Correct choice is (a) l=log(to the base 2)L

For EXPLANATION: The quantized sample which are DIGITALLY encoded into l bit value code-word. The LENGTH l can be calculated as l=log(to the base 2)L.

77.

The signals which are obtained by encoding each quantized signal into a digital word is called as(a) PAM signal(b) PCM signal(c) FM signal(d) Sampling and quantizationThe question was posed to me in quiz.My question is taken from Pulse Code Modulation topic in section Formatting and Baseband Modulation of Digital Communications

Answer»

The correct ANSWER is (B) PCM signal

Explanation: Pulse code modulation is the NAME for the class of SIGNALS which are obtained by ENCODING the quantized signals into a digital word.

78.

The ratio of average signal power and quantization noise is(a) 3L^2(b) L^2/3(c) 2L^3(d) L^3/2This question was posed to me during an interview.My question comes from Sources of Corruption topic in portion Formatting and Baseband Modulation of Digital Communications

Answer»

The CORRECT option is (a) 3L^2

The BEST explanation: On calculating the signal power and the quantization NOISE, and on taking its ratio it depends on the number of quantization level L and we get as 3L^2.

79.

Signal to noise ratio is infinite when(a) Quantization noise is zero(b) Number of levels are infinite(c) Quantization noise is zero & Number of levels are infinite(d) None of the mentionedI have been asked this question by my college professor while I was bunking the class.My question is taken from Sources of Corruption topic in portion Formatting and Baseband Modulation of Digital Communications

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The correct OPTION is (C) Quantization noise is ZERO & Number of levels are infinite

To elaborate: In the limit L tends to INFINITY and signal to quantization noise ratio tends to infinity when quantization levels are infinite and quantization noise is zero.

80.

Signal to noise ratio increases as ___________ increases.(a) Quantization level(b) Square of quantization level(c) Square root of quantization level(d) None of the mentionedI got this question at a job interview.I'm obligated to ask this question of Sources of Corruption topic in section Formatting and Baseband Modulation of Digital Communications

Answer»

Right answer is (b) Square of quantization level

The explanation is: On calculating the number of levels, quantization ERROR and POWER and also signal to noise RATION we can find that signal to noise ratio DEPENDS DIRECTLY on square of number of quantization levels.

81.

The _____________ corresponds to average quantization noise power.(a) Mean(b) Variance(c) Probability density function(d) None of the mentionedThe question was posed to me in an online interview.The origin of the question is Sources of Corruption topic in section Formatting and Baseband Modulation of Digital Communications

Answer»

Correct option is (B) Variance

The best I can explain: The variance corresponds to AVERAGE quantization noise POWER. It is CALCULATED assuming the quantization noise and probability DISTRIBUTION function.

82.

When channel bandwidth is greater than the pulse bandwidth, it causes(a) Intersignal interference(b) Intersymbol interference(c) Bandwidth error(d) None of the mentionedThis question was posed to me during an online exam.My enquiry is from Sources of Corruption topic in division Formatting and Baseband Modulation of Digital Communications

Answer»

Correct choice is (B) Intersymbol interference

Best explanation: When CHANNEL BANDWIDTH is greater than pulse bandwidth, the signal widens and expands EXCEEDING the symbol duration which CAUSES intersymbol interference.

83.

Timing jitter can be reduced by(a) Good power supply isolation(b) Stable clock reference(c) Good power supply isolation & Stable clock reference(d) None of the mentionedThis question was addressed to me in final exam.My doubt is from Sources of Corruption topic in section Formatting and Baseband Modulation of Digital Communications

Answer»

The CORRECT ANSWER is (c) Good POWER supply isolation & Stable clock reference

For explanation: Jitter occurs when there is a slight POSITION change in the sampled signals. This timing jitter can be controlled by power supply isolation and clock reference.

84.

Saturation noises can be avoided or reduced by(a) Automatic gain control(b) Amplifying(c) Filtering(d) None of the mentionedThis question was addressed to me by my college director while I was bunking the class.I want to ask this question from Sources of Corruption topic in portion Formatting and Baseband Modulation of Digital Communications

Answer»

The correct choice is (a) Automatic gain CONTROL

To explain: When the DIFFERENCE between input and output signal increases, we say that analog to digital converter is working in saturation. This introduces saturation noise or error. This can be REDUCED by using automatic gain control.

85.

The reasons for the threshold effect are(a) Thermal noise(b) Interference from other users(c) Interference from circuit switching transients(d) All of the mentionedThis question was addressed to me during an online exam.Enquiry is from Sources of Corruption topic in section Formatting and Baseband Modulation of Digital Communications

Answer»

The CORRECT option is (d) All of the mentioned

The explanation: The channel noise which is CAUSED by thermal noise interference from other USERS and CIRCUIT switching TRANSIENTS is called as threshold effect.

86.

The distortion in quantization is called as(a) Round off error(b) Truncation error(c) Round off & Truncation error(d) None of the mentionedThe question was posed to me by my school teacher while I was bunking the class.I want to ask this question from Sources of Corruption topic in chapter Formatting and Baseband Modulation of Digital Communications

Answer»

Correct OPTION is (c) Round off & Truncation error

Best explanation: After sampling and QUANTIZATION of input signals, the ouput SAMPLED sequence CONSISTS of some distortion which can be called as round off error or truncation error.

87.

In quantization process, the amount of quantization noise is _______________ to number of levels.(a) Directly proportional(b) Inversely proportional(c) Independent(d) None of the mentionedI have been asked this question in an internship interview.This interesting question is from Sources of Corruption topic in portion Formatting and Baseband Modulation of Digital Communications

Answer»

Correct answer is (b) Inversely proportional

Easy explanation: The distortion INTRODUCED to APPROXIMATE the analog signal is CALLED as quantization noise. The amount of this noise is inversely proportional to number of levels EMPLOYED in quantization process.

88.

The main sources of corruption are(a) Sampling and quantizing effects(b) Channel effects(c) Sampling, quantizing and channel effects(d) None of the mentionedThis question was addressed to me in an interview for internship.My query is from Sources of Corruption in portion Formatting and Baseband Modulation of Digital Communications

Answer»

Right option is (c) Sampling, QUANTIZING and CHANNEL effects

Easiest explanation: The ANALOG SIGNAL obtained from sampling quantization and transmitted pulses will have corruption from several sources in which the two main sources are quantizing and sampling effect and channel effect.

89.

Flat top sampling or practical sampling has(a) Same frequency(b) Same amplitude(c) Same time difference(d) None of the mentionedThe question was asked in final exam.Question is taken from Formatting Analog Information topic in section Formatting and Baseband Modulation of Digital Communications

Answer»

Correct choice is (B) Same amplitude

The best explanation: In FLAT TOP SAMPLING the top of the PULSES are flat which in turn means that they have the same amplitude.

90.

Multiplication of input signal with pulse train is done in ________ sampling.(a) Impulse sampling(b) Natural sampling(c) Flat top sampling(d) None of the mentionedThis question was addressed to me by my school principal while I was bunking the class.My enquiry is from Formatting Analog Information in division Formatting and Baseband Modulation of Digital Communications

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The correct option is (B) Natural SAMPLING

Best explanation: In impulse sampling the INPUT signal is MULTIPLIED with impulse TRAIN and in natural sampling it is multiplied with pulse train.

91.

Which process is more economical?(a) Undersampling(b) Oversampling(c) Aliasing(d) None of the mentionedThe question was posed to me at a job interview.I want to ask this question from Formatting Analog Information topic in section Formatting and Baseband Modulation of Digital Communications

Answer»

Correct OPTION is (b) Oversampling

Explanation: Oversampling is most economic way of sampling or for CONVERTING ANALOG information to digital as performing signal processing USING digital SYSTEM is less costlier than doing it with high performace analog system.

92.

The effects of aliasing are ________(a) Attenuation of high frequency spectral replicates(b) Non uniform spectral gain applied to desired baseband spectrum(c) Attenuation and non uniform spectral gain(d) None of the mentionedThis question was posed to me in exam.Origin of the question is Formatting Analog Information in division Formatting and Baseband Modulation of Digital Communications

Answer»

The CORRECT OPTION is (c) Attenuation and NON uniform spectral gain

Explanation: Aliasing is due to UNDERSAMPLING and its effects are attenuation and non uniform spectral gain.

93.

The process in which the top of each pulse in the output samples retains the shape of the analog segment is called as ________(a) Natural sampling(b) Ideal sampling(c) Aliasing(d) None of the mentionedI got this question during an interview.Query is from Formatting Analog Information in portion Formatting and Baseband Modulation of Digital Communications

Answer»

Right option is (a) Natural sampling

Explanation: In the METHOD of natural sampling, the top of each pulse in the sampled SEQUENCE retains the same SHAPE of the analog input SIGNAL.

94.

The fourier tranform of one impulse train is also another impulse train with a period of the output equal to the(a) Period of the input(b) Reciprocal of the period of input signal(c) Half the period of input(d) Twice the period of the inputThe question was posed to me during an interview.My doubt stems from Formatting Analog Information in portion Formatting and Baseband Modulation of Digital Communications

Answer»

The CORRECT answer is (b) Reciprocal of the period of input signal

The EXPLANATION is: When we PERFORM fourier TRANFORM on one impulse TRAIN we will get another impulse train with its period reciprocally related to each other.

95.

The output of sampling process are called as ________(a) Pulse code modulation(b) Pulse amplitude modulation(c) Frequency modulation(d) Amplitude modulationThe question was asked during an internship interview.My question comes from Formatting Analog Information topic in portion Formatting and Baseband Modulation of Digital Communications

Answer»

Right answer is (b) Pulse AMPLITUDE MODULATION

Explanation: In sampling process, input DATA is split up into samples whose OUTPUT samples are called as pulse amplitude modulation as the amplitude of the samples is DERIVED from the input waveform.

96.

According to Sampling theorem(a) Ts is greater than 1/2fm(b) Ts is lesser than 1/2fm(c) Ts is equal to 1/2fm(d) Ts is lesser than or equal to 1/2fmThe question was posed to me during an interview for a job.My question is from Formatting Analog Information topic in section Formatting and Baseband Modulation of Digital Communications

Answer»

The correct option is (d) Ts is lesser than or equal to 1/2fm

Easiest EXPLANATION: By SAMPLING THEOREM the INPUT can be derived from the output SAMPLES if Ts is lesser than or equal to 1/2fm.

97.

The sampling process includes methods such as(a) Filtering(b) Sample and hold(c) Amplifying(d) None of the mentionedThe question was posed to me in exam.This intriguing question originated from Formatting Analog Information topic in portion Formatting and Baseband Modulation of Digital Communications

Answer»

The CORRECT choice is (B) Sample and hold

The BEST I can explain: The analog data is converted to digital data through sampling. Sampling is done using sample and hold mechanism which uses transistor, capacitor or SHUTTER etc.