Digital Communications MCQs – Formatting and Baseband Modulation MCQs ( Digital Communications ) MCQs
Latest Digital Communications MCQs
By practicing these MCQs of Formatting and Baseband Modulation MCQs ( Digital Communications ) MCQs – Latest Competitive MCQs , an individual for exams performs better than before. This post comprising of objective questions and answers related to “Formatting and Baseband Modulation MCQs ( Digital Communications ) Mcqs “. As wise people believe “Perfect Practice make a Man Perfect”. It is therefore practice these mcqs of Digital Communications to approach the success. Tab this page to check “Formatting and Baseband Modulation MCQs ( Digital Communications )” for the preparation of competitive mcqs, FPSC mcqs, PPSC mcqs, SPSC mcqs, KPPSC mcqs, AJKPSC mcqs, BPSC mcqs, NTS mcqs, PTS mcqs, OTS mcqs, Atomic Energy mcqs, Pak Army mcqs, Pak Navy mcqs, CTS mcqs, ETEA mcqs and others.
Formatting and Baseband Modulation MCQs ( Digital Communications ) MCQs – Digital Communications MCQs
The most occurred mcqs of Formatting and Baseband Modulation MCQs ( Digital Communications ) in past papers. Past papers of Formatting and Baseband Modulation MCQs ( Digital Communications ) Mcqs. Past papers of Formatting and Baseband Modulation MCQs ( Digital Communications ) Mcqs . Mcqs are the necessary part of any competitive / job related exams. The Mcqs having specific numbers in any written test. It is therefore everyone have to learn / remember the related Formatting and Baseband Modulation MCQs ( Digital Communications ) Mcqs. The Important series of Formatting and Baseband Modulation MCQs ( Digital Communications ) Mcqs are given below:
Formatting Analog Information
1. The sampling process includes methods such as
a) Filtering
b) Sample and hold
c) Amplifying
d) None of the mentioned
Answer: b
Explanation: 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.
2. The output of sampling process are called as ________
a) Pulse code modulation
b) Pulse amplitude modulation
c) Frequency modulation
d) Amplitude modulation
Answer: b
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.
3. 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 1/2fm
Answer: d
Explanation: By sampling theorem the input can be derived from the output samples if Ts is lesser than or equal to 1/2fm.
4. 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 input
Answer: b
Explanation: When we perform fourier tranform on one impulse train we will get another impulse train with its period reciprocally related to each other.
5. 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 mentioned
Answer: a
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.
6. 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 mentioned
Answer: c
Explanation: Aliasing is due to undersampling and its effects are attenuation and non uniform spectral gain.
7. Aliasing can be removed using
a) Prefiltering
b) Postfiltering
c) Prefiltering & Postfiltering
d) None of the mentioned
Answer: c
Explanation: Aliasing can be removed using both prefiltering and postfiltering but prefiltering is most effective and preferred.
8. Which process is more economical?
a) Undersampling
b) Oversampling
c) Aliasing
d) None of the mentioned
Answer: b
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.
9. Flat top sampling or practical sampling has
a) Same frequency
b) Same amplitude
c) Same time difference
d) None of the mentioned
Answer: b
Explanation: In flat top sampling the top of the pulses are flat which in turn means that they have the same amplitude.
10. 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 mentioned
Answer: b
Explanation: In impulse sampling the input signal is multiplied with impulse train and in natural sampling it is multiplied with pulse train.
Sources of Corruption
1. The main sources of corruption are
a) Sampling and quantizing effects
b) Channel effects
c) Sampling, quantizing and channel effects
d) None of the mentioned
Answer: c
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.
2. The distortion in quantization is called as
a) Round off error
b) Truncation error
c) Round off & Truncation error
d) None of the mentioned
Answer: c
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.
3. 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 mentioned
Answer: b
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.
4. Saturation noises can be avoided or reduced by
a) Automatic gain control
b) Amplifying
c) Filtering
d) None of the mentioned
Answer: a
Explanation: 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.
5. 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 mentioned
Answer: c
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.
6. 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 mentioned
Answer: d
Explanation: The channel noise which is caused by thermal noise interference from other users and circuit switching transients is called as threshold effect.
7. When channel bandwidth is greater than the pulse bandwidth, it causes
a) Intersignal interference
b) Intersymbol interference
c) Bandwidth error
d) None of the mentioned
Answer: b
Explanation: When channel bandwidth is greater than pulse bandwidth, the signal widens and expands exceeding the symbol duration which causes intersymbol interference.
8. The _____________ corresponds to average quantization noise power.
a) Mean
b) Variance
c) Probability density function
d) None of the mentioned
Answer: b
Explanation: The variance corresponds to average quantization noise power. It is calculated assuming the quantization noise and probability distribution function.
9. 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 mentioned
Answer: b
Explanation: 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.
10. 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 mentioned
Answer: c
Explanation: 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.
11. The ratio of average signal power and quantization noise is
a) 3L2
b) L2/3
c) 2L3
d) L3/2
Answer: a
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 3L2.
Pulse Code Modulation
1. 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 quantization
Answer: b
Explanation: Pulse code modulation is the name for the class of signals which are obtained by encoding the quantized signals into a digital word.
2. 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/2
Answer: a
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.
3. Quantization noise can be reduced by ________ the number of levels.
a) Decreasing
b) Increasing
c) Doubling
d) Squaring
Answer: b
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.
4. In PCM encoding, quantization level varies as a function of ________
a) Frequency
b) Amplitude
c) Square of frequency
d) Square of amplitude
Answer: b
Explanation: 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.
5. 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 mentioned
Answer: c
Explanation: 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.
6. 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 error
Answer: b
Explanation: 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.
7. In PCM the samples are dependent on ________
a) Time
b) Frequency
c) Quanization leavel
d) Interval between quantization level
Answer: a
Explanation: The samples depend on time,an accurate clock is required for accurate reproduction.
8. 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 mentioned
Answer: b
Explanation: Differential PCM encodes the PCM value based on the difference between the previous sample and the present sample value.
9. Delta modulation uses _____ bits per sample.
a) One
b) Two
c) Four
d) Eight
Answer: a
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.
10. Sample resolution for LPCM ____ bits per sample.
a) 8
b) 16
c) 24
d) All of the mentioned
Answer: d
Explanation: Common sampling resolution for LPCM are 8, 16, 20, 24 bits per sample.
11. Adaptive DPCM is used to
a) Increase bandwidth
b) Decrease bandwidth
c) Increase SNR
d) None of the mentioned
Answer: b
Explanation: Adaptive DPCM is used to decrease required bandwidth for the given SNR.
Uniform and Non Uniform Quantization
1. The size of the quantile interval is called as
a) Inter level
b) Step size
c) Quantile size
d) Level width
Answer: b
Explanation: The interval between the quantization levels is called as step size.
2. Uniform quantization provides better quantization for
a) Weak signals
b) Strong signals
c) Weak & Strong signals
d) None of the mentioned
Answer: b
Explanation: Signal to noise ratio is worse for weak level signals.so it provides better quantization for high level signals.
3. Non uniform quantization provides better quantization for
a) Weak signals
b) Coarse signals
c) Weak & Coarse signals
d) None of the mentioned
Answer: a
Explanation: According to signal to noise level ratio non uniform quantization provides better quantization for weak signals.
4. In non uniform quantization, the quantization noise is _______ to signal size.
a) Inversely proportional
b) Directly proportional
c) Equal
d) Double
Answer: b
Explanation: In sampling and quantization, the quantization noise is directly dependent on signal size.
5. 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 mentioned
Answer: b
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.
6. Companding is the process of
a) Compression
b) Expansion
c) Compression & Expansion
d) None of the mentioned
Answer: c
Explanation: 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.
7. Which value of μ corresponds to linear amplification?
a) μ=0
b) μ=1
c) μ>0
d) μ<0
Answer: a
Explanation: In μ-law compression characteristics, we get linear amplification or uniform quantization when μ=0.
8. What is the standard value of μ in μ-law ?
a) 128
b) 255
c) 256
d) 0
Answer: b
Explanation: The standard value of μ in μ-law is 255.
9. The standard value of A in A-law is
a) 87
b) 88
c) 86.7
d) 87.6
Answer: d
Explanation: Another famous compression characteristic used is A-law. In this law, the standard value of A is 87.6.
10. 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 mentioned
Answer: b
Explanation: The human ear is sensitive to quantization error in small values so non uniform quantization is more preferable than uniform quantization.
Baseband Transmission – 1
1. Which waveforms are also called as line codes?
a) PCM
b) PAM
c) FM
d) AM
Answer: a
Explanation: When pulse modulation is applied to binary symbol we obtain pulse code modulated waveforms. These waveforms are also called as line codes.
2. When pulse code modulation is applied to non binary symbols we obtain waveform called as
a) PCM
b) PAM
c) M-ary
d) line codes
Answer: c
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.
3. Examples of PCM waveforms are
a) Non return to zero
b) Phase encoded
c) Multilevel binary
d) All of the mentioned
Answer: d
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.
4. Which type is used and preferred in digital logic circuits?
a) NRZ-L
b) NRZ-M
c) NRZ-S
d) None of the mentioned
Answer: a
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.
5. Which method is called as differential encoding?
a) NRZ-L
b) NRZ-M
c) NRZ-S
d) None of the mentioned
Answer: b
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.
6. Which method is preferred in magnetic tape recording?
a) NRZ-L
b) NRZ-M
c) NRZ-S
d) None of the mentioned
Answer: b
Explanation: NRZ-M is also called as differential encoding and it is most preferred in magentic tape recording.
Baseband Demodulation And Detection MCQs
7. NRZ-S is complement of _______
a) NRZ-L
b) NRZ-M
c) NRZ-L & NRZ-M
d) None of the mentioned
Answer: b
Explanation: NRZ-S is a complement of NRZ-M. Logic 0 is represented by a change in voltage level and logic 1 is represented as no change in voltage level.
8. The return to zero waveform consists of
a) Unipolar RZ
b) Bipolar RZ
c) RZ-AMI
d) All of the mentioned
Answer: d
Explanation: Different types of return to zero waveforms are unipolar RZ, bipolar RZ, RZ-AMI. These are used in baseband transmission and in magnetic recording.
9. Phase encoded group consists of
a) Manchester coding
b) Bi-phase-mark
c) Miller coding
d) All of the mentioned
Answer: d
Explanation: Different types of phase encoded waveform consists of manchester coding, bi-phase-mark, bi-phase-space, delay modulation.
10. In which waveform logic 1 is represented by half bit wide pulse and logic 0 is represented by absence of pulse?
a) Unipolar RZ
b) Bipolar RZ
c) RZ-AMI
d) Manchester coding
Answer: a
Explanation: In unipolar RZ waveform, logic 1 is represented by half bit wide pulse and logic 0 is represented by the absence of a pulse.
11. In which waveform logic 1 and logic 0 are represented by opposite one half bit wide pulses?
a) Unipolar RZ
b) Bipolar RZ
c) RZ-AMI
d) Manchester coding
Answer: b
Explanation: In bipolar return to zero waveform ones and zeroes are represented by opposite level pulses one half bit wide pulses.
12. In which waveform logic 1 is represented by equal amplitude alternating pulses?
a) Unipolar RZ
b) Bipolar RZ
c) RZ-AMI
d) Manchester coding
Answer: c
Explanation: In RZ-AMI logic 1 is represented by equal amplitude alternating pulses and logic 0 is represented by the absence of a pulse.
Baseband Transmission-2
1. Application of phase encoded binary signals are
a) Optical communication
b) Magnetic recording
c) Satellite telemetry
d) All of the mentioned
Answer: d
Explanation: Some of the fields where phase encoded waveforms is being used are optical communication, magnetic tape recording, satellite telemetry etc.
2. In which waveform one is represented by half bit wide pulse positioned during the first half and zero is represented by half bit wide pulse positioned in the second half?
a) Bi-p-L
b) Bi-p-M
c) Bi-p-S
d) Delay modulation
Answer: a
Explanation: In bi-phase-level one is represented by half bit wide pulse positioned during the first half and zero is represented by half bit wide pulse positioned in the second half.
3. Which binary waveform uses three levels?
a) Bipolar RZ
b) RZ-AMI
c) Bipolar RZ & RZ-AMI
d) None of the mentioned
Answer: c
Explanation: In PCM waveforms signals generally use two levels. But few signals use three levels such as bipolar RZ, RZ-AMI, dicode, duobinary etc.
4. Which waveform type has the feature of clocking?
a) Manchester coding
b) Bbi-p-M
c) Delay modulation
d) NRZ-L
Answer: a
Explanation: In manchester coding transition occurs in the middle of every bit interval. Thus it has a feature of clocking.
5. Which waveform has the feature of error detection?
a) NRZ-L
b) RZ-AMI
c) Manchester coding
d) Duobinary
Answer: d
Explanation: Duobinary scheme does error detection without introducing any additional error bits into the data sequence.
6. Which waveform scheme introduces bandwidth compression?
a) Duobinary
b) Manchester coding
c) Phase encoded waveform
d) Multilevel codes
Answer: d
Explanation: Multilevel codes increases the bandwidth efficiency by reducing the bandwidth utilization for the given data rate.
7. Which waveform type has better noise immunity?
a) NRZ
b) RZ
c) Phase encoded
d) Multilevel codes
Answer: a
Explanation: NRZ waveforms has better error performance than RZ signal waveforms.
8. In pulse modulation, reciprocal of T(time) is
a) Bandwidth
b) Symbol rate
c) Signal voltage
d) None of the mentioned
Answer: b
Explanation: T is the time period of the signal. In pulse modulation, the symbol rate Rs can be given as reciprocal of T.
9. PCM word size can be described by
a) Time period
b) Symbol rate
c) Number of quantization levels
d) None of the mentioned
Answer: c
Explanation: Each analog sample is transformed into PCM word made up of a group of bits. The PCM word size can be described by number of quantization levels allowed for each sample.
10. Some of the M-ary waveforms are
a) PAM
b) PPM
c) PDM
d) All of the mentioned
Answer: d
Explanation: When we pulse modulate non binary signals we get M-ary waveform. Some of the examples of M-ary waveforms are PAM, PPM, PDM.
11. Which method should be implemented for reducing bandwidth?
a) Multilevel codes
b) Multilevel signalling
c) PAM
d) PDM
Answer: b
Explanation: The transmission bandwidth required for binary digital waveforms is large. To reduce the bandwidth, multilevel signalling method can be used.
12. In M-ary PPM waveform, modulation is effected by
a) Delaying
b) Advancing
c) Delaying & Advancing
d) None of the mentioned
Answer: c
Explanation: In M-ary PPM waveform, modulation is effected by delaying or advancing pulse occurrence by an amount that corresponds to information symbol.
13. For both PPM and PDM _______ is kept constant.
a) Amplitude
b) Time period
c) Frequency
d) Number of levels
Answer: a
Explanation: In PPM waveform delaying or advancing of pulse is done. In PDM the pulse width is varied. Thus in both the cases amplitude is maintained constant.
Correlative Coding
1. The method in which small amount of controlled ISI is introduced into the data stream rather than trying to eliminate it completely is called as
a) Correlative coding
b) Duobinary signalling
c) Partial response signalling
d) All of the mentioned
Answer: d
Explanation: The interference at the detector can be cancelled out using these methods in which some controlled amount of ISI is introduced into the data stream.
2. From digital filter we will get the output pulse as the _______ of the current and the previous pulse.
a) Summation
b) Difference
c) Product
d) Ratio
Answer: a
Explanation: The digital filter incorporates one digit delay and thus it adds the incoming pulse with the value of the previous pulse.
3. In duobinary signalling method, for M-ary transmission, the number of output obtained is
a) 2M
b) 2M+1
c) 2M-1
d) M2
Answer: c
Explanation: In duobinary coding, the number of output obtained for M-ary transmission is 2M-1.
4. The method using which the error propagation in dubinary signalling can be avoided is
a) Filtering
b) Precoding
c) Postcoding
d) None of the mentioned
Answer: b
Explanation: In duobinary signalling method if one error occurs it repeats everywhere through out the next steps. To avoid this precoding method can be used.
5. In precoding technique, the binary sequence is _____ with the previous precoded bit.
a) And-ed
b) Or-ed
c) EXOR-ed
d) Added
Answer: c
Explanation: To avoid error propogation precoding method is used. In this each bit is encoded individually without having any effect due to its prior bit or decisions.
6. The duobinary filter, He (f) is called as
a) Sine filter
b) Cosine filter
c) Raised cosine filter
d) None of the mentioned
Answer: b
Explanation: The transfer function is 2T cos(πfT) which is called as cosine filter.
7. The method which has greater bandwidth efficiency is called as
a) Duobinary signalling
b) Polybinary signalling
c) Correlative coding
d) All of the mentioned
Answer:b
Explanation: If more than three levels are introduced in duobinary signalling technique the bandwidth efficiency increases This method is called as polybinary signalling.
8. In polybinary signalling method the present bit of binary sequence is algebraically added with ______ number of previous bits.
a) j
b) 2j
c) j+2
d) j-2
Answer: d
Explanation: In polybinary signalling method the present binary digit of the sequence is formed from the modulo-2 addition of the j-2 preceding digits of the sequence and the present digit.
9. The primary advantage of this method is
a) redistribution of spectral density
b) to favor low frequencies
c) redistribution of spectral density & to favor low frequencies
d) none of the mentioned
Answer: c
Explanation: Each bit can be independently detected in-spite of strong correlation and this provides redistribution of spectral density and also favors low frequencies.
10. Source encoding procedure does
a) Sampling
b) Quantization
c) Compression
d) All of the mentioned
Answer: d
Explanation: Source encoding includes a sampling of continuous time signals, quantization of continuous valued signals and compression of those sources.
Sampling and Quantization – 1
1. Which signals are function of time?
a) Random signal
b) Deterministic signal
c) Random & Deterministic signal
d) None of the mentioned
Answer: b
Explanation: Deterministic signals are function of time.
2. Auto-correlation function is a
a) Even function
b) Odd function
c) Even & Odd function
d) None of the mentioned
Answer: a
Explanation: Auto-correlation function is an even function of time.
3. Shot noise occurs in
a) Transistors
b) Valves
c) Transistors & Valves
d) None of the mentioned
Answer: c
Explanation: Shot noise occurs in both valves and transistors.
4. Source coding reduces
a) Redundancy
b) Average bit rate
c) Redundancy & Average bit rate
d) None of the mentioned
Answer: c
Explanation: Source coding reduces both average bit rate and reduces redundancy.
5. Delay element in delta modulation acts as
a) First order predictor
b) Second order predictor
c) Third order predictor
d) Fourth order predictor
Answer: a
Explanation: Delay element act as first order predictor.
6. Non uniform quantization includes
a) Compression
b) Expansion
c) Compression & Expansion
d) None of the mentioned
Answer: c
Explanation: Compression and expansion give the feature of non uniform quantization.
7. The quantization will be finer when
a) Smaller the number of discrete amplitudes
b) Larger the number of discrete amplitudes
c) Does not depend on amplitudes
d) None of the mentioned
Answer: b
Explanation: Larger the number of discrete amplitudes, finer will be the quantization.
8. Different cases of sampling include
a) Ideal impulse sampling
b) Flat-topped sampling
c) Sampling with rectangular pulses
d) All of the mentioned
Answer: d
Explanation: The three cases of sampling are ideal impulse sampling, sampling with rectangular pulses and flat topped sampling.
9. Transmitted pulse becomes distorted due to
a) Ideal transmission characteristic
b) Non ideal transmission characteristic
c) All of the mentioned
d) None of the mentioned
Answer: b
Explanation: A transmitted pulse gradually becomes distorted due to non ideal transmission characteristic of the channel.
10. In which mixing is easier?
a) Analog signal
b) Digital signal
c) Analog & Digital signal
d) None of the mentioned
Answer: b
Explanation: Mixing of digital signals are easier than that of mixing analog signals.
Sampling and Quantization – 2
1. Which filter does not have sharp output?
a) Linear phase filter
b) Delayed symmetric filter
c) Linear phase & Delayed symmetric filter
d) None of the mentioned
Answer: c
Explanation: Linear phase filter or delayed symmetric filter does not have sharp output but sampling can be done in real time.
2. Using ARMA filter
a) Sampling can be done in real time
b) Gives sharp output
c) All of the mentioned
d) None of the mentioned
Answer: b
Explanation: ARMA filter gives sharp output but sampling cannot be done in real time.
3. To avoid aliasing
a) Reduce the bandwidth
b) Cut out high frequency
c) Reduce the bandwidth & Cut out high frequency
d) None of the mentioned
Answer: c
Explanation: To avoid aliasing bandwidth should be reduced and high frequency should be cut out.
4. Which requires interpolation filtering?
a) Up-sampler
b) D to A converter
c) Up-sampler & D to A converter
d) None of the mentioned
Answer: c
Explanation: Up-sampler and D to A conversion need interpolation filtering.
5. A to D conversion process uses
a) Sampler
b) Quantizer
c) Sampler & Quantizer
d) None of the mentioned
Answer: c
Explanation: A to D conversion process requires both sampler and quantizer.
6. Reconstruction filter is difficult to implement in hardware.
a) True
b) False
Answer: b
Explanation: Reconstruction filter is simpler and easy to implement in hardware.
7. Which process requires low pass filter?
a) Up-sampling
b) Down-sampling
c) Up-sampling & Down-sampling
d) None of the mentioned
Answer: c
Explanation: Up-sampling requires low pass filter after increasing the data rate and down-sampling requires low pass filter before decimation.
8. Decreasing the data rate is called as
a) Aliasing
b) Down sampling
c) Up sampling
d) None of the mentioned
Answer: b
Explanation: Decreasing the data rate is called as down sampling and increasing the data rate is called as up sampling.
9. Original signal came to be retraced from sampled version using
a) Low pass filtering
b) High pass filtering
c) Low & High pass filtering
d) None of the mentioned
Answer: a
Explanation: Original signal can be obtained from its sampled version by using low pass filtering.
10. The signal can be reconstructed
a) At Nyquist rate
b) Above Nyquist rate
c) At & above the Nyquist rate
d) None of the mentioned
Answer: c
Explanation: The signal can be reconstructed from the sampled version at or above Nyquist rate using simple low pass filtering.
11. Which device is needed for the reconstruction of signal?
a) Low pass filter
b) Equalizer
c) Low pass filter & Equalizer
d) None of the mentioned
Answer: c
Explanation: Equalizer followed by a low pass filter is necessary for reconstruction of a signal from its sampled version.