A/D Converters Sampling Questions and Answers - Sanfoundry

by Manish

This set of Digital Signal Processing Multiple Choice Questions & Answers (MCQs) focuses on
“Oversampling A/D Converters”.

1. 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) False
View Answer

Answer: a
Explanation: The dynamic range of the signal, which is proportional to its standard deviation σ x ,
should match the range R of the quantizer, it follows that ∆ is proportional to σx. Hence for a
given number o f bits, the power o f the quantization noise is proportional to the variance of
the signal to be quantized.

2. What is the variance of the difference between two successive signal samples, d(n) = x(n) –
x(n-1) ?

View Answer

Answer: b

Explanation:

3.What is the variance of the difference between two successive signal samples, d(n) = x(n) –
ax(n-1) ?

View Answer
Answer: c

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). Therefore

4. If the difference d(n) = x(n) –ax(n-1), then what is the optimum choice for a = ?

View Answer

Answer: a

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

5. 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)
View Answer

Answer: c
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).

6. The differential predictive signal quantizer system is known as?

a) DCPM
b) DMPC
c) DPCM
d) None of the mentioned
View Answer
Answer: c
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 ).

7. 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 mentioned
View Answer

Answer: a
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 ).

8. 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 channels
View Answer

Answer: d
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 ).

9. To reduce the dynamic range of the difference signal d(n) = x(n) – x ̂(n), thus a predictor of
order p has the form?

View Answer

Answer: b
Explanation: T he goal of the predictor is to provide an estimate 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) – x ̂(n). Thus a predictor of order p has the form

10. The simplest form of differential predictive quantization is called?

a) AM
b) BM
c) DM
d) None of the mentioned
View Answer

Answer: c
Explanation: The simplest form of differential predictive quantization is called delta modulation
(DM ).

11. What is the abbreviation of DM?

a) Diameter Modulation
b) Distance Modulation
c) Delta Modulation
d) None of the mentioned
View Answer

Answer: c
Explanation: The simplest form of differential predictive quantization is called delta modulation
(DM ).

12. 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 level
View Answer

Answer: b
Explanation: T he simplest form o f differential predictive quantization is called delta
modulation (DM ). In DM, the quantizer is a simple 1-bit (two -level) quantizer.

13. 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 predictor
View Answer

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

14. 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 & accumulator
View Answer

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

15. 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 & accumulator
View Answer

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

Sample and Hold Questions and Answers - Sanfoundry

by Manish

This set of Digital Signal Processing Multiple Choice Questions & Answers (MCQs) focuses on
“Sample and Hold”.

1. 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 mentioned
View Answer

Answer: b
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 ).

2. 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 mentioned
View Answer

Answer: a
Explanation: 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 o f the digital word applied to its
input.

3. T he 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) False
View Answer

Answer: a
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. T he 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 o f the signal at the time the system is switched from the sample
mode to the hold mode.

4. 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 to
View Answer
Answer: c
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.

5. 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 mode
View Answer

Answer: a
Explanation: 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.

6. 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 mentioned
View Answer

Answer: d
Explanation: 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 o f the
sampling aperture, and changes in the voltage held during conversion ( “droop”) do occur in
practical devices.

7. 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) False
View Answer

Answer: b
Explanation: 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.

8. The noise power σn2 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) False
View Answer

Answer: a
Explanation: The noise power σn2 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.

9. What is the process of down sampling called?

a) Decimation
b) Fornication
c) Both Decimation & Fornication
d) None of the mentioned
View Answer

Answer: a
Explanation: To avoid aliasing, w e first filter out the out-of-band (fl, F2) noise by processing the
wideband signal. The signal is then passed through the low pass filter and re-sampled (down
sampled) at the lower rate. The down sampling process is called decimation.

10. 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) False
View Answer

Answer: a
Explanation: If the interpolation factor is I = 256, the A /D converter output can be obtained by
averaging successive non-overlapping blocks o f 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.

11. 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 mentioned
View Answer

Answer: c
Explanation: The crosshatched areas illustrate two types of quantization error in DM , slope-
overload distortion and granular noise.

12. The slope-overload distortion is avoided, if which of the following conditions satisfy?
a) Min| dx(t)/d(t) | ≤ ∆/T
b) Max| dx(t)/d(t) | ≤ ∆/T
c) |dx(t)/d(t) | ≤∆/T
d) None of the mentioned
View Answer

Answer: b
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) | ≤∆/T .

13. In DM, By increasing∆, reduces the overload distortion but increases the granular noise, and
vice versa
a) True
b) False
View Answer

Answer: a
Explanation: The granular noise occurs w hen the DM tracks a relatively flat (slowly changing)
input signal. We note that increasing ∆ reduces overload distortion but increases the granular
noise, and vice versa.

14. 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 mentioned
View Answer

Answer: a
Explanation: We note that increasing ∆ 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.

15. What are the effects produced by Dm by setting up an integrator at the front of DM?
a) Simplifies the DM decoder
b) Increases correlation of the signal into the DM input
c) Emphasizes the low frequencies of x(t)
d) All of the mentioned
View Answer

Answer: d
Explanation: One way to reduce these two types of distortion is to use an integrator in front of
the DM. This has two effects. First, it emphasizes the low frequencies of x (t) and increases the
correlation of the signal into the DM input. Second, it simplifies the DM decoder because the
differentiator (inverse system) required at the decoder is canceled by the DM integrator.

Bandpass Signal Sampling Questions and Answers - Sanfoundry

by Manish

This set of Digital Signal Processing Multiple Choice Questions & Answers (MCQs) focuses on
“Sampling of Band Pass Signals”.

1. The frequency shift can be achieved by multiplying the band pass signal as given in equation
by the quadrature carriers cos[2πFct] and sin[2πFct] and
lowpass filtering the products to eliminate the signal components of 2F c.
a) True
b) False
View Answer

Answer: a
Explanation: It is certainly be advantageous to perform a frequency shift of the band pass signal
by and sampling the equivalent low pass signal. Such a frequency shift can be achieved by
multiplying the band pass signal as given in the above equation by the quadrature carriers
cos[2πFct] and sin[2πFct] and low pass filtering the products to eliminate the signal components
at 2Fc. Clearly, the multiplication and the subsequent filtering are first performed in the analog
domain and then the outputs o f the filters are sampled.

2. What is the final result obtained by substituting Fc=kB-B/2 , T= 1/2B and say n = 2m i.e., for
even and n=2m-1 for odd in equation
View Answer

Answer: d

Explanation:

3. Which low pass signal component occurs at the rate of B samples per second with even
numbered samples of x(t)?
a) uc– lowpass signal component
b) us– lowpass signal component
c) uc & us – lowpass signal component
d) None of the mentioned
View Answer

Answer: a
Explanation: With the even-numbered samples o f x(t), which occur at the rate o f B samples
per second, produce samples of the low pass signal component u c.

4. Which low pass signal component occurs at the rate of B samples per second with odd
numbered samples of x(t)?
a) uc– lowpass signal component
b) us– lowpass signal component
c) uc & us – lowpass signal component
d) None of the mentioned
View Answer
Answer: b
Explanation: : With the odd-numbered samples o f x(t), which occur at the rate o f B samples
per second, produce samples of the low pass signal component us.

5. What is the reconstruction formula for the bandpass signal x(t) with samples taken at the
rate of 2B samples per second?

View Answer

Answer: a

Explanation:

, where T=1/2B

6. What is the new centre frequency for the increased bandwidth signal ?
a) Fc‘= Fc+B/2+B’/2
b) Fc‘= Fc+B/2-B’/2
c) Fc‘= Fc-B/2-B’/2
d) None of the mentioned
View Answer

Answer: b
Explanation: A new centre frequency for the increased bandwidth signal is Fc‘= Fc+B/2-B’/2

7. According to the sampling theorem for low pass signals with T1=1/B, then what is the
expression for uc(t) = ?
View Answer

Answer: a

Explanation: To reconstruct the equivalent low pass signals. Thus, according to the sampling

theorem for low pass signals with T

=1/B.

8. According to the sampling theorem for low pass signals with T1=1/B, then what is the
expression for us(t) = ?

View Answer

Answer: b

Explanation: To reconstruct the equivalent low pass signals. Thus, according to the sampling

theorem for low pass signals with T

1

=1/B.

9. What is the expression for low pass signal component uc(t) that can be expressed in terms of
samples of the bandpass signal ?

View Answer

Answer: b

Explanation: The low pass signal components u

(t) can be expressed in terms of samples of the

band pass signal as follows:

10. What is the expression for low pass signal component us(t) that can be expressed in terms
of samples of the bandpass signal ?

View Answer

Answer: a
Explanation: The low pass signal components u

(t) can be expressed in terms of samples of the

band pass signal as follows:

11. What is the Fourier transform of x(t) ?

View Answer

Answer: d

Explanation:

12. What is the basic relationship between the spectrum o f the real band pass signal x( t ) and
the spectrum of the equivalent low pass signal xl(t) ?

View Answer

Answer: d
Explanation:

, where X

(F) is the Fourier transform of x

(t). This is the basic relationship between the spectrum o f the real band pass signal x ( t ) and
the spectrum of the equivalent low pass signal x

(t).

Bandpass Signal Representation Questions

and Answers - Sanfoundry
by Manish

This set of Digital Signal Processing Multiple Choice Questions & Answers (MCQs) focuses on
“The Representation of Bandpass Signals”.

1. Which of the following is the right way of representation of equation that contains only the
positive frequencies in a given x(t) signal?
a) X+(F)=4V(F)X(F)
b) X+(F)=V(F)X(F)
c) X+(F)=2V(F)X(F)
d) X+(F)=8V(F)X(F)
View Answer

Answer: c
Explanation: In a real valued signal x(t), has a frequency content concentrated in a narrow band
of frequencies in the vicinity of a frequency Fc. Such a signal which has only positive
frequencies can be expressed as X+(F)=2V(F)X(F)
Where X+(F) is a Fourier transform of x(t) and V(F) is unit step function.

2. What is the equivalent time –domain expression of X+(F)=2V(F)X(F)?

a) F(+1)2V(F)*F(+1)X(F)
b) F(-1)4V(F)*F(-1)X(F)
c) F(-1)V(F)*F(-1)X(F)
d) F(-1)2V(F)*F(-1)X(F)
View Answer

Answer: d

Explanation: Given Expression , X

(F)=2V(F)X(F).It can be calculated as follows

3. In time-domain expression, The signal x+(t) is known as

a) Systematic signal
b) Analytic signal
c) Pre-envelope of x(t)
d) Both Analytic signal & Pre-envelope of x(t)
View Answer

Answer: d

Explanation: From the given expression,

4. In equation

Then the value of ẋ(t) is ?

View Answer

Answer: b

Explanation:
5. If the signal ẋ(t) can be viewed as the output of the filter with impulse response h(t) = 1/πt , -∞
<t<∞
when excited by the input signal x(t) then such a filter is called as___
a) Analytic transformer
b) Hilbert transformer
c) Both Analytic & Hilbert transformer
d) None of the mentioned
View Answer

Answer: B
Explanation: The signal ẋ(t) can be viewed as the output of the filter with impulse response h(t)
= 1/πt ,
-∞ < t < ∞ when excited by the input signal x(t) then such a filter is called as Hilbert transformer.

6. What is the frequency response of a Hilbert transform H(F)=?

View Answer

Answer: a
Explanation:

We Observe that │H (F) │=1 and the phase response ʘ(F) = -1/2π for F > 0 and ʘ(F) = 1/2π for
F < 0.

7. What is the equivalent lowpass representation obtained by performing a frequency translation

of X+(F) to Xl(F)= ?
a) X+(F+Fc)
b) X+(F-Fc)
c) X+(F*Fc)
d) X+(Fc-F)
View Answer

Answer: a
Explanation: The analytic signal x+(t) is a bandpass signal. We obtain an equivalent lowpass
representation by performing a frequency translation of X+(F).

8. What is the equivalent time domain relation of xl(t) i.e., lowpass signal?

View Answer

Answer: c

9. If we substitute the equation in equation x (t) +j ẋ (t) = xl(t) e(j2πFct)

and equate real and imaginary parts on side, then what are the relations that we obtain?
View Answer

Answer: b
Explanation: If we substitute the given equation in other, then we get the required result

10. In the relation, the low frequency components uc and

us are called __ of the bandpass signal x(t).
a) Quadratic components
b) Quadrature components
c) Triplet components
d) None of the mentioned
View Answer

Answer: b
Explanation: The low -frequency signal components uc(t) and us(t) can be viewed as amplitude
modulations impressed on the carrier components cos2πFct and sin2πFct , respectively. Since
these carrier components are in phase quadrature, uc(t) and us(t) are called the Quadrature
components of the bandpass signal x (t).

11. What is the other way of representation of bandpass signal x(t)?

View Answer

Answer: a

Explanation: The above signal is formed from quadrature components,

where Re denotes the real part of complex valued quantity.

12. In the equation What is the lowpass signal xl (t) is usually called the
___ of the real signal x(t) ?
a) Mediature envelope
b) Complex envelope
c) Equivalent envelope
D) All of the mentioned
View Answer

Answer: b
Explanation: In the equation x(t) = Re[xl(t)e(j2πFct)],Re denotes the real part of the complex valued
quantity in the brackets following. The lowpass signal x_l (t) is usually called the Complex
envelope of the real signal x(t) , and is basically the equivalent low pass signal.

13. If a possible representation of a band pass signal is obtained by expressing xl (t) as

then what are the equations of a(t) and θ(t) ?

View Answer

Answer: a

Explanation: A third possible representation of a band pass signal is obtained by expressing

14. What is the possible representation of x(t) if xl(t)=a(t)e(jθ(t)) ?

a) x(t) = a(t) cos(2pFct – ?(t))
b) x(t) = a(t) cos(2pFct + ?(t))
c) x(t) = a(t) sin(2pFct + ?(t))
d) x(t) = a(t) sin(2pFct – ?(t))
View Answer

Answer: b

Explanation:

Hence proved.
15. In the equation x(t) = a(t)cos[2πFct+θ(t) ], Which of the following relations between a(t) and
x(t), θ(t) and x(t) are true?
a) a(t), θ(t) are called the Phases of x(t)
b) a(t) is the Phase of x(t), θ(t) is called the Envelope of x(t)
c) a(t) is the Envelope of x(t), θ(t) is called the Phase of x(t)
d) None of the mentioned
View Answer

Answer: c
Explanation: In the equation x(t) = a(t) cos[2πFct+θ(t) ], the signal a(t) is called the Envelope of
x(t), and θ(t) is called the phase of x(t)

Quantization and Coding Questions and

Answers - Sanfoundry
by Manish

This set of Digital Signal Processing Multiple Choice Questions & Answers (MCQs) focuses on
“Quantization and Coding”.

1. The basic task of the A/D converter is to convert a discrete set of digital code words into a
continuous range of input amplitudes.
a) True
b) False
View Answer

Answer: b
Explanation: The basic task of the A /D converter is to convert a continuous range of input
amplitude into a discrete set of digital code words. This conversion involves the processes of
Quantization and Coding.

2. What is the type of quantizer, if a Zero is assigned a quantization level?

a) Midrise type
b) Mid tread type
c) Mistreat type
d) None of the mentioned
View Answer

Answer: b
Explanation: If a zero is assigned a quantization level, the quantizer is of the mid treat type.

3. What is the type of quantizer, if a Zero is assigned a decision level?

a) Midrise type
b) Mid tread type
c) Mistreat type
d) None of the mentioned
View Answer

Answer: a
Explanation: If a zero is assigned a decision level, the quantizer is of the midrise type.

4. What is the term used to describe the range of an A/D converter for bipolar signals?
a) Full scale
b) FSR
c) Full-scale region
d) FS
View Answer

Answer: b
Explanation: The term Full-scale range (FSR) is used to describe the range of an A /D converter
for bipolar signals (i.e., signals with both positive and negative amplitudes).

5. What is the term used to describe the range of an A/D converter for uni-polar signals?
a) Full scale
b) FSR
c) Full-scale region
d) FSS
View Answer

Answer: a
Explanation: The term Full scale (FS) is used for uni-polar signals

6. What is the fixed range of the quantization error eq(n)?

a) – ∆/6 < eq(n) ≤ ∆/6
b) – ∆/4 < eq(n) ≤ ∆/4
c) – ∆/2 <eq(n) ≤ ∆/2
d) – ∆/16 <eq(n) ≤ ∆/16
View Answer

Answer: c
Explanation: The quantization error eq(n) is always in the range – ∆/2 < eq(n) ≤ ∆/2 , where ∆ is
quantizer step size.

7. If the dynamic range of the signal is smaller than the range of quantizer, the samples that
exceed the quantizer are clipped, resulting in large quantization error.
a) True
b) False
View Answer
Answer: b
Explanation: If the dynamic range o f the signal, defined as xmax-xmin, is larger than the range of
the quantizer, the samples that exceed the quantizer range are clipped, resulting in a large
(greater than ∆/2) quantization error.

8. What is the relation defined by the operation of quantizer?

a) xq(n) ≡ Q[x(n)]= xk
b) xq(n) = Q[x(n)]= xk , if x(n) ∈ I_k
c) xq(k) ≡ Q[x(k)]= xk
d) None of the mentioned
View Answer

Answer: b
Explanation: The possible outputs of the quantizer (i.e., the quantization levels) are denoted as
x1,x 2,…xL . The operation of the quantizer is defined by the relation, xq(n) ≡ Q[x(n) ]= xk,if x(n)
∈ Ik.

9. What is the step size or the resolution of an A/D converter?

a) ∆ = (R)/2(b+1)
b) ∆ = (R )/2(b-1)
c) ∆ = (R )/3(b+1)
d) ∆ = (R )/2
View Answer

Answer: a
Explanation: The coding process in an A /D converter assigns a unique binary number to each
quantization level. If we have L levels, we need at least L different binary numbers. With a word
length of b + 1 bits we can represent 2^(b+1) distinct binary numbers. Hence we should have
2^(b+1) > L or, equivalently, b + 1 > log2 L. Then the step size or the resolution of the A /D
converter is given by
∆ = (R )/2(b+1), where R is the range of the quantizer.

10. In the practical A/D converters, if the first transition may not occur at exactly + 1/2 LSB
,then such kind of error is known as ____________
a) Scale-factor error
b) Offset error
c) Linearity error
d) All of the mentioned
View Answer

Answer: b
Explanation: We note that practical A /D converters may have offset error (the first transition
may not occur at exactly +1/2 LSB).

11. In the practical A/D converters, if the difference between the values at which the first
transition and the last transition occur is not equal to FS – 2LSB, then such error is known as
_________
a) Scale-factor error
b) Offset error
c) Linearity error
d) All of the mentioned
View Answer

Answer: a
Explanation: We note that practical A /D converters scale-factor (or gain) error (the difference
between the values at which the first transition and the last transition occur is not equal to FS —
2LSB ).

12. In the practical A/D converters, if the differences between transition values are not all equal
or uniformly changing, then such error is known as ?
a) Scale-factor error
b) Offset error
c) Linearity error
d) All of the mentioned
View Answer

Answer: c
Explanation: We note that practical A /D converters, linearity error (the differences between
transition values are not all equal or uniformly changing).

Digital to Analog Conversion Samples

Questions and Answers - Sanfoundry
by Manish

This set of Digital Signal Processing Multiple Choice Questions & Answers (MCQs) focuses on
“Digital to Analog Conversion Sample and Hold”.

1. What is the ideal reconstruction formula or ideal interpolation formula for x(t) = _________

View Answer
Answer: a

Explanation:

where the sampling interval T = 1/F

=1/2B,F

is the sampling frequency and B is the bandwidth of the analog signal.

2. What is the new ideal interpolation formula described after few problems with previous one?

View Answer

Answer: b

Explanation: The reconstruction of the signal x ( t) from its samples as an interpolation problem
and have described the function:
3. What is the frequency response of the analog filter corresponding to the ideal interpolator?

View Answer

Answer: c

Explanation: The analog filter corresponding to the ideal interpolator has a frequency response:

, H(F) is the Fourier transform of the interpolation function g(t).

4. The reconstruction o f the signal from its samples as a linear filtering process in which a
discrete-time sequence of short pulses (ideally impulses) with amplitudes equal to the signal
samples, excites an analog filter.
a) True
b) False
View Answer

Answer: a
Explanation: The reconstruction o f the signal from its samples as a linear filtering process in
which a discrete-time sequence of short pulses (ideally impulses) with amplitudes equal to the
signal samples, excites an analog filter.

5. The ideal reconstruction filter is an ideal low pass filter and its impulse response extends for
all time.
a) True
b) False
View Answer
Answer: a
Explanation: The ideal reconstruction filter is an ideal low pass filter and its impulse response
extends for all time. Hence the filter is noncausal and physically nonrealizable. Although the
interpolation filter with impulse response given can be approximated closely with some delay,
the resulting function is still impractical for most applications where D /A conversion are
required.

6. D /A conversion is usually performed by combining a D /A converter with a sample-and-hold

(S/H ) and followed by a low pass (smoothing) filter.
a) True
b) False
View Answer

Answer: a
Explanation: D /A conversion is usually performed by combining a D /A converter with a
sample-and hold (S/H) and followed by a low pass (smoothing) filter. T he D /A converter
accepts at its input, electrical signals that correspond to a binary word, and produces an output
voltage or current that is proportional to the value o f the binary word.

7. The time required for the output o f the D /A converter to reach and remain within a given
fraction of the final value, after application of the input code word is called?
a) Converting time
b) Setting time
c) Both Converting & Setting time
d) None of the mentioned
View Answer

Answer: b
Explanation: An important parameter o f a D /A converter is its settling time, which is defined as
the time required for the output o f the D /A converter to reach and remain within a given
fraction (usually,±1/2 LSB) of the final value, after application of the input code word.

8. In D/A converter, the application of the input code word results in a high-amplitude transient,
called?
a) Glitch
b) Deglitch
c) Glitter
d) None of the mentioned
View Answer

Answer: a
Explanation: The application o f the input code word results in a high-amplitude transient, called
a “glitch.” This is especially the case when two consecutive code words to the A /D differ by
several bits.
9. In a D/A converter, the usual way to solve the glitch is to use deglitcher. How is the Deglitcher
designed?
a) By using a low pass filter
b) By using a S/H circuit
c) By using a low pass filter & S/H circuit
d) None of the mentioned
View Answer

Answer: b
Explanation: The usual way to remedy this problem is to use an S/H circuit designed to serve as
a “deglitcher”. Hence the basic task of the S/H is to hold the output of the D /A converter
constant at the previous output value until the new sample at the output o f the D /A reaches
steady state, and then it samples and holds the new value in the next sampling interval. Thus the
S/H approximates the analog signal by a series of rectangular pulses whose height is equal to the
corresponding value of the signal pulse.

10. What is the impulse response of an S/H, when viewed as a linear filter?

d) None
View Answer

Answer: a

Explanation: W hen viewed as a linear filter, the S/H has an impulse response: