A/D Converter Oversampling MCQ [Free PDF] – Objective Question Answer for A/D Converter Oversampling Quiz

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

Answer: A

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.

 

2. 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. \(σ_d^2=3σ_x^2 [1-γ_{xx} (1)]\)

Answer: B

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

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

= \(2σ_x^2 [1+γ_{xx} (1)]\).

 

3. 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. \(σ_d^2=2σ_x^2 [1+a^2]\)

Answer: C

An even better approach is to quantize the difference, d(n) = x(n)–ax(n-1), where a is a parameter selected to minimize the variance in d(n).

Therefore \(σ_d^2=σ_x^2 [1-a^2]\) .

 

4. 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. \({γ_{xx} (1)}{σ_d^2}\)

Answer: A

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 \({γ_{xx} (1)}{γ_{xx} (0)} = {γ_{xx} (1)}{σ_x^2}\).

 

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)

Answer: C

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

Answer: C

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

Answer: A

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

Answer: D

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) – \(\hat{x}(n)\), 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. \(\hat{x}(n)=\sum_{k=0}^pa_k x(n-k)\)

Answer: B

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 \(\hat{x}(n)=\sum_{k=1}^pa_k x(n-k)\).

 

10. The simplest form of differential predictive quantization is called?
A. AM
B. BM
C. DM
D. None of the mentioned

Answer: C

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

Answer: C

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

Answer: B

The simplest form of 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 predictors is used?

A. Zero-order predictor
B. Second-order predictor
C. First-order predictor
D. Third-order predictor

Answer: C

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

Answer: D

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

Answer: A

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

 

16. 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

Answer: B

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.

 

17. 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

Answer: A

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.

 

18. 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. False

Answer: A

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.

 

19. 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

Answer: C

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.

 

20. In the 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 equal to the duration of sample mode and hold mode

Answer: A

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.

 

21. In the practical A/D converters, what are the distortions and time-related degradations that 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

Answer: D

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.

 

22. 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

Answer: B

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.

 

23. 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

Answer: A

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 of an appropriate filter.

 

24. What is the process of down-sampling called?

A. Decimation
B. Fornication
C. Both Decimation & Fornication
D. None of the mentioned

Answer: A

To avoid aliasing, we first filter out the out-of-band (fl, F J 2) 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.

 

25. 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

Answer: A

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.

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