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Digital Signal Processing

The document contains a series of multiple-choice questions (MCQs) related to Digital Signal Processing (DSP), covering topics such as signals and systems, Fourier transforms, filters, and applications of DSP. Each question is followed by the correct answer, providing a comprehensive review tool for exam preparation. The content is structured to facilitate understanding of key concepts and techniques in DSP.

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0% found this document useful (0 votes)
2 views25 pages

Digital Signal Processing

The document contains a series of multiple-choice questions (MCQs) related to Digital Signal Processing (DSP), covering topics such as signals and systems, Fourier transforms, filters, and applications of DSP. Each question is followed by the correct answer, providing a comprehensive review tool for exam preparation. The content is structured to facilitate understanding of key concepts and techniques in DSP.

Uploaded by

madhut199503
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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SAMPLE MCQ QUESTION’S FOR EXAM

PREPARATION
SUBJECT: - DIGITAL SIGNAL PROCESSING

1) Signals and Systems A signal is defined as:


A function that conveys information
A type of noise
A random process
A constant function
ANSWER: - A function that conveys information
2) Which of the following is a continuous-time signal?
Speech
Discrete-time sinusoidal wave
Digital clock signal
Quantized audio sample
ANSWER: - Speech
3) A system is said to be linear if it satisfies the property of:
Additivity
Time-invariance
Frequency shifting
Quantization
ANSWER: - Additivity
4) The term "causal system" refers to:
A system that depends only on past and present inputs
A system that depends on future inputs
A system that amplifies inputs
A system with memory
ANSWER: - A system that depends only on past and present inputs
5) The impulse response of a linear time-invariant (LTI) system characterizes:
The system's response to a unit impulse
The system's frequency response
The system's output for any input signal
The system's transient behavior
ANSWER: - The system's response to a unit impulse

6) Fourier Transform The Fourier Transform converts a time-domain signal


into the:
Frequency domain
Space domain
Time-frequency domain
None of the above
ANSWER: - Frequency domain

7) For a continuous-time periodic signal, the Fourier Series represents the


signal as:
A sum of sine and cosine functions
A sum of exponential functions
A product of sinusoids
None of the above
ANSWER: - A sum of sine and cosine functions

8) The inverse Fourier Transform converts:


A frequency-domain signal to a time-domain signal
A time-domain signal to a frequency-domain signal
A time-domain signal to a Laplace-domain signal
A time-domain signal to a discrete signal
ANSWER: - A frequency-domain signal to a time-domain signal
9) The Fourier Transform of a unit impulse function δ(t) is:
1

δ(ω)
Constant zero
ANSWER: - 1

10) Which property of the Fourier Transform states that scaling in the time
domain corresponds to inverse scaling in the frequency domain?
Time-scaling
Time-shifting
Frequency-shifting
Duality
ANSWER: - Time-scaling

11) The Nyquist rate is defined as:


Twice the highest frequency of the signal
The bandwidth of the signal
Half the bandwidth of the signal
The frequency of the lowest harmonic
ANSWER: - Twice the highest frequency of the signal

12) Aliasing occurs when:


The sampling rate is lower than the Nyquist rate
The sampling rate is equal to the Nyquist rate
The sampling rate is higher than the Nyquist rate
None of the above
ANSWER: - The sampling rate is lower than the Nyquist rate
13) A practical method to avoid aliasing is to:
Use a low-pass filter before sampling
Increase the signal amplitude
Use a high-pass filter before sampling
Sample at a lower rate
ANSWER: - Use a low-pass filter before sampling
14) Which of the following is true about the reconstruction of a signal from its
samples?
Reconstruction is possible if the signal was sampled at or above the Nyquist
rate
Reconstruction is possible if the sampling rate is below the Nyquist rate
Reconstruction is only possible for discrete signals
None of the above
ANSWER: - Reconstruction is possible if the signal was sampled at or above the
Nyquist rate

15) The process of converting a continuous-time signal into a discrete-time


signal is called:
Sampling
Quantization
Modulation
Interpolation
ANSWER: - Sampling
16) Discrete-Time Signals and Systems A discrete-time signal is a function
defined at:
Integer values of time
Continuous values of time
Random time instances
None of the above
ANSWER: - Integer values of time
17) The z-transform is used to analyze:
Discrete-time systems
Continuous-time systems
Analog systems
Non-linear systems
ANSWER: - Discrete-time systems

18) The region of convergence (ROC) of a z-transform is the region where:


The z-transform converges
The system becomes stable
The signal has no poles
The z-transform is zero
ANSWER: - The z-transform converges

19) The convolution of two discrete-time signals in the time domain


corresponds to:
Multiplication in the z-domain
Division in the z-domain
Addition in the z-domain
Subtraction in the z-domain
ANSWER: - Multiplication in the z-domain

20) A discrete-time system is said to be stable if the ROC of the z-transform:


Contains the unit circle
Is outside the unit circle
Excludes the unit circle
None of the above
ANSWER: - Contains the unit circle
21) Filters A low-pass filter allows:
Low frequencies to pass and attenuates high frequencies
High frequencies to pass and attenuates low frequencies
All frequencies to pass equally
None of the above
ANSWER: - Low frequencies to pass and attenuates high frequencies

22) A high-pass filter is used to:


Block frequencies below a cutoff frequency
Pass frequencies below a cutoff frequency
Pass all frequencies equally
None of the above
ANSWER: - Block frequencies below a cutoff frequency

23) The cutoff frequency of a filter is defined as:


The frequency where the filter starts attenuating signals
The frequency where the filter passes all signals
The frequency where the filter has maximum gain
The frequency at which the filter completely blocks all signals
ANSWER: - The frequency where the filter starts attenuating signals

24) The transition band in a filter is the region where:


The filter transitions between passing and attenuating frequencies
The signal is completely passed
The signal is completely blocked
None of the above
ANSWER: - The filter transitions between passing and attenuating frequencies
25) Which of the following is NOT a commonly used filter design method?
Fourier series method
Butterworth filter
Chebyshev filter
Elliptic filter
ANSWER: - Fourier series method

26) Discrete Fourier Transform (DFT) The DFT is used to analyze:


Discrete-time signals
Continuous-time signals
Non-periodic signals
Linear systems only
ANSWER: - Discrete-time signals

27) The DFT of a signal with N samples produces:


Discrete-time signals
Continuous-time signals
Non-periodic signals
Linear systems only
ANSWER: - Discrete-time signals

28) The DFT of a signal with N samples produces:


N frequency components
N/2 frequency components
N+1 frequency components
N-1 frequency components
ANSWER: - N frequency components
29) The inverse DFT is used to:
Convert a signal from frequency-domain to time-domain
Convert a signal from time-domain to frequency-domain
Perform convolution in the time-domain
Compute the Fourier Transform
ANSWER: - Convert a signal from frequency-domain to time-domain

30) Which of the following is an advantage of the Fast Fourier Transform


(FFT)?
It reduces the computation complexity from O(N²) to O(N log N)
It requires more computation time
It cannot be used for real-time processing
It is limited to only real-valued inputs
ANSWER: - It reduces the computation complexity from O(N²) to O(N log N)

31) The DFT is inherently periodic in:


Frequency
Time
Both time and frequency
Neither time nor frequency
ANSWER: - Frequency

32) Signal Analysis Techniques The Short-Time Fourier Transform (STFT) is


used for:
Analyzing non-stationary signals
Analyzing stationary signals only
Reducing signal distortion
None of the above
ANSWER: - Analyzing non-stationary signals
33) Which of the following transforms is used for analyzing time-frequency
representation?
Wavelet Transform
Laplace Transform
Fourier Transform
Z-Transform
ANSWER: - Wavelet Transform

34) The Hilbert transform is used to obtain:


The phase information of a signal
The amplitude envelope of a signal
The frequency response of a filter
The energy of a signal
ANSWER: - The phase information of a signal

35) In signal processing, quantization refers to:


Approximating the continuous amplitude values to discrete values
Sampling the signal
Compressing the signal
Enhancing the signal
ANSWER: - Approximating the continuous amplitude values to discrete values

36) Which of the following is a method for improving the performance of a


digital filter?
Using an infinite impulse response (IIR) filter
Decreasing the order of the filter
Increasing the sampling rate
None of the above
ANSWER: - Using an infinite impulse response (IIR) filter
37) Applications of DSP Digital Signal Processing is widely used in:
All of the Below
Audio and speech processing
Image and video processing
Radar and communications
ANSWER: - All of the Below

38) Which of the following is an application of DSP in telecommunications?


All of the Below
Noise reduction
Image compression
Speech recognition
ANSWER: - All of the Below

39) Which of the following methods is used for speech coding?


All of the Below
PCM (Pulse Code Modulation)
DPCM (Differential Pulse Code Modulation)
LPC (Linear Predictive Coding)
ANSWER: - All of the Below

40) In digital image processing, which operation is often used for edge
detection?
Convolution with a kernel
Smoothing
Histogram equalization
None of the above
ANSWER: - Convolution with a kernel
41) In audio processing, the term "reverberation" refers to:
The direct sound
Reflected sound that arrives at the listener after a delay
Noise in the background
None of the above
ANSWER: - Reflected sound that arrives at the listener after a delay

42) Advanced Topics in DSP Which of the following is a characteristic of FIR


(Finite Impulse Response) filters?
They can have feedback loops.
They can be designed to have a linear phase response.
They always have a constant gain.
They are unstable.
ANSWER: - They can be designed to have a linear phase response.

43) What does the term "zero-padding" refer to in the context of DFT?
Adding zeros to the beginning of the signal
Adding zeros to the end of the signal to increase its length
Removing zeros from the signal
None of the above
ANSWER: - Adding zeros to the end of the signal to increase its length

44) The term "dithering" in quantization is used to:


Increase the bit depth
Reduce quantization error
Enhance signal-to-noise ratio
None of the above
ANSWER: - Reduce quantization error
45) In digital filters, the term "phase delay" refers to:
The delay in the amplitude response
The shift of the output signal in time with respect to the input signal
The delay caused by quantization
None of the above
ANSWER: - The shift of the output signal in time with respect to the input
signal

46) What is the primary purpose of a decimator in DSP?


To increase the sampling rate
To decrease the sampling rate
To filter out high frequencies
To add noise
ANSWER: - To decrease the sampling rate

47) Basic Concepts The main purpose of using DSP in audio applications is to:
Increase the volume
Modify and enhance the sound quality
Convert audio to video
None of the above
ANSWER: - Modify and enhance the sound quality

48) Which of the following statements about analog and digital signals is true?
Digital signals are continuous in nature.
Analog signals are discrete in nature.
Digital signals can be processed using computers.
Both A and C
ANSWER: - Digital signals can be processed using computers.
49) The quantization process in DSP results in:
Loss of information
Increased accuracy
Enhanced signal integrity
None of the above
ANSWER: - Loss of information

50) What is the primary advantage of digital over analog signals?


Digital signals are more easily degrade
Digital signals can be processed and stored more easily.
Analog signals require less bandwidth.
None of the above
ANSWER: - Digital signals can be processed and stored more easily.

51) Which of the following components is essential in a digital audio


workstation (DAW)?
MIDI controller
Microphone
Audio interface
All of the above
ANSWER: - All of the above

52) Filters and Their Applications In signal processing, what does an IIR filter
stand for?
Infinite Impulse Response
Intermediate Impulse Response
Integrated Impulse Response
Indeterminate Impulse Response
ANSWER: - Infinite Impulse Response
53) What type of filter is designed to eliminate a specific frequency?
Band-pass filter
Band-stop filter
Low-pass filter
High-pass filter
ANSWER: - Band-stop filter

54) In filter design, the term "roll-off" refers to:


The rate at which the filter attenuates signals beyond the cutoff frequency
The maximum gain of the filter
The bandwidth of the filter
None of the above
ANSWER: - The rate at which the filter attenuates signals beyond the cutoff
frequency

55) Which filter is known for having a maximally flat frequency response in the
passband?
Chebyshev filter
Butterworth filter
Bessel filter
Elliptic filter
ANSWER: - Butterworth filter

56) Which technique is used to obtain the frequency response of a digital


filter?
Direct computation
Fourier Transform
z-Transform
Both B and C
ANSWER: - Both B and C
57) DSP Hardware and Software Which of the following is commonly used for
implementing DSP algorithms?
Digital signal processors (DSPs)
General-purpose microprocessors
FPGAs (Field-Programmable Gate Arrays)
All of the above
ANSWER: - All of the above

58) The primary advantage of using a DSP chip over a general-purpose CPU is:
Lower cost
More power consumption
Enhanced performance for signal processing tasks
Easier programming
ANSWER: - Enhanced performance for signal processing tasks

59) MATLAB is commonly used in DSP for:


Hardware design
Algorithm simulation and prototyping
Low-level programming
None of the above
ANSWER: - Algorithm simulation and prototyping

60) What does "real-time processing" mean in the context of DSP?


Processing that occurs after the data is stored
Processing that occurs with a time delay
Processing that occurs as the input data is received
Processing that is done offline
ANSWER: - Processing that occurs as the input data is received
61) 2. Which of the following is a limitation of the Z-transform?
Limited to causal systems
Only works for periodic signals
Cannot handle non-linear systems
Not applicable to continuous-time signals
ANSWER: - Limited to causal systems

62) 4. In an FIR filter, the output is determined by...


The current and past input values
Only the past input values
The current and past output values
Only the current input value
ANSWER: - The current and past input values

63) 6. What is the primary purpose of a decimation process in DSP?


To reduce the number of samples
To increase the number of samples
To filter unwanted frequencies
To modify the signal amplitude
ANSWER: - To reduce the number of samples

64) 7. Which is the main advantage of an FIR filter over an IIR filter?
FIR filters have linear phase response
FIR filters are computationally simpler
IIR filters are more stable
IIR filters are easier to design
ANSWER: - FIR filters have linear phase response
65) 9. The system described by the difference equation y[n] = 0.5y[n-1] + x[n]
is...
Recursive
Non-recursive
Linear
Time-invariant
ANSWER: - Recursive

66) 11. What does the term "aliasing" refer to in signal processing?
A signal's frequency component being misrepresented
The conversion of continuous-time signals to discrete-time
The process of filtering a signal
The reconstruction of a signal from its samples
ANSWER: - A signal's frequency component being misrepresented

67) 17. What is the Z-transform of a unit impulse function?


1
z
1/z
0
ANSWER: - 1

68) 19. The inverse Z-transform is used to...


Convert from the Z-domain to the time domain
Convert from the time domain to the Z-domain
Filter signals in the Z-domain
Analyze stability in the time domain
ANSWER: - Convert from the Z-domain to the time domain
69) 21. The transfer function of a system is defined as...
The ratio of the output to the input in the Z-domain
The ratio of the input to the output in the time domain
The system's frequency response
The difference equation of the system
ANSWER: - The ratio of the output to the input in the Z-domain

70) 22. What does the term "sampling" refer to in DSP?


Converting a continuous-time signal to a discrete-time signal
Increasing the frequency of a signal
Filtering a signal in the time domain
Modulating a signal to a higher frequency
ANSWER: - Converting a continuous-time signal to a discrete-time signal

71) 26. The main application of an FIR filter is...


To remove noise from signals
To amplify signals
To create delay in signals
To modify frequency content
ANSWER: - To remove noise from signals

72) 27. What is the primary purpose of the Fast Fourier Transform (FFT)?
Speed up the process of computing the DFT
Compute continuous Fourier Transforms
Convert analog signals to digital signals
Filter signals in the frequency domain
ANSWER: - Speed up the process of computing the DFT
73) 33. Which of the following is the primary goal of digital filtering?
To reduce noise and unwanted components from a signal
To increase the amplitude of a signal
To modulate a signal's frequency
To convert an analog signal into a digital signal
ANSWER: - To reduce noise and unwanted components from a signal

74) 34. A discrete-time system is said to be linear if...


The output is proportional to the input
The system is time-varying
The system only uses past inputs
The system is non-recursive
ANSWER: - The output is proportional to the input

75) 37. In the Z-domain, poles and zeros are used to...
Determine the stability and frequency response of the system
Represent the magnitude and phase of the system
Filter the signal in the time domain
None of the above
ANSWER: - Determine the stability and frequency response of the system

76) 38. The purpose of an anti-aliasing filter is to...


Filter high frequencies before sampling
Filter low frequencies before sampling
Smooth the signal after sampling
Remove noise from a signal
ANSWER: - Filter high frequencies before sampling
77) 40. In a system characterized by y[n] = x[n] + x[n-1], the system is...
Causal and linear
Non-causal and non-linear
Time-invariant and non-linear
Time-varying and causal
ANSWER: - Causal and linear

78) 8. The Discrete Fourier Transform (DFT) is used to analyze...


Discrete signals
Continuous signals
Analog signals
Mixed signals
ANSWER: - Discrete signals

79) 13. What is the main advantage of an IIR filter over an FIR filter?
IIR filters are computationally more efficient
FIR filters have finite impulse response
IIR filters are easier to design
FIR filters are more stable
ANSWER: - IIR filters are computationally more efficient

80) 15. In a discrete-time system, stability is determined by...


The system's transfer function
The impulse response
The number of poles and zeros
The frequency response
ANSWER: - The system's transfer function
81) 16. The Fast Fourier Transform (FFT) is an efficient algorithm for...
Computing the Discrete Fourier Transform (DFT)
Converting analog signals to digital
Filtering signals
Sampling signals
ANSWER: - Computing the Discrete Fourier Transform (DFT)

82) 23. The main purpose of a low-pass filter is to...


Pass low frequencies and attenuate high frequencies
Pass high frequencies and attenuate low frequencies
Pass all frequencies equally
Pass only the highest frequencies
ANSWER: - Pass low frequencies and attenuate high frequencies

83) 25. The discrete-time Fourier transform (DTFT) is used to analyze...


Discrete signals
Continuous-time signals
Analog signals
Mixed signals
ANSWER: - Discrete signals

84) 29. The system described by the equation y[n] = x[n] + 0.5x[n-1] is...
Non-recursive
Recursive
Linear
Non-linear
ANSWER: - Non-recursive
85) 30. Which of the following is a method of performing time-domain
convolution?
Using the overlap-and-add method
Using a Fourier series
Using the Laplace transform
Using the inverse Z-transform
ANSWER: - Using the overlap-and-add method

86) 32. In the context of signal processing, "sampling" refers to...


The process of converting continuous signals into discrete signals
The continuous-time representation of a signal
The conversion of a signal to a higher frequency
The process of signal quantization
ANSWER: - The process of converting continuous signals into discrete signals

87) 39. What does "time-invariance" in a system mean?


The system's response does not change over time
The system's output changes over time
The system operates only in real time
The system's frequency response changes with time
ANSWER: - The system's response does not change over time

88) 1. What is the Fourier Transform of a constant signal?


A Dirac delta function
1
0
A constant
ANSWER: - A Dirac delta function
89) 10. A band-pass filter allows frequencies...
Within a specific frequency range
Above a certain cutoff
Below a certain cutoff
All of the above
ANSWER: - Within a specific frequency range

90) 14. The term "digital signal processing" refers to...


Processing of discrete-time signals
Processing of continuous signals
Processing of analog signals
Processing of all types of signals
ANSWER: - Processing of discrete-time signals

91) 18. Which type of filter has the characteristic of allowing frequencies
within a specified range?
Low-pass filter
High-pass filter
Band-pass filter
Notch filter
ANSWER: - Band-pass filter

92) 20. In DSP, what does the term "convolution" refer to?
The addition of signals
The integration of signals
The overlap-and-add process for filtering
The multiplication of signals
ANSWER: - The overlap-and-add process for filtering
93) 24. What is a key property of a stable system in DSP?
It does not produce bounded outputs for bounded inputs
Its impulse response grows without bound
It produces bounded outputs for bounded inputs
Its transfer function is non-linear
ANSWER: - It produces bounded outputs for bounded inputs

94) 28. A system is said to be causal if...


The output depends only on future inputs
The output depends on past and current inputs
The output depends only on past inputs
The system's transfer function is stable
ANSWER: - The output depends only on past inputs

95) 31. What is the primary characteristic of a low-pass filter?


It passes high-frequency components and blocks low-frequency components
It blocks all frequencies
It passes low-frequency components and blocks high-frequency components
It passes all frequencies equally
ANSWER: - It passes low-frequency components and blocks high-frequency
components

96) 35. In a digital signal processing system, the word "quantization" refers to...
Reducing the frequency of a signal
Sampling a signal
Representing a signal with finite precision
Filtering the signal
ANSWER: - Representing a signal with finite precision
97) 3. What does the Nyquist criterion for stability refer to?
Relationship between frequency and time
The maximum bandwidth of a signal
The sampling theorem
The stability of a discrete-time system
ANSWER: - The stability of a discrete-time system

98) 5. Which of the following is a type of frequency response used in DSP?


High-pass filter
Low-pass filter
Band-pass filter
All of the above
ANSWER: - All of the above

99) 12. Which method is used to reduce aliasing in digital signal processing?
Decimation
Over-sampling
Anti-aliasing filter
All of the above
ANSWER: - All of the above

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