Q.1 What is a signal in the context of signal processing?
A device that measures electrical current
A function that conveys information
A type of digital memory
A physical wire that carries electricity
Explanation - A signal is any time-varying quantity that carries information about the state of a system.
Correct answer is: A function that conveys information
Q.2 Which of the following is a continuous-time signal?
x[n] = sin(πn/4)
x(t) = e^{-2t}
x[k] = cos(k/2)
x[n] = 1/(n^2 + 1)
Explanation - Continuous-time signals are defined for all real time values, whereas the other options are defined only for integer indices.
Correct answer is: x(t) = e^{-2t}
Q.3 Which of the following is a discrete-time signal?
x(t) = cos(2πt)
x[n] = cos(πn/6)
x(t) = 5
x[n] = t^2
Explanation - Discrete-time signals are defined only at integer sample indices; x[n] meets that criterion.
Correct answer is: x[n] = cos(πn/6)
Q.4 Sampling an analog signal involves which of the following?
Converting it into a digital signal using a quantizer
Measuring its amplitude at discrete times
Filtering out high-frequency components
Increasing its amplitude uniformly
Explanation - Sampling captures the signal’s amplitude at equally spaced time intervals, producing a discrete-time representation.
Correct answer is: Measuring its amplitude at discrete times
Q.5 What does the Nyquist theorem state?
A signal can be reconstructed from its samples if the sampling rate is at least twice the highest frequency present.
All signals can be perfectly compressed by sampling at any rate.
The sampling rate must equal the signal’s amplitude.
Only periodic signals can be sampled.
Explanation - Nyquist's theorem dictates the minimum sampling frequency required to avoid loss of information.
Correct answer is: A signal can be reconstructed from its samples if the sampling rate is at least twice the highest frequency present.
Q.6 Aliasing occurs when a signal is sampled at what rate relative to its bandwidth?
Above twice the bandwidth
Exactly twice the bandwidth
Below twice the bandwidth
Any rate, as long as it is high enough
Explanation - Sampling below twice the maximum frequency causes different frequency components to become indistinguishable.
Correct answer is: Below twice the bandwidth
Q.7 Which of the following is an example of a low-pass filter?
A circuit that blocks frequencies above 1 kHz
A circuit that blocks frequencies below 1 kHz
A circuit that blocks all frequencies
A circuit that blocks only DC
Explanation - A low-pass filter allows low frequencies to pass while attenuating high frequencies.
Correct answer is: A circuit that blocks frequencies above 1 kHz
Q.8 Which function represents an impulse in continuous time?
δ(t)
u(t)
cos(t)
t^2
Explanation - δ(t) (Dirac delta) is an impulse function that is zero everywhere except at t=0, where it is infinite, and its integral over all time is 1.
Correct answer is: δ(t)
Q.9 The convolution of two signals x(t) and h(t) is denoted as:
x(t) * h(t)
x(t) / h(t)
x(t) + h(t)
x(t) – h(t)
Explanation - Convolution is the integral of the product of one signal with the time-reversed and shifted version of the other.
Correct answer is: x(t) * h(t)
Q.10 What does the Fourier transform of a time-limited signal provide?
Its time-domain representation
Its frequency spectrum
Its phase only
Its amplitude squared
Explanation - The Fourier transform maps a time-domain signal into its frequency-domain representation, showing how much of each frequency is present.
Correct answer is: Its frequency spectrum
Q.11 Which of these is a discrete-time Fourier transform (DTFT) of a unit impulse sequence δ[n]?
1
0
π
e^{jω}
Explanation - The DTFT of δ[n] is 1 for all frequencies, because δ[n] has no frequency content except a constant.
Correct answer is: 1
Q.12 In a sampled system, what is the significance of the term 'bandwidth'?
Maximum sample rate
Maximum frequency present in the signal
Number of samples per second
Signal amplitude
Explanation - Bandwidth refers to the highest frequency component that the signal contains.
Correct answer is: Maximum frequency present in the signal
Q.13 Which of these is a characteristic of a band-limited signal?
It contains only one frequency
It has infinite bandwidth
It has no frequency components above a certain limit
It is a random noise
Explanation - Band-limited signals are constrained to a finite frequency range.
Correct answer is: It has no frequency components above a certain limit
Q.14 The Laplace transform is primarily used for analyzing which type of systems?
Discrete-time systems only
Linear time-invariant continuous-time systems
Non-linear systems
Random processes
Explanation - The Laplace transform helps solve differential equations for LTI continuous-time systems.
Correct answer is: Linear time-invariant continuous-time systems
Q.15 Which of the following best describes an 'analog-to-digital converter (ADC) gain'?
The amplification applied before sampling
The maximum input voltage range
The number of bits used for quantization
The sampling frequency
Explanation - ADC gain refers to the pre-sampling amplification of the signal to match the ADC's input range.
Correct answer is: The amplification applied before sampling
Q.16 Which statement about quantization is correct?
It introduces noise into the signal
It can be avoided by using infinite resolution
It is the same as sampling
It reduces the bandwidth of a signal
Explanation - Quantization errors arise from rounding sample values to discrete levels, introducing noise.
Correct answer is: It introduces noise into the signal
Q.17 What is the primary purpose of a digital low-pass filter?
To increase high-frequency noise
To remove high-frequency components from a signal
To convert analog signals to digital
To compress data
Explanation - Digital low-pass filtering attenuates frequencies above a specified cutoff, preserving low-frequency content.
Correct answer is: To remove high-frequency components from a signal
Q.18 Which of the following is the correct definition of 'alias frequency'?
A frequency that appears due to sampling at too high a rate
The true frequency of a signal
A frequency that appears because the sampling rate is too low
A frequency that is always zero
Explanation - Alias frequencies are spurious frequencies resulting from undersampling.
Correct answer is: A frequency that appears because the sampling rate is too low
Q.19 Which transform is used for analyzing non-stationary signals?
Fourier Transform
Short-Time Fourier Transform
Z-Transform
Laplace Transform
Explanation - The Short-Time Fourier Transform (STFT) provides frequency content over time, suitable for non-stationary signals.
Correct answer is: Short-Time Fourier Transform
Q.20 What does the 'unit step function' u(t) equal for t < 0?
0
1
-1
Undefined
Explanation - u(t) is 0 for negative time and 1 for non-negative time.
Correct answer is: 0
Q.21 Which of the following is a property of convolution in the time domain?
It is commutative
It is associative
It is distributive over addition
All of the above
Explanation - Convolution is commutative, associative, and distributive.
Correct answer is: All of the above
Q.22 The discrete Fourier transform (DFT) of a real even-symmetric sequence is:
Real and even-symmetric
Complex and odd-symmetric
Real and odd-symmetric
Complex and even-symmetric
Explanation - For real even-symmetric input, the DFT is real and even-symmetric.
Correct answer is: Real and even-symmetric
Q.23 Which of the following is NOT a common sampling strategy?
Uniform sampling
Non-uniform sampling
Random sampling
Analog sampling
Explanation - Sampling inherently produces discrete data; analog sampling does not exist.
Correct answer is: Analog sampling
Q.24 In digital signal processing, what does the 'frequency resolution' refer to?
The difference in frequency between adjacent DFT bins
The maximum frequency a system can handle
The sampling rate
The bandwidth of a filter
Explanation - Frequency resolution = Fs / N, where Fs is sampling frequency and N is number of points.
Correct answer is: The difference in frequency between adjacent DFT bins
Q.25 What is the main advantage of a 'finite impulse response (FIR) filter'?
Infinite stability
Phase linearity
Lower computational cost
Higher order than IIR
Explanation - FIR filters can be designed with linear phase, which is desirable for many applications.
Correct answer is: Phase linearity
Q.26 Which of the following is a key property of the Dirac delta function?
Its integral over all time is zero
It is non-zero only at t = 0
It has infinite energy
Its Fourier transform is a constant
Explanation - The Fourier transform of δ(t) is 1 for all frequencies.
Correct answer is: Its Fourier transform is a constant
Q.27 What does the 'Z-transform' primarily analyze?
Continuous-time signals
Discrete-time signals
Random processes
Frequency domain of analog signals
Explanation - The Z-transform converts discrete-time signals into the complex frequency domain.
Correct answer is: Discrete-time signals
Q.28 Which of these is a characteristic of an 'ideal low-pass filter'?
Infinite impulse response
Perfect attenuation of frequencies above cutoff
Linear phase
All of the above
Explanation - An ideal low-pass filter would have a rectangular frequency response, infinite impulse response, and linear phase.
Correct answer is: All of the above
Q.29 In the context of signal processing, what is the term 'aliasing' specifically referring to?
The loss of signal amplitude
The folding of higher frequencies into lower frequencies during sampling
The distortion of the time domain signal
The increase in signal-to-noise ratio
Explanation - Aliasing is when higher frequency components overlap into lower frequency bands due to insufficient sampling.
Correct answer is: The folding of higher frequencies into lower frequencies during sampling
Q.30 A signal that is constant over time is called:
Periodic
Stochastic
Deterministic
Zero-mean
Explanation - A deterministic signal has a known, fixed value at each time instance.
Correct answer is: Deterministic
Q.31 Which of these represents the magnitude spectrum of a sinusoid?
Two spikes at ±f0
A continuous band from 0 to f0
A single spike at zero frequency
No spikes, only noise
Explanation - A sinusoid's Fourier transform has two delta peaks at positive and negative frequencies.
Correct answer is: Two spikes at ±f0
Q.32 A digital system that implements a filter with feedback is called a:
FIR filter
IIR filter
FFT
DAC
Explanation - IIR (Infinite Impulse Response) filters use feedback, giving potentially infinite impulse response.
Correct answer is: IIR filter
Q.33 Which of the following best describes a 'bandpass filter'?
Blocks all frequencies
Passes frequencies above a certain threshold
Passes frequencies within a specific range
Passes only DC component
Explanation - Bandpass filters allow a band of frequencies to pass while attenuating others.
Correct answer is: Passes frequencies within a specific range
Q.34 The 'sampling theorem' ensures that a bandlimited signal can be reconstructed from its samples if the sampling frequency:
Is less than the bandwidth
Is equal to the bandwidth
Is greater than twice the bandwidth
Is arbitrary
Explanation - To reconstruct without aliasing, Fs must be > 2*B.
Correct answer is: Is greater than twice the bandwidth
Q.35 A 'digital-to-analog converter (DAC)' performs what function?
Converts analog signals to digital
Converts digital signals to analog
Amplifies signals
Filters signals
Explanation - DAC outputs a continuous voltage proportional to the digital input.
Correct answer is: Converts digital signals to analog
Q.36 Which of the following is NOT an advantage of digital signal processing?
Precision control
Programmability
Inherent noise
Complex algorithm implementation
Explanation - Digital systems generally have lower noise compared to analog.
Correct answer is: Inherent noise
Q.37 In signal analysis, a 'power spectral density' (PSD) represents:
The signal's amplitude over time
The distribution of power into frequency components
The phase of the signal
The energy of the signal
Explanation - PSD indicates how power is distributed across frequency.
Correct answer is: The distribution of power into frequency components
Q.38 The 'Hilbert transform' is used to compute:
The analytic signal and envelope
The Fourier transform
The convolution
The Laplace transform
Explanation - Hilbert transform gives the quadrature component used to form the analytic signal.
Correct answer is: The analytic signal and envelope
Q.39 What is the main difference between an 'analog filter' and a 'digital filter'?
Analog filters have no cutoff frequency
Digital filters process continuous-time signals directly
Analog filters use physical components; digital filters use algorithms
Digital filters cannot have linear phase
Explanation - Analog filters rely on resistors, capacitors, etc., while digital filters are implemented in software or hardware using discrete data.
Correct answer is: Analog filters use physical components; digital filters use algorithms
Q.40 The term 'anti-aliasing filter' refers to:
A filter applied after sampling
A filter that increases sampling frequency
A filter that removes high-frequency components before sampling
A filter that converts analog to digital
Explanation - Anti-aliasing filters prevent high frequencies from causing aliasing during sampling.
Correct answer is: A filter that removes high-frequency components before sampling
Q.41 What does the term 'spectral leakage' describe?
Energy loss due to filter attenuation
Discrepancy between actual and measured spectra due to windowing
Signal distortion in the time domain
Noise introduced by ADC
Explanation - Spectral leakage occurs when the finite observation window spreads energy across frequencies.
Correct answer is: Discrepancy between actual and measured spectra due to windowing
Q.42 Which of these is a typical application of the Fast Fourier Transform (FFT)?
Real-time audio processing
Analog circuit design
Mechanical engineering
Structural analysis
Explanation - FFT is widely used in real-time signal processing tasks like audio.
Correct answer is: Real-time audio processing
Q.43 In the context of DSP, what is an 'overlap-add' method used for?
Filtering long signals using block processing
Sampling signals at a higher rate
Quantizing signals to lower precision
Designing analog filters
Explanation - Overlap-add is a technique for efficient convolution of long signals by processing blocks and summing overlaps.
Correct answer is: Filtering long signals using block processing
Q.44 Which of the following is a common window function?
Rectangular
Gaussian
Both A and B
None of the above
Explanation - Both rectangular and Gaussian are widely used window functions.
Correct answer is: Both A and B
Q.45 A 'rectangular pulse' in time domain corresponds to:
A sinc function in frequency
A delta function in frequency
A cosine function in frequency
No specific frequency representation
Explanation - The Fourier transform of a rectangular pulse is a sinc function.
Correct answer is: A sinc function in frequency
Q.46 Which of the following best describes 'phase response' of a filter?
Amplitude variation across frequencies
Change of signal delay with frequency
Amount of power attenuation
Filter's stability margin
Explanation - Phase response indicates how the phase shift varies with frequency.
Correct answer is: Change of signal delay with frequency
Q.47 A 'time-domain' representation of a signal is primarily concerned with:
Signal values over frequency
Signal values over time
Signal phase only
Signal amplitude only
Explanation - Time-domain shows how a signal evolves with time.
Correct answer is: Signal values over time
Q.48 Which of the following is a property of the Fourier transform of a real even-symmetric function?
It is purely imaginary
It is real and even-symmetric
It is complex with no symmetry
It is zero everywhere
Explanation - Even real functions produce real, even Fourier transforms.
Correct answer is: It is real and even-symmetric
Q.49 What does the 'inverse FFT' compute?
The magnitude spectrum
The time-domain signal from frequency bins
The phase spectrum
The convolution of two signals
Explanation - Inverse FFT transforms frequency-domain data back to time domain.
Correct answer is: The time-domain signal from frequency bins
Q.50 Which of the following statements about 'random processes' is true?
They have deterministic outcomes
Their statistical properties are independent of time
They are always band-limited
They cannot be measured
Explanation - Stationary random processes have time-invariant statistics.
Correct answer is: Their statistical properties are independent of time
Q.51 The 'zero-padding' technique is used in DFT to:
Decrease frequency resolution
Increase frequency resolution
Reduce the number of samples
Increase the signal amplitude
Explanation - Zero-padding adds zeros to a signal to interpolate the DFT, effectively refining frequency resolution.
Correct answer is: Increase frequency resolution
Q.52 In a digital filter, a 'recursive' structure refers to:
Using only past input samples
Using both past input and output samples
Using only future input samples
Using a fixed set of coefficients
Explanation - Recursive (IIR) filters use feedback from past outputs.
Correct answer is: Using both past input and output samples
Q.53 Which of the following describes an 'analog signal'?
Defined at discrete time intervals
Defined for all time values and can take any real value
Can only be binary
Cannot be processed by computers
Explanation - Analog signals vary continuously over time.
Correct answer is: Defined for all time values and can take any real value
Q.54 What is the purpose of a 'quantizer' in an ADC?
To sample the signal at high rate
To convert continuous amplitude to discrete levels
To filter out high frequencies
To increase signal amplitude
Explanation - A quantizer maps a continuous range to discrete values.
Correct answer is: To convert continuous amplitude to discrete levels
Q.55 A 'frequency spectrum' of a pure cosine wave contains:
Two discrete spectral lines at ±f0
A continuous band from 0 to f0
Only a DC component
A single peak at zero frequency
Explanation - Cosine is the sum of two complex exponentials at ±f0.
Correct answer is: Two discrete spectral lines at ±f0
Q.56 Which transform is most appropriate for analyzing signals in the complex frequency domain?
Fourier Transform
Laplace Transform
Z-Transform
Both B and C
Explanation - Laplace (continuous) and Z-Transform (discrete) analyze complex frequency characteristics.
Correct answer is: Both B and C
Q.57 In a convolution operation between signals x[n] and h[n], what does the result represent?
The product of the two signals
The sum of the two signals
The output of an LTI system with impulse response h[n]
The difference between the two signals
Explanation - Convolution of input x[n] with impulse response h[n] yields system output.
Correct answer is: The output of an LTI system with impulse response h[n]
Q.58 Which of the following is a characteristic of a 'bandstop filter'?
Passes only high frequencies
Blocks a specific frequency band
Passes only low frequencies
Passes all frequencies equally
Explanation - Bandstop filters attenuate a defined frequency band while passing others.
Correct answer is: Blocks a specific frequency band
Q.59 The 'sinc' function is the Fourier transform of:
A Gaussian pulse
A rectangular pulse
A sinusoid
A delta function
Explanation - The sinc function arises from the FT of a rectangular window.
Correct answer is: A rectangular pulse
Q.60 Which of the following best describes 'phase distortion'?
Amplitude changes across frequencies
Phase changes that vary non-linearly with frequency
Decrease in filter stability
Increase in signal power
Explanation - Phase distortion occurs when the filter’s phase response is not linear.
Correct answer is: Phase changes that vary non-linearly with frequency
Q.61 In DSP, the 'Hamming window' primarily:
Increases spectral leakage
Reduces spectral leakage
Changes the DC component
Increases the bandwidth
Explanation - Hamming windows smooth the edges of a finite sequence, reducing leakage.
Correct answer is: Reduces spectral leakage
Q.62 Which of the following is the correct definition of 'bandwidth' of a filter?
The difference between the highest and lowest frequencies the filter can process
The highest frequency the filter can process
The lowest frequency the filter can process
The time duration of the filter's impulse response
Explanation - Bandwidth is the frequency range over which the filter passes signals.
Correct answer is: The difference between the highest and lowest frequencies the filter can process
Q.63 Which of these is NOT a common digital filter design technique?
Windowing
Butterworth
Chebyshev
Rectifier
Explanation - Rectifier is an electronic component, not a filter design method.
Correct answer is: Rectifier
Q.64 What does the 'spectral centroid' measure in audio signals?
Average frequency weighted by magnitude
Maximum frequency present
Total energy
Phase shift
Explanation - Spectral centroid is a weighted average of frequencies, indicating perceived brightness.
Correct answer is: Average frequency weighted by magnitude
Q.65 Which of the following is a key feature of 'adaptive filters'?
Fixed coefficients
Coefficients change over time
Only linear phase
Only used in analog circuits
Explanation - Adaptive filters adjust coefficients to adapt to changing signal statistics.
Correct answer is: Coefficients change over time
Q.66 A 'phase-locked loop (PLL)' is primarily used for:
Generating random numbers
Synchronizing a local oscillator to an input signal
Filtering high-frequency noise
Amplifying signals
Explanation - PLL locks the phase of an oscillator to match a reference signal.
Correct answer is: Synchronizing a local oscillator to an input signal
Q.67 Which of the following is a valid reason for using 'oversampling' in ADCs?
To increase quantization noise
To simplify anti-aliasing filtering
To reduce sampling frequency
To decrease dynamic range
Explanation - Higher sampling rates allow gentler anti-aliasing filters.
Correct answer is: To simplify anti-aliasing filtering
Q.68 The 'zero-crossing rate' of a signal is commonly used to quantify:
Signal amplitude
Signal frequency content
Signal power
Signal delay
Explanation - Zero-crossing rate correlates with the average frequency of a waveform.
Correct answer is: Signal frequency content
Q.69 In digital image processing, which of the following is considered a 'spatial domain' operation?
Fourier transform
Histogram equalization
Wavelet transform
All of the above
Explanation - Histogram equalization operates directly on pixel values in the spatial domain.
Correct answer is: Histogram equalization
Q.70 Which of these signals is considered 'white noise'?
A sine wave
A random signal with constant power spectral density
A random signal with decaying power spectral density
A constant signal
Explanation - White noise has equal power across frequencies.
Correct answer is: A random signal with constant power spectral density
Q.71 What is the purpose of a 'window function' in FFT analysis?
To extend the signal length
To reduce spectral leakage
To increase frequency resolution
To sample at a higher rate
Explanation - Windowing tapers the signal edges, reducing discontinuities.
Correct answer is: To reduce spectral leakage
Q.72 Which of the following is a measure of how well a filter preserves the relative timing between frequency components?
Amplitude response
Phase linearity
Group delay
All of the above
Explanation - Group delay is the derivative of phase vs frequency, indicating timing preservation.
Correct answer is: Group delay
Q.73 The 'modulation theorem' states that multiplying a signal by a sinusoid:
Shifts its frequency spectrum
Attenuates its amplitude
Increases its bandwidth by zero
Has no effect on the spectrum
Explanation - Multiplication in time corresponds to convolution in frequency, shifting components.
Correct answer is: Shifts its frequency spectrum
Q.74 A 'Hilbert transformer' is used to generate:
The analytic signal of a real input
The inverse Fourier transform
The frequency response of a filter
The phase only of a signal
Explanation - Hilbert transformer creates a 90° phase-shifted version for analytic signal construction.
Correct answer is: The analytic signal of a real input
Q.75 In a 'phase-locked loop', the 'loop filter' is primarily used to:
Filter the input signal
Control the oscillator's frequency
Measure the phase difference
Convert analog to digital
Explanation - The loop filter smooths the phase error and drives the voltage-controlled oscillator.
Correct answer is: Control the oscillator's frequency
Q.76 Which of the following describes a 'saturated' amplifier?
It amplifies signals without distortion
It clips the output when input exceeds a threshold
It has infinite gain
It only amplifies low frequencies
Explanation - Saturation occurs when output is limited, causing clipping.
Correct answer is: It clips the output when input exceeds a threshold
Q.77 The 'spectrum of a signal' can be obtained via which mathematical operation?
Convolution
Differentiation
Fourier transform
Integration
Explanation - Fourier transform converts time-domain signals to frequency domain.
Correct answer is: Fourier transform
Q.78 Which of the following is a common use of a 'Hilbert transform' in audio processing?
Detecting zero crossings
Creating stereo panning
Computing the amplitude envelope
Amplifying bass frequencies
Explanation - The analytic signal's magnitude gives the envelope.
Correct answer is: Computing the amplitude envelope
Q.79 A 'block diagram' in signal processing typically represents:
The flow of signals through various operations
The exact numeric values of signals
The physical layout of an electronic circuit
The frequency response only
Explanation - Block diagrams show how signals are processed step-by-step.
Correct answer is: The flow of signals through various operations
Q.80 Which of the following best describes 'sinc interpolation'?
Using a rectangular window to interpolate data
Reconstructing a bandlimited signal from its samples
Adding noise to a signal
Compressing data by averaging samples
Explanation - Sinc interpolation applies the sinc function to each sample to reconstruct continuous signal.
Correct answer is: Reconstructing a bandlimited signal from its samples
Q.81 Which of the following is a 'recursive' filter implementation?
Using a finite number of past inputs only
Using past outputs in addition to past inputs
Using future inputs only
Using only the current input
Explanation - Recursive filters (IIR) use feedback from previous outputs.
Correct answer is: Using past outputs in addition to past inputs
Q.82 A 'frequency-modulated' signal has:
Amplitude varying with time
Frequency varying with time
Phase varying with time
Both amplitude and frequency constant
Explanation - In FM, the carrier frequency changes according to the message signal.
Correct answer is: Frequency varying with time
Q.83 The 'Nyquist rate' is defined as:
Half the sampling rate
The maximum sampling rate
Twice the highest signal frequency
The frequency of the Nyquist frequency
Explanation - Nyquist rate ensures no aliasing when sampling.
Correct answer is: Twice the highest signal frequency
Q.84 Which of the following best explains 'aliasing' in the frequency domain?
High frequencies are shifted into lower frequency bands during sampling
Low frequencies are attenuated during sampling
The signal is lost during sampling
Sampling increases bandwidth
Explanation - Aliasing occurs when high-frequency components fold back into the baseband.
Correct answer is: High frequencies are shifted into lower frequency bands during sampling
Q.85 Which of these is an example of a 'random noise' process?
A sine wave
A white noise signal
A step function
A cosine wave
Explanation - White noise is a random process with uniform power distribution.
Correct answer is: A white noise signal
Q.86 The 'Fourier series' is used to represent:
Periodic continuous-time signals as sums of sinusoids
Aperiodic signals in frequency domain
The convolution of two signals
The time delay of a signal
Explanation - Fourier series decomposes periodic signals into harmonics.
Correct answer is: Periodic continuous-time signals as sums of sinusoids
Q.87 Which of the following is a key difference between analog and digital filtering?
Analog filters can be implemented in software
Digital filters can have fractional delay
Analog filters are always linear
Digital filters can be implemented in hardware or software
Explanation - Digital filters are flexible and can be coded or built on DSP chips.
Correct answer is: Digital filters can be implemented in hardware or software
Q.88 In signal processing, what does the 'power spectral density' (PSD) describe?
The distribution of power across frequencies
The amplitude envelope of a signal
The phase shift at different frequencies
The time delay of a signal
Explanation - PSD indicates how power is spread over frequency.
Correct answer is: The distribution of power across frequencies
Q.89 Which of these is an example of 'digital signal modulation'?
Amplitude shift keying
Frequency shift keying
Both A and B
None of the above
Explanation - ASK and FSK are common digital modulation schemes.
Correct answer is: Both A and B
Q.90 In an 'analog-to-digital converter (ADC)', what role does the 'reference voltage' play?
It sets the maximum input voltage the ADC can handle
It determines the sampling rate
It defines the number of bits of resolution
It acts as the output voltage
Explanation - Reference voltage establishes the ADC's full-scale range.
Correct answer is: It sets the maximum input voltage the ADC can handle
Q.91 Which of the following best describes a 'discrete-time Fourier transform (DTFT)'?
A transformation of continuous signals to discrete time
An integral transform for discrete signals over a continuous frequency variable
A finite sum of complex exponentials for discrete signals
An algorithm to compute the FFT
Explanation - DTFT maps discrete-time signals to continuous frequency domain.
Correct answer is: An integral transform for discrete signals over a continuous frequency variable
Q.92 What is the main purpose of a 'bandpass filter' in communications?
To isolate a desired frequency band for transmission or reception
To remove all frequency components
To amplify low frequencies
To convert analog to digital
Explanation - Bandpass filters allow a specific band while rejecting others.
Correct answer is: To isolate a desired frequency band for transmission or reception
Q.93 Which of these is an example of a 'nonlinear' system?
An integrator
A limiter that clips the input
An ideal low-pass filter
A differentiator
Explanation - A limiter applies a nonlinear operation (clipping).
Correct answer is: A limiter that clips the input
Q.94 Which of these statements about 'spectral leakage' is true?
It occurs when the signal is perfectly periodic within the observation window
It is reduced by using window functions
It only occurs in analog signals
It increases the resolution of the FFT
Explanation - Windows smooth the edges of a finite sample, reducing leakage.
Correct answer is: It is reduced by using window functions
Q.95 The 'phase difference' between two sinusoids of the same frequency can be described as:
A time delay
A frequency shift
An amplitude scaling
A noise component
Explanation - Phase difference corresponds to a time shift between signals.
Correct answer is: A time delay
Q.96 The 'mean square value' of a sinusoid with amplitude A is:
A^2
A^2/2
A/2
A^2/4
Explanation - Average power of a sine wave is (A^2)/2.
Correct answer is: A^2/2
Q.97 Which of the following is a property of the Fourier transform of an even function?
It is purely imaginary
It is purely real
It is complex with no symmetry
It is zero
Explanation - Even time-domain signals produce real-valued Fourier transforms.
Correct answer is: It is purely real
Q.98 In a digital communication system, the 'bit error rate (BER)' is defined as:
The ratio of error bits to total transmitted bits
The number of bits per second
The modulation index
The power per bit
Explanation - BER measures the likelihood of a bit being received incorrectly.
Correct answer is: The ratio of error bits to total transmitted bits
Q.99 Which of the following describes the 'aliasing' effect in a frequency domain plot?
All frequency components shift up by the sampling frequency
Spectral components above Fs/2 wrap back into the baseband
Frequency components are attenuated by half
The spectrum becomes completely flat
Explanation - Aliasing maps high-frequency components into lower frequencies.
Correct answer is: Spectral components above Fs/2 wrap back into the baseband
Q.100 Which of the following is NOT a typical application of the Fast Fourier Transform (FFT)?
Audio analysis
Image processing
Sorting large data sets
Communication system design
Explanation - FFT is for frequency analysis, not sorting algorithms.
Correct answer is: Sorting large data sets
Q.101 The 'Nyquist rate' is related to the 'bandwidth' of the signal as:
Nyquist rate = Bandwidth
Nyquist rate = 2 * Bandwidth
Nyquist rate = Bandwidth / 2
Nyquist rate = 4 * Bandwidth
Explanation - The Nyquist rate is twice the highest frequency present.
Correct answer is: Nyquist rate = 2 * Bandwidth
Q.102 In digital filtering, what does the term 'group delay' refer to?
The time delay of the filter's output relative to its input
The derivative of the filter's phase response with respect to frequency
The filter's impulse response duration
The filter's attenuation at the cutoff frequency
Explanation - Group delay measures how phase changes with frequency, indicating distortion.
Correct answer is: The derivative of the filter's phase response with respect to frequency
Q.103 Which of the following best describes a 'zero-phase' filter?
It introduces a constant time delay
It has symmetric impulse response leading to no phase shift
It attenuates high frequencies only
It amplifies low frequencies only
Explanation - Zero-phase filters have symmetrical impulse responses, resulting in linear phase.
Correct answer is: It has symmetric impulse response leading to no phase shift
Q.104 A 'digital filter' implemented in a microcontroller typically requires:
Infinite precision arithmetic
Finite word-length arithmetic and coefficient quantization
Analog components such as resistors
No memory allocation
Explanation - DSP on microcontrollers uses finite-precision fixed or floating-point math.
Correct answer is: Finite word-length arithmetic and coefficient quantization
Q.105 In the context of signal processing, 'decimation' refers to:
Increasing the sampling rate
Decreasing the sampling rate
Increasing the resolution of the FFT
Filtering high frequencies only
Explanation - Decimation reduces the number of samples by discarding some.
Correct answer is: Decreasing the sampling rate
Q.106 Which of the following best explains the 'Nyquist–Shannon sampling theorem'?
Sampling below twice the bandwidth causes distortion
Sampling above the Nyquist rate ensures perfect reconstruction
Sampling at any rate preserves all information
Sampling rate has no effect on signal reconstruction
Explanation - The theorem guarantees lossless reconstruction if sampled above twice the max frequency.
Correct answer is: Sampling above the Nyquist rate ensures perfect reconstruction
Q.107 Which of the following is NOT a method of digital filtering?
FIR (Finite Impulse Response)
IIR (Infinite Impulse Response)
Analog RC filter
Recursive filter
Explanation - Analog RC filters are analog; the other options are digital techniques.
Correct answer is: Analog RC filter
Q.108 The term 'modulation' in signal processing refers to:
Adding a carrier to a signal
Removing noise from a signal
Changing the amplitude or frequency of a carrier wave
Sampling a continuous signal
Explanation - Modulation encodes information onto a carrier by varying its properties.
Correct answer is: Changing the amplitude or frequency of a carrier wave
Q.109 Which of these represents a 'white noise' spectrum?
Flat across all frequencies
Peaked at low frequencies
Peaked at high frequencies
Zero everywhere except at DC
Explanation - White noise has equal power at all frequencies.
Correct answer is: Flat across all frequencies
Q.110 What is the 'bandwidth' of a digital communication channel with a flat response from 0 to 5000 Hz?
5000 Hz
2500 Hz
10000 Hz
0 Hz
Explanation - Bandwidth equals the difference between highest and lowest frequencies.
Correct answer is: 5000 Hz
Q.111 Which of the following is a property of the Laplace Transform of an LTI system?
It yields a transfer function in the s-domain
It requires time-domain convolution
It is only valid for discrete signals
It cannot be inverted back to time domain
Explanation - The Laplace Transform produces the system's transfer function H(s).
Correct answer is: It yields a transfer function in the s-domain
Q.112 Which of the following best describes a 'quantization error' in ADCs?
Error due to the finite number of bits in the digital representation
Error due to aliasing
Error due to sampling rate
Error due to input amplitude
Explanation - Quantization error arises when mapping continuous values to discrete levels.
Correct answer is: Error due to the finite number of bits in the digital representation
Q.113 In a 'digital audio equalizer', which type of filter would you use to boost frequencies above 8 kHz?
High-pass filter
Low-pass filter
Band-stop filter
Band-pass filter
Explanation - A high-pass filter passes frequencies above its cutoff frequency.
Correct answer is: High-pass filter
Q.114 Which of the following best describes an 'ideal low-pass filter' impulse response?
Finite duration and zero outside the window
Infinite duration and decaying amplitude
Zero for all time
Infinite amplitude at zero time only
Explanation - The ideal low-pass filter has a sinc impulse response extending infinitely.
Correct answer is: Infinite duration and decaying amplitude
Q.115 In an analog signal chain, the 'pre-amplifier' typically:
Reduces the input signal amplitude
Amplifies the input signal to match ADC input range
Converts digital to analog
Filters high-frequency components
Explanation - Pre-amp boosts the signal level for optimal ADC input.
Correct answer is: Amplifies the input signal to match ADC input range
Q.116 The term 'band-limited' means a signal:
Has finite time duration
Contains only discrete frequencies
Contains no frequencies beyond a specified cutoff
Is always sinusoidal
Explanation - Band-limited signals have a maximum frequency beyond which no components exist.
Correct answer is: Contains no frequencies beyond a specified cutoff
Q.117 Which of the following is true about 'phase-locked loops (PLLs)'?
They are only used in audio processing
They can generate a stable oscillator frequency from a reference signal
They increase sampling rate
They have no relation to synchronization
Explanation - PLLs lock to a reference signal's phase and frequency.
Correct answer is: They can generate a stable oscillator frequency from a reference signal
Q.118 Which of these best describes the 'rectangular window' in time-domain analysis?
It smooths the edges of a signal window
It truncates the signal to a finite length
It multiplies the signal by a sine wave
It increases the spectral resolution
Explanation - A rectangular window selects a finite portion of the signal, causing spectral leakage.
Correct answer is: It truncates the signal to a finite length
Q.119 In signal processing, what is the 'frequency response' of a system?
The input signal amplitude over time
The output signal amplitude at different frequencies
The phase shift at different frequencies
The impulse response in time domain
Explanation - Frequency response shows how the system modifies signal amplitude across frequencies.
Correct answer is: The output signal amplitude at different frequencies
Q.120 Which of the following is an example of a 'recursive' digital filter structure?
FIR filter implemented by convolution
IIR filter using feedback from previous outputs
A moving average filter
A finite impulse response filter with no feedback
Explanation - Recursive (IIR) filters have feedback loops involving past outputs.
Correct answer is: IIR filter using feedback from previous outputs
Q.121 What is 'aliasing' most effectively mitigated by?
Increasing the sample rate
Using a high-pass filter before sampling
Using an anti-aliasing filter before sampling
Applying a low-pass filter after sampling
Explanation - Anti-aliasing filters suppress high-frequency content before sampling.
Correct answer is: Using an anti-aliasing filter before sampling
Q.122 Which of the following best describes a 'zero-crossing detector'?
Detects when a signal changes sign
Computes the Fourier transform
Filters out high frequencies
Measures signal amplitude
Explanation - Zero-crossing detectors find instances where the signal crosses the zero axis.
Correct answer is: Detects when a signal changes sign
Q.123 In the context of signal processing, what is meant by 'phase shift'?
A delay in the signal's time domain
An amplitude change in the frequency domain
A change in the signal's frequency
A change in the signal's polarity only
Explanation - Phase shift corresponds to a time delay relative to the input.
Correct answer is: A delay in the signal's time domain
Q.124 The 'sampling interval' is the reciprocal of:
Sampling rate
Signal bandwidth
Signal amplitude
Signal power
Explanation - Sampling interval Ts = 1/Fs.
Correct answer is: Sampling rate
Q.125 Which of the following best describes an 'analogue signal'?
Defined only at discrete time instants
Can take any real value at any time
Can only be binary
Cannot be converted to digital
Explanation - Analog signals vary continuously over time.
Correct answer is: Can take any real value at any time
Q.126 Which of the following is the correct interpretation of the term 'gain' in an amplifier?
The amount of input signal attenuated
The ratio of output to input amplitude
The input impedance
The frequency response of the amplifier
Explanation - Gain quantifies how much the amplifier increases the signal amplitude.
Correct answer is: The ratio of output to input amplitude
Q.127 In digital signal processing, which of these is a method to reduce aliasing artifacts?
Use a low-order filter
Use a high-order filter
Increase the sampling rate
Reduce the sampling rate
Explanation - A higher sampling rate reduces aliasing by keeping signals above Nyquist.
Correct answer is: Increase the sampling rate
Q.128 Which of the following best defines the 'impulse response' of a linear time-invariant (LTI) system?
The output when the input is a step function
The output when the input is a unit impulse
The frequency response of the system
The power spectrum of the output
Explanation - Impulse response characterizes the system’s behavior to a delta input.
Correct answer is: The output when the input is a unit impulse
Q.129 Which of the following best describes a 'bandstop filter'?
A filter that passes all frequencies
A filter that blocks a specific band of frequencies
A filter that only passes low frequencies
A filter that only passes high frequencies
Explanation - Bandstop filters attenuate frequencies within a chosen band.
Correct answer is: A filter that blocks a specific band of frequencies
Q.130 The 'Fourier transform' of a sine wave yields:
Two delta functions at ±frequency
A constant spectrum
A single delta at zero frequency
A rectangular function
Explanation - A sine wave corresponds to two spectral lines at positive and negative frequencies.
Correct answer is: Two delta functions at ±frequency
Q.131 In a digital communication system, what does the symbol 'SNR' stand for?
Signal to Noise Ratio
Signal to Number Ratio
Sample to Noise Ratio
Signal to Normal Ratio
Explanation - SNR measures signal power relative to noise power.
Correct answer is: Signal to Noise Ratio
Q.132 Which of the following best describes a 'unit step function' in discrete time, denoted by u[n]?
1 for all n < 0 and 0 for n ≥ 0
0 for all n < 0 and 1 for n ≥ 0
A sinusoid
A random noise sequence
Explanation - The unit step jumps from 0 to 1 at n=0.
Correct answer is: 0 for all n < 0 and 1 for n ≥ 0
Q.133 In the context of signal processing, what does the term 'time-frequency representation' refer to?
A plot of amplitude vs. time
A plot of amplitude vs. frequency only
A representation that shows frequency content as it varies over time
A representation of phase only
Explanation - Time-frequency methods like STFT display how spectral content changes over time.
Correct answer is: A representation that shows frequency content as it varies over time
Q.134 The term 'aliasing' is most closely associated with which of the following?
Oversampling
Undersampling
Full sampling
Random sampling
Explanation - Aliasing occurs when the sampling rate is below Nyquist.
Correct answer is: Undersampling
Q.135 What is the primary role of an 'anti-aliasing filter' in an ADC system?
To increase the ADC's resolution
To suppress frequencies above half the sampling rate
To amplify the input signal
To convert the signal from analog to digital
Explanation - It prevents high-frequency components from folding into the baseband.
Correct answer is: To suppress frequencies above half the sampling rate
Q.136 Which of the following is a 'finite impulse response' (FIR) filter characteristic?
It has infinite duration impulse response
It always has linear phase if coefficients are symmetric
It uses feedback
It can be unstable
Explanation - Symmetric FIR coefficients produce linear phase.
Correct answer is: It always has linear phase if coefficients are symmetric
Q.137 What is the purpose of 'zero-padding' a signal before computing its DFT?
To reduce aliasing
To improve frequency resolution
To shorten the computation time
To reduce amplitude of the signal
Explanation - Zero-padding interpolates the spectrum, providing finer frequency sampling.
Correct answer is: To improve frequency resolution
Q.138 Which of the following best describes the 'spectrum analyzer'?
A device that displays the magnitude of a signal as a function of frequency
A device that measures signal amplitude over time
A device that converts analog to digital
A device that amplifies signals
Explanation - Spectrum analyzers provide frequency domain visualization of signals.
Correct answer is: A device that displays the magnitude of a signal as a function of frequency
Q.139 In digital signal processing, the 'Z-transform' variable z is related to:
Time domain
Frequency domain
Both time and frequency domains
Phase domain
Explanation - The Z-transform generalizes the DTFT and includes a complex variable z.
Correct answer is: Both time and frequency domains
Q.140 Which of the following is a common application of the Fast Fourier Transform (FFT) in engineering?
Audio signal processing
Image compression
Speech synthesis
All of the above
Explanation - FFT is widely used across audio, image, and speech processing.
Correct answer is: All of the above
Q.141 In the context of digital filtering, what does 'IIR' stand for?
Infinite Impulse Response
Integer Input Register
Instantaneous Integer Response
Integrated Input Range
Explanation - IIR filters have feedback, leading to theoretically infinite impulse response.
Correct answer is: Infinite Impulse Response
Q.142 Which of the following is a common property of the 'Hilbert transform'?
It shifts the phase of all frequency components by 90 degrees
It amplifies all frequency components by 1
It removes the DC component
It reduces the signal’s bandwidth
Explanation - The Hilbert transform provides a 90° phase shift, creating the analytic signal.
Correct answer is: It shifts the phase of all frequency components by 90 degrees
Q.143 Which of the following is true about a 'band-limited' signal?
It has infinite bandwidth
It contains frequency components only within a finite range
It can be perfectly reconstructed at any sampling rate
It contains only a single frequency
Explanation - Band-limited signals have a maximum frequency beyond which no energy exists.
Correct answer is: It contains frequency components only within a finite range
Q.144 What does the 'frequency response' of a filter tell you?
The impulse response in time domain
The phase shift only
The magnitude and phase change across frequencies
The noise figure
Explanation - Frequency response describes how a filter modifies signal components at each frequency.
Correct answer is: The magnitude and phase change across frequencies
Q.145 In signal processing, which of the following represents a 'unit sample' (δ[n]) in discrete time?
Zero for all n except n=0 where it is one
One for all n
A sinusoidal sequence
A random sequence
Explanation - The discrete delta is 1 at n=0 and 0 elsewhere.
Correct answer is: Zero for all n except n=0 where it is one
Q.146 Which of the following best defines 'sampling rate'?
The number of samples per second
The amplitude of the sampled signal
The time interval between samples
The bandwidth of the signal
Explanation - Sampling rate is the frequency at which a continuous signal is sampled.
Correct answer is: The number of samples per second
Q.147 Which of these is a common method to reduce spectral leakage when computing the FFT?
Using a rectangular window
Using a Hamming window
Increasing the signal amplitude
Using a zero-phase filter
Explanation - Hamming window smooths the edges, reducing leakage.
Correct answer is: Using a Hamming window
Q.148 The 'convolution integral' is used to compute the output of a system when the input is:
A step function
A unit impulse
A sinusoid
An arbitrary signal
Explanation - Convolution integrates the product of input and impulse response over time.
Correct answer is: An arbitrary signal
Q.149 What is the main advantage of a 'linear phase' filter?
It provides maximum attenuation
It preserves the waveform shape of signals
It has the shortest impulse response
It requires no computation
Explanation - Linear phase ensures that all frequency components are delayed equally.
Correct answer is: It preserves the waveform shape of signals
Q.150 In a digital communication system, the term 'symbol rate' refers to:
The number of bits per second
The number of symbols transmitted per second
The frequency of the carrier
The amplitude of the signal
Explanation - Symbol rate (baud) counts how many modulation symbols are sent per second.
Correct answer is: The number of symbols transmitted per second
Q.151 Which of the following best describes the 'sinc' function?
A periodic function used in filter design
A function that decays exponentially
A function with infinite support and zero crossings
A function that is always positive
Explanation - sinc(x) = sin(πx)/(πx) has infinite extent and many zero crossings.
Correct answer is: A function with infinite support and zero crossings
Q.152 Which of the following is a typical application of the 'short-time Fourier transform (STFT)'?
Analyzing stationary signals only
Analyzing signals whose frequency content changes over time
Compressing audio data
Removing DC offset
Explanation - STFT provides a time-varying frequency representation.
Correct answer is: Analyzing signals whose frequency content changes over time
Q.153 In digital filtering, what does the term 'feedforward' refer to?
Using past outputs to compute the current output
Using past inputs to compute the current output
Using current inputs only
Using future inputs
Explanation - Feedforward (FIR) relies on current and past input samples.
Correct answer is: Using past inputs to compute the current output
Q.154 Which of the following is an example of a 'non-causal' filter?
A filter that depends only on past inputs
A filter that depends on future inputs
A filter that depends on current input only
A filter that depends on both past and future inputs equally
Explanation - Non-causal filters require knowledge of future samples, impossible in real-time.
Correct answer is: A filter that depends on future inputs
Q.155 The 'frequency response' of an ideal low-pass filter is characterized by:
A sharp transition at the cutoff frequency and zero beyond
A gradual roll-off
A linear phase shift
No attenuation at any frequency
Explanation - Ideal low-pass filters have a rectangular magnitude response.
Correct answer is: A sharp transition at the cutoff frequency and zero beyond
Q.156 Which of the following best describes a 'spectrogram'?
A plot of amplitude over time
A plot of frequency vs. time, showing spectral content
A plot of phase vs. frequency
A plot of amplitude vs. frequency only
Explanation - A spectrogram visualizes how the frequency spectrum evolves over time.
Correct answer is: A plot of frequency vs. time, showing spectral content
Q.157 In signal processing, what is a 'zero-order hold' (ZOH) used for?
To increase sample rate
To hold each sample value constant until the next sample
To convert analog to digital
To filter out high-frequency components
Explanation - ZOH maintains the sampled value for the duration of the sampling period.
Correct answer is: To hold each sample value constant until the next sample
Q.158 Which of the following is a characteristic of an 'IIR filter'?
Finite impulse response only
Uses only feedforward structure
Uses feedback from previous outputs
Always stable
Explanation - IIR filters include recursive feedback loops.
Correct answer is: Uses feedback from previous outputs
Q.159 Which of these best describes the term 'alias' in digital signal processing?
A frequency shift due to modulation
An artifact resulting from sampling below the Nyquist rate
A type of filter
A signal amplification technique
Explanation - Alias refers to spurious frequency components that appear when sampling is insufficient.
Correct answer is: An artifact resulting from sampling below the Nyquist rate
Q.160 The 'phase response' of a filter can affect:
Only the amplitude of the output
Only the timing of the output
Both amplitude and timing of the output
Neither amplitude nor timing
Explanation - Phase changes can distort wave shapes, affecting perceived signal timing.
Correct answer is: Both amplitude and timing of the output
Q.161 Which of the following is NOT a valid operation in digital signal processing?
Convolution
Filtering
Analog to digital conversion
Quantization
Explanation - Analog to digital is a hardware process; DSP works on digital data.
Correct answer is: Analog to digital conversion
Q.162 The 'power spectrum' of a signal is the magnitude of its:
Fourier transform squared
Laplace transform squared
Z-transform squared
Time domain representation
Explanation - Power spectrum = |X(ω)|², where X(ω) is the Fourier transform.
Correct answer is: Fourier transform squared
Q.163 In an analog filter, the term 'cutoff frequency' refers to:
Frequency at which the output is zero
Frequency where the output drops to -3 dB of the passband
Frequency at which the filter stops functioning
Any frequency above which the filter passes all signals
Explanation - Cutoff is defined at the -3 dB point of the filter’s magnitude response.
Correct answer is: Frequency where the output drops to -3 dB of the passband
Q.164 Which of the following statements about 'phase distortion' is correct?
It occurs when all frequency components experience the same delay
It causes the output waveform shape to change
It has no effect on the signal
It only occurs in analog systems
Explanation - Non-linear phase shifts distort the waveform shape, altering the signal.
Correct answer is: It causes the output waveform shape to change
Q.165 A 'unit delay' operation in discrete-time signal processing is represented by:
x[n-1]
x[n+1]
x[n]
x[-n]
Explanation - x[n-1] delays the sequence by one sample.
Correct answer is: x[n-1]
Q.166 Which of the following is a common application of the 'Fast Fourier Transform' in audio processing?
Spectral analysis
Noise reduction
Both A and B
All of the above
Explanation - FFT is used in analysis, equalization, and noise suppression.
Correct answer is: All of the above
Q.167 Which of the following best describes the 'spectrum' of a signal?
Its representation in the time domain
Its representation in the frequency domain
The filter’s impulse response
The signal’s phase only
Explanation - The spectrum shows how signal energy is distributed over frequency.
Correct answer is: Its representation in the frequency domain
Q.168 In a digital communication system, why is 'clock synchronization' important?
To avoid aliasing
To ensure accurate sampling and decoding of symbols
To increase bandwidth
To reduce power consumption
Explanation - Clock sync aligns the receiver’s sampling instant with transmitted symbols.
Correct answer is: To ensure accurate sampling and decoding of symbols
Q.169 Which of these is an example of a 'phase modulation' technique?
Amplitude shift keying
Phase shift keying
Frequency shift keying
Pulse code modulation
Explanation - PSK modulates information by shifting the phase of the carrier.
Correct answer is: Phase shift keying
Q.170 What is the main purpose of 'digital filtering' in a signal chain?
To convert digital signals back to analog
To shape the frequency content of a signal
To sample the signal
To quantize the signal
Explanation - Digital filters manipulate the signal’s spectral characteristics.
Correct answer is: To shape the frequency content of a signal
Q.171 Which of the following best defines the term 'alias frequency'?
The frequency after sampling that appears below the Nyquist frequency
The original frequency of a signal
The frequency at which a filter cuts off
The frequency of the sampling clock
Explanation - Aliasing maps higher frequencies into lower frequency ranges during sampling.
Correct answer is: The frequency after sampling that appears below the Nyquist frequency