Question 1 of 60Signal and System
The sifting property of the unit impulse function states that:
A∫x(t)δ(t−t₀)dt = x(t₀)
B∫x(t)dt = δ(t)
Cδ(t) = x(t) always
D∫δ(t)dt = 0
Answer is hidden
Question 2 of 60Signal and System
The sinc function is the Fourier transform pair of:
AA unit impulse
BAn ideal rectangular (low pass) spectrum
CA pure sinusoid only
DWhite noise
Answer is hidden
Question 3 of 60Signal and System
The impulse response of an LTI system is defined as its output when the input is:
AA unit step
BA unit impulse
CA sinusoid of infinite frequency
DZero
Answer is hidden
Question 4 of 60Signal and System
The output of a continuous-time LTI system for any input is found via:
AMultiplication of input and impulse response
BConvolution of input with the impulse response
CAddition of input and impulse response
DDifferentiation of the input
Answer is hidden
Question 5 of 60Signal and System
The Discrete-Time Fourier Series (DTFS), unlike the CTFS, represents a periodic signal using:
AAn infinite number of harmonics always
BA finite sum of N harmonics
CNo harmonics at all
DOnly the fundamental frequency
Answer is hidden
Question 6 of 60Signal and System
The Discrete-Time Fourier Transform (DTFT) of a sequence is always:
AAperiodic in frequency
BPeriodic in frequency with period 2π
CDefined only for finite sequences
DIdentical to the DTFS
Answer is hidden
Question 7 of 60Signal and System
Energy Spectral Density (ESD) of a signal is given by:
A|X(ω)|
B|X(ω)|²
CX(ω)³
D1/X(ω)
Answer is hidden
Question 8 of 60Signal and System
Power Spectral Density (PSD) applies most naturally to:
AFinite-energy signals only
BPower signals (with finite average power, possibly infinite energy)
COnly deterministic impulse signals
DOnly DC signals
Answer is hidden
Question 9 of 60Signal and System
The Hilbert transform of a signal changes its:
AMagnitude spectrum only
BPhase by 90°, without changing the magnitude spectrum
CAmplitude to zero
DFrequency content entirely
Answer is hidden
Question 10 of 60Signal and System
The signum function sgn(t) is defined as:
AAlways zero
B+1 for t>0, −1 for t<0, 0 at t=0
COnly positive values
DEqual to u(t) exactly
Answer is hidden
Question 11 of 60Linear Time Invariant System
The sampling theorem states that perfect reconstruction requires the sampling rate fₛ to satisfy:
Afₛ < fₘₐₓ
Bfₛ ≥ 2fₘₐₓ (the Nyquist rate)
Cfₛ = fₘₐₓ exactly
Dfₛ = 0
Answer is hidden
Question 12 of 60Linear Time Invariant System
The linearity property of an LTI system means it obeys:
AOnly time-invariance
BSuperposition
COnly causality
DOnly BIBO stability
Answer is hidden
Question 13 of 60Linear Time Invariant System
A causal system's output at any given time depends on:
AFuture inputs only
BPresent and past inputs only
CNothing at all
DOnly the system's memory-less property
Answer is hidden
Question 14 of 60Linear Time Invariant System
The frequency response H(ω) of an LTI system is obtained by:
ATaking the derivative of the impulse response
BTaking the Fourier transform of the impulse response
CSquaring the input signal
DSampling the output signal
Answer is hidden
Question 15 of 60Linear Time Invariant System
The impulse response of an ideal low pass filter is a:
ARectangular pulse
BSinc function
CUnit step
DDelta function
Answer is hidden
Question 16 of 60Linear Time Invariant System
The step response of an ideal low pass filter exhibits:
APerfect, distortion-free transition
BOvershoot/ringing, known as the Gibbs phenomenon
CZero output at all times
DInfinite bandwidth with no distortion
Answer is hidden
Question 17 of 60Linear Time Invariant System
BIBO stability of an LTI system requires that its impulse response be:
AInfinite in duration always
BAbsolutely summable/integrable
CExactly zero
DA pure sinusoid
Answer is hidden
Question 18 of 60Linear Time Invariant System
For a causal discrete-time system described by a transfer function, BIBO stability requires all poles to lie:
AOutside the unit circle
BStrictly inside the unit circle in the z-plane
CExactly on the unit circle
DAt the origin only
Answer is hidden
Question 19 of 60Linear Time Invariant System
Implementation of a discrete-time LTI system is typically based on:
AContinuous integration only
BA difference equation, realized with delay elements, multipliers, and adders
CRandom number generation
DOnly analog components
Answer is hidden
Question 20 of 60Linear Time Invariant System
A memoryless system's output at any time depends on:
APast and future inputs
BOnly the present input value
CNothing at all
DOnly the system's impulse response history
Answer is hidden
Question 21 of 60Z-Transform & Discrete Fourier Transform
The z-transform of a sequence x[n] is defined as:
AX(z) = Σx[n]zⁿ
BX(z) = Σx[n]z⁻ⁿ
CX(z) = ∫x(t)e⁻Ṙωᳳdt
DX(z) = x[n]²
Answer is hidden
Question 22 of 60Z-Transform & Discrete Fourier Transform
The DTFT of a sequence can be obtained from its z-transform by:
ASetting z=0
BEvaluating X(z) on the unit circle (z=e^(jω))
CTaking the derivative of X(z)
DSetting z to infinity
Answer is hidden
Question 23 of 60Z-Transform & Discrete Fourier Transform
The Region of Convergence (ROC) of a causal sequence is typically:
AThe interior of a circle
BThe exterior of a circle (|z|>r)
CThe entire z-plane always
DA single point only
Answer is hidden
Question 24 of 60Z-Transform & Discrete Fourier Transform
Convolution in the time domain corresponds, in the z-domain, to:
AAddition of z-transforms
BMultiplication of z-transforms
CDivision of z-transforms
DDifferentiation of z-transforms
Answer is hidden
Question 25 of 60Z-Transform & Discrete Fourier Transform
The system function H(z), evaluated on the unit circle, gives the system's:
ATransient response only
BSteady-state sinusoidal (frequency) response
CPole locations only
DTime-domain impulse response directly
Answer is hidden
Question 26 of 60Z-Transform & Discrete Fourier Transform
For a causal LTI system, BIBO stability requires all poles of H(z) to lie:
AOutside the unit circle
BStrictly inside the unit circle
CExactly on the unit circle
DAt z=1 only
Answer is hidden
Question 27 of 60Z-Transform & Discrete Fourier Transform
The Discrete Fourier Transform (DFT) can be understood as:
AA continuous transform with no relation to the DTFT
BThe DTFT of a finite-length sequence, sampled at N discrete frequency points
COnly applicable to analog signals
DIdentical to the z-transform in all respects
Answer is hidden
Question 28 of 60Z-Transform & Discrete Fourier Transform
Multiplying the DFTs of two sequences and taking the inverse DFT yields:
ATheir linear convolution
BTheir circular convolution
CTheir cross-correlation only
DZero, always
Answer is hidden
Question 29 of 60Z-Transform & Discrete Fourier Transform
Sampling the DTFT in the frequency domain corresponds, in the time domain, to:
ACompressing the original sequence
BPeriodic repetition of the original sequence
CDeleting the original sequence
DNo change at all
Answer is hidden
Question 30 of 60Z-Transform & Discrete Fourier Transform
The inverse z-transform can be computed using:
AOnly numerical integration in the time domain
BPartial fraction expansion, power series expansion, or the contour integral (residue) method
COnly the sampling theorem
DOnly the Hilbert transform
Answer is hidden
Question 31 of 60Implementation of Discrete-Time System
A direct-form realization of a digital filter implements the difference equation using:
AOnly analog components
BDelay elements, multipliers, and adders
COnly a single multiplier
DNo delay elements at all
Answer is hidden
Question 32 of 60Implementation of Discrete-Time System
A key advantage of the lattice structure over direct form is:
AHigher sensitivity to coefficient quantization
BBetter numerical robustness to coefficient quantization
CInability to realize IIR filters
DRequiring no reflection coefficients
Answer is hidden
Question 33 of 60Implementation of Discrete-Time System
A lattice-ladder structure is used specifically to realize:
AOnly FIR filters
BFull IIR filters, including both poles and zeros
COnly analog filters
DOnly the DFT
Answer is hidden
Question 34 of 60Implementation of Discrete-Time System
Fixed-point number representation is characterized by:
AA variable number of bits for integer/fractional parts
BA fixed number of bits allocated to integer and fractional parts, with limited dynamic range
CInfinite precision always
DRepresenting numbers as mantissa times an exponent
Answer is hidden
Question 35 of 60Implementation of Discrete-Time System
Floating-point representation offers which advantage over fixed-point?
ASimpler hardware
BWider dynamic range, at the cost of more complex hardware
CNo rounding error ever
DFixed precision at all magnitudes
Answer is hidden
Question 36 of 60Implementation of Discrete-Time System
Rounding error, compared to truncation error, is generally:
ABiased in one direction
BSymmetric, bounded by ±half the quantization step
CAlways larger
DImpossible to bound
Answer is hidden
Question 37 of 60Implementation of Discrete-Time System
Truncation, as a finite-precision effect, produces an error that is:
ASymmetric around zero
BBiased (one-directional)
CAlways exactly zero
DLarger than rounding error in all cases
Answer is hidden
Question 38 of 60Implementation of Discrete-Time System
Quantizing filter coefficients to finite precision primarily causes:
ANo change to the filter's poles/zeros
BPerturbation of the actual pole/zero locations from their ideal design values
CAutomatic improvement in filter stability
DElimination of the need for a filter structure
Answer is hidden
Question 39 of 60Implementation of Discrete-Time System
Coefficient quantization effects are especially critical for filters whose poles are located:
AFar from the unit circle
BNear the unit circle (e.g. narrow-band, high-Q filters)
CAt the origin of the z-plane
DOutside the region of convergence only
Answer is hidden
Question 40 of 60Implementation of Discrete-Time System
The conversion between direct-form coefficients and lattice reflection coefficients is typically performed using:
AA single multiplication
BA recursive algorithm (e.g. Levinson-Durbin-type recursion)
CRandom guessing
DThe sampling theorem
Answer is hidden
Question 41 of 60IIR & FIR Filter Design
Classical IIR filter design typically starts from:
AA random coefficient set
BA well-known analog prototype filter (e.g. Butterworth, Chebyshev)
CAn FIR filter converted directly
DA window function only
Answer is hidden
Question 42 of 60IIR & FIR Filter Design
The impulse-invariant method for IIR design obtains the digital filter by:
ASampling the analog prototype's frequency response
BSampling the analog prototype's impulse response
CIgnoring the analog prototype entirely
DUsing only FIR techniques
Answer is hidden
Question 43 of 60IIR & FIR Filter Design
FIR filter design by Fourier approximation computes coefficients as:
AThe derivative of the desired frequency response
BThe truncated inverse Fourier transform of the desired frequency response
CA random sequence
DThe z-transform of the input signal
Answer is hidden
Question 44 of 60IIR & FIR Filter Design
The window function method for FIR design is used primarily to:
AIncrease ripple near band edges
BTaper the truncated impulse response, reducing ripple at the cost of a wider transition band
CEliminate the need for any filter design method
DConvert an FIR filter into an IIR filter
Answer is hidden
Question 45 of 60IIR & FIR Filter Design
Which window is NOT commonly used in FIR filter design by the window method?
AHamming
BHanning
CBlackman
DGaussian elimination
Answer is hidden
Question 46 of 60IIR & FIR Filter Design
The frequency sampling method for FIR design computes coefficients by:
ATaking the derivative of a specified response
BApplying the inverse DFT to a desired frequency response specified at discrete points
CRandomly selecting values
DUsing only analog components
Answer is hidden
Question 47 of 60IIR & FIR Filter Design
The Remez exchange algorithm (Parks-McClellan) designs FIR filters by:
AMinimizing the maximum (Chebyshev/minimax) approximation error
BMaximizing the ripple in the passband
CIgnoring the stopband entirely
DUsing only a single window function
Answer is hidden
Question 48 of 60IIR & FIR Filter Design
An equi-ripple FIR filter design distributes approximation error:
AOnly in the passband, never the stopband
BEqually across ripples in both passband and stopband
CRandomly and unevenly
DOnly at a single frequency point
Answer is hidden
Question 49 of 60IIR & FIR Filter Design
Compared to window-based FIR design, equi-ripple (Remez) design generally achieves:
AA higher filter order for the same specifications
BA lower (more efficient) filter order for the same specifications
CIdentical results in all cases
DNo usable frequency response
Answer is hidden
Question 50 of 60IIR & FIR Filter Design
A key drawback of the impulse-invariant method for IIR design is:
AComplete elimination of aliasing
BPotential aliasing in the resulting digital filter's frequency response
CInability to design any IIR filter at all
DRequiring an FIR prototype
Answer is hidden
Question 51 of 60Digital Filter
A digital filter processes its input according to:
AA continuous analog equation only
BA difference equation (FIR or IIR)
CNo mathematical relationship at all
DOnly random noise addition
Answer is hidden
Question 52 of 60Digital Filter
The FFT algorithm reduces the computational complexity of computing a DFT from:
AO(N) to O(N²)
BO(N²) to O(Nlog₂N)
CO(Nlog₂N) to O(N³)
DThere is no complexity reduction
Answer is hidden
Question 53 of 60Digital Filter
Fast convolution using the FFT relies on the principle that:
AConvolution in time equals addition in frequency
BConvolution in time equals multiplication in frequency
CThe FFT cannot be used for convolution
DConvolution requires no transform at all
Answer is hidden
Question 54 of 60Digital Filter
The FFT is commonly used in spectral analysis to:
AIncrease the signal's amplitude
BEstimate a signal's frequency content
CEliminate the need for sampling
DConvert digital signals to analog
Answer is hidden
Question 55 of 60Digital Filter
Frequency-domain filtering using the FFT involves:
AMultiplying the signal's spectrum by the desired frequency response, then taking the inverse FFT
BAdding noise to the time-domain signal directly
CIgnoring the frequency response entirely
DUsing only analog filters
Answer is hidden
Question 56 of 60Digital Filter
OFDM (Orthogonal Frequency Division Multiplexing), used in modern communications, relies heavily on:
AAnalog filtering only
BFFT/IFFT processing
CDirect-form IIR filters exclusively
DManual Fourier series calculation
Answer is hidden
Question 57 of 60Digital Filter
Digital filters can be implemented in which of the following ways?
AHardware (DSP/FPGA/ASIC) or software on a general-purpose processor
BOnly as purely mechanical devices
COnly using analog components
DThey cannot be implemented practically
Answer is hidden
Question 58 of 60Digital Filter
Cascade (second-order sections) implementation of a digital filter is chosen mainly to improve:
AComputational speed only, with no other benefit
BNumerical robustness/reduced sensitivity to coefficient quantization
CThe filter's physical size only
DThe need for an FFT
Answer is hidden
Question 59 of 60Digital Filter
The FFT enables practical frequency-domain processing mainly because it:
AIncreases computational complexity for large datasets
BMakes frequency-domain analysis of large datasets computationally feasible
CRemoves the need for any digital signal processing
DWorks only for very small datasets
Answer is hidden
Question 60 of 60Digital Filter
2-D FFT is an application area relevant to:
AAudio-only processing
BImage processing
COnly analog telephony
DMechanical engineering exclusively
Answer is hidden