Q.1 What does a low‑pass filter do to a signal?
Keeps high‑frequency components
Keeps low‑frequency components
Reverses the signal
Adds noise
Explanation - A low‑pass filter allows frequencies below a specified cutoff to pass through while attenuating higher frequencies.
Correct answer is: Keeps low‑frequency components
Q.2 Which component is commonly used to construct a digital filter?
Resistor
Capacitor
Amplifier
Multiplier
Explanation - Digital filters perform operations like multiplication and addition on discrete samples; a multiplier is essential for implementing many digital filter algorithms.
Correct answer is: Multiplier
Q.3 In a Butterworth filter, the magnitude response is:
Sharp and steep
Flat in the passband
Random
Constant at all frequencies
Explanation - The Butterworth filter is designed for a maximally flat frequency response in the passband, giving a smooth transition to the stopband.
Correct answer is: Flat in the passband
Q.4 Which of the following is a property of an ideal filter?
Infinite time delay
No transition band
Non‑causal
All of the above
Explanation - An ideal filter has a perfectly sharp cutoff (no transition band), infinite group delay, and is non‑causal because its impulse response is non‑zero for negative time.
Correct answer is: All of the above
Q.5 What does the Q factor of a resonant filter represent?
Filter bandwidth
Quality of resonance
Sampling rate
Signal amplitude
Explanation - The Q factor indicates how under‑damped a resonant circuit is; high Q means a narrow bandwidth and a sharp peak in the frequency response.
Correct answer is: Quality of resonance
Q.6 A FIR filter has which type of impulse response?
Infinite
Finite
Zero
Non‑linear
Explanation - FIR stands for Finite‑Impulse‑Response, meaning the output depends only on a finite number of input samples.
Correct answer is: Finite
Q.7 What is the main advantage of a IIR filter over an FIR filter?
Higher computational cost
Fewer coefficients for a given performance
Non‑causal operation
Infinite impulse response
Explanation - IIR filters can achieve sharp frequency responses with fewer coefficients, making them computationally efficient compared to FIR filters.
Correct answer is: Fewer coefficients for a given performance
Q.8 Which technique is used to design a filter that approximates a desired frequency response?
Least‑squares method
Random search
Brute force
Hand‑tuning
Explanation - The least‑squares method optimizes the filter coefficients to minimize the squared error between the desired and actual responses.
Correct answer is: Least‑squares method
Q.9 In digital filter design, the Nyquist frequency is:
Half the sampling rate
Full sampling rate
Zero
The highest passband frequency
Explanation - The Nyquist frequency is defined as half of the sampling rate and represents the highest frequency that can be uniquely represented without aliasing.
Correct answer is: Half the sampling rate
Q.10 Which window function reduces the sidelobe level in a FIR filter design?
Rectangular
Hamming
Random
Triangular
Explanation - The Hamming window attenuates sidelobes better than a rectangular window, improving the filter's spectral leakage properties.
Correct answer is: Hamming
Q.11 What is the effect of increasing the filter order in a digital filter?
Reduces computational load
Worsens the passband ripple
Sharpens the transition band
No effect
Explanation - A higher filter order increases the number of coefficients, allowing a steeper roll‑off between passband and stopband.
Correct answer is: Sharpens the transition band
Q.12 Which of the following is NOT a characteristic of an IIR filter?
Infinite impulse response
Possibly non‑causal
Requires less memory
Always stable
Explanation - IIR filters can be unstable if poles lie outside the unit circle; stability must be carefully ensured during design.
Correct answer is: Always stable
Q.13 Which method can be used to realize a digital filter using only integer arithmetic?
Floating‑point arithmetic
Fixed‑point arithmetic
Analog simulation
None of the above
Explanation - Fixed‑point arithmetic uses integers and a scaling factor, making it suitable for embedded DSP systems with limited hardware resources.
Correct answer is: Fixed‑point arithmetic
Q.14 What does the group delay of a filter measure?
Amplitude distortion
Phase shift per frequency
Time delay across frequencies
Filter stability
Explanation - Group delay is the derivative of the filter’s phase response with respect to frequency; it indicates how different frequencies are delayed.
Correct answer is: Time delay across frequencies
Q.15 Which filter design technique uses a bilinear transform?
Chebyshev I
Elliptic
Kaiser
Bilinear
Explanation - The bilinear transform maps the s‑plane poles of an analog prototype to the z‑plane of a digital filter, preserving stability.
Correct answer is: Bilinear
Q.16 What is the purpose of a notch filter?
To eliminate a narrow band of frequencies
To amplify all frequencies
To smooth the signal
To compress the dynamic range
Explanation - Notch filters remove or attenuate a very specific frequency band, often used to suppress power line interference.
Correct answer is: To eliminate a narrow band of frequencies
Q.17 Which statement about zero‑phase filters is true?
They introduce phase distortion
They require forward and reverse filtering
They can only be analog
They have infinite group delay
Explanation - Zero‑phase filters achieve zero phase shift by filtering the signal in both forward and reverse directions, canceling phase effects.
Correct answer is: They require forward and reverse filtering
Q.18 In the context of filter design, what does the term 'passband ripple' refer to?
Amplitude variation within the passband
Delay variation across frequencies
Noise added by the filter
The slope of the transition band
Explanation - Passband ripple is the amount of deviation from a perfectly flat response within the frequencies that are allowed to pass.
Correct answer is: Amplitude variation within the passband
Q.19 Which of these is a commonly used digital filter design tool?
MATLAB’s fdatool
MS Word
Excel
PowerPoint
Explanation - MATLAB’s fdatool provides a graphical interface for designing and analyzing digital filters.
Correct answer is: MATLAB’s fdatool
Q.20 A Hilbert transform filter is primarily used to produce:
A real‑valued signal
A 90° phase‑shifted version of a signal
A high‑pass filtered signal
An amplitude‑modulated signal
Explanation - The Hilbert transform shifts the phase of each frequency component by 90°, creating an analytic signal.
Correct answer is: A 90° phase‑shifted version of a signal
Q.21 What is the main drawback of using a high‑order FIR filter?
Increased computational complexity
Reduced stability
No frequency response
Limited passband
Explanation - High‑order FIR filters require many arithmetic operations per sample, leading to higher CPU usage and power consumption.
Correct answer is: Increased computational complexity
Q.22 In a digital filter implemented in a microcontroller, which instruction set is most critical for performance?
Floating‑point unit (FPU)
Graphics processing unit (GPU)
Tensor core
None of the above
Explanation - The FPU handles the arithmetic operations of the filter efficiently; without it, performance drops significantly.
Correct answer is: Floating‑point unit (FPU)
Q.23 What does the term 'aliasing' refer to in digital signal processing?
Signal distortion due to non‑linearity
Frequency components folding back into the baseband
Hardware noise
Signal amplification
Explanation - Aliasing occurs when a signal is undersampled, causing high‑frequency components to masquerade as lower frequencies.
Correct answer is: Frequency components folding back into the baseband
Q.24 Which type of filter is best suited for removing high‑frequency noise while preserving low‑frequency information?
High‑pass filter
Low‑pass filter
Band‑stop filter
Band‑pass filter
Explanation - A low‑pass filter allows low‑frequency signals to pass and attenuates high‑frequency noise.
Correct answer is: Low‑pass filter
Q.25 What is the purpose of the window function in FIR filter design?
To reduce the filter's memory usage
To control the trade‑off between main lobe width and side lobe levels
To increase the sampling rate
To convert analog signals to digital
Explanation - Different window functions affect the filter’s frequency response characteristics, balancing sharpness and ripple.
Correct answer is: To control the trade‑off between main lobe width and side lobe levels
Q.26 Which filter type has a symmetrical impulse response?
FIR filter
IIR filter
Notch filter
All filters are symmetrical
Explanation - FIR filters are designed with symmetric coefficients, resulting in linear phase and a symmetrical impulse response.
Correct answer is: FIR filter
Q.27 In filter design, what does 'k' typically represent?
Number of poles
Cutoff frequency in radians per second
Filter order
Bandwidth
Explanation - The variable 'k' is often used to denote the number of filter stages or the order of the transfer function.
Correct answer is: Filter order
Q.28 Which of the following is NOT a reason to use a digital filter over an analog filter?
Adjustable parameters
Unlimited memory
Higher precision
Hardware cost
Explanation - Digital filters are limited by the memory of the processor; analog filters don't have this constraint.
Correct answer is: Unlimited memory
Q.29 What does the 'stopband attenuation' specify in a filter design?
Maximum gain in the passband
Minimum attenuation in the stopband
Cutoff frequency
Group delay
Explanation - Stopband attenuation measures how much the filter reduces unwanted frequencies in the stopband.
Correct answer is: Minimum attenuation in the stopband
Q.30 Which type of digital filter is most sensitive to coefficient quantization errors?
FIR filter
IIR filter
Band‑pass filter
Low‑pass filter
Explanation - IIR filters have recursive coefficients that can amplify quantization errors, potentially leading to instability.
Correct answer is: IIR filter
Q.31 What is the role of a pole in an IIR filter's transfer function?
It determines the filter’s zeros
It controls the filter’s gain
It defines the filter’s resonant frequency
It has no effect
Explanation - Poles correspond to frequencies at which the filter's response peaks and can affect stability.
Correct answer is: It defines the filter’s resonant frequency
Q.32 Which of the following is a common application of a digital low‑pass filter?
Audio equalization
Image sharpening
Video compression
Noise reduction in sensor data
Explanation - Low‑pass filters are often used to smooth sensor signals by removing high‑frequency noise components.
Correct answer is: Noise reduction in sensor data
Q.33 Which window function is known for having the narrowest main lobe?
Rectangular
Hamming
Blackman
None of the above
Explanation - The rectangular window has the narrowest main lobe but suffers from high sidelobe levels, leading to significant leakage.
Correct answer is: Rectangular
Q.34 What is the effect of increasing the sampling rate on a digital filter?
Increases the filter’s passband ripple
Reduces the risk of aliasing
Decreases computational load
None of the above
Explanation - Higher sampling rates push the Nyquist frequency higher, allowing more bandwidth before aliasing occurs.
Correct answer is: Reduces the risk of aliasing
Q.35 Which type of filter is typically used to eliminate a 50/60 Hz power line interference?
Low‑pass filter
High‑pass filter
Notch filter
Band‑pass filter
Explanation - A notch filter is specifically designed to attenuate a narrow band of frequencies, such as power line noise.
Correct answer is: Notch filter
Q.36 Which filter design technique is known for its use of elliptic integrals?
Chebyshev I
Elliptic
Butterworth
Kaiser
Explanation - Elliptic filters use elliptic integrals to achieve the steepest roll‑off for a given order, at the cost of ripple in both passband and stopband.
Correct answer is: Elliptic
Q.37 In the context of DSP, what does 'DSP' stand for?
Digital Signal Processing
Dynamic Signal Plotting
Data Storage Protocol
Digital Storage Processor
Explanation - DSP refers to the use of digital methods to analyze, modify, or synthesize signals.
Correct answer is: Digital Signal Processing
Q.38 Which of the following is NOT a component of a digital filter’s transfer function?
Zeros
Poles
Gain
Frequency response
Explanation - Zeros, poles, and gain define the transfer function; the frequency response is derived from it.
Correct answer is: Frequency response
Q.39 What is the main advantage of using a window method for FIR filter design?
It provides a stable filter with zero phase distortion
It allows easy adjustment of passband and stopband widths
It eliminates the need for any computational resources
All of the above
Explanation - Window methods simplify FIR design by truncating the impulse response and controlling the trade‑offs via window choice.
Correct answer is: It allows easy adjustment of passband and stopband widths
Q.40 What does the 'knee' frequency refer to in a filter’s magnitude response?
The frequency where the slope of the response changes
The frequency at which the filter stops working
The maximum passband frequency
The frequency of zero output
Explanation - The knee frequency marks the point where the filter’s attenuation begins to increase rapidly.
Correct answer is: The frequency where the slope of the response changes
Q.41 Which filter type is typically used to preserve the phase of a signal while filtering?
FIR linear phase filter
IIR filter
Notch filter
High‑pass filter
Explanation - FIR filters designed with symmetrical coefficients can maintain a constant group delay, preserving the signal’s phase.
Correct answer is: FIR linear phase filter
Q.42 In designing a digital filter, which parameter defines the number of taps or coefficients?
Filter order
Cutoff frequency
Sampling rate
Zero‑pole count
Explanation - The filter order directly determines the number of coefficients required for the filter’s impulse response.
Correct answer is: Filter order
Q.43 What is the purpose of oversampling in filter design?
To reduce computational complexity
To shift the Nyquist frequency lower
To allow more relaxed filter specifications before decimation
To increase aliasing
Explanation - Oversampling raises the sampling rate, providing a wider transition band and easing the design of the initial filter stage.
Correct answer is: To allow more relaxed filter specifications before decimation
Q.44 Which of the following best describes a digital Butterworth filter?
Non‑causal with steep roll‑off
Linear phase with flat magnitude in passband
Has equal ripple in stopband
Requires infinite memory
Explanation - A digital Butterworth filter is designed to have a maximally flat magnitude response in the passband and a smooth transition to the stopband.
Correct answer is: Linear phase with flat magnitude in passband
Q.45 Which filter property is directly affected by the placement of poles in the z‑plane?
Passband ripple
Stability
Sampling rate
Quantization noise
Explanation - If poles lie outside the unit circle in the z‑plane, the filter becomes unstable.
Correct answer is: Stability
Q.46 What does the 'Hamming distance' measure in a filter context?
The difference between two filter coefficients sets
The number of taps in a filter
The transition band width
The filter’s delay
Explanation - Hamming distance counts the differing bits between two binary strings, useful in quantization and error detection.
Correct answer is: The difference between two filter coefficients sets
Q.47 Which of the following is a key advantage of using a fixed‑point DSP architecture for real‑time filter implementation?
Higher precision
Lower power consumption
Unlimited dynamic range
Automatic stability guarantees
Explanation - Fixed‑point DSPs consume less power and often have simpler hardware compared to floating‑point units.
Correct answer is: Lower power consumption
Q.48 What does the term 'alias' mean in the context of sampling theory?
A frequency component appearing at an incorrect frequency due to undersampling
A digital artifact introduced by quantization
A phase shift in the signal
A type of filter design
Explanation - Aliasing causes high‑frequency components to masquerade as lower frequencies when the sampling rate is too low.
Correct answer is: A frequency component appearing at an incorrect frequency due to undersampling
Q.49 Which of the following is a characteristic of an IIR filter designed using bilinear transform?
Infinite impulse response
Non‑causal
Frequency warping
No poles
Explanation - The bilinear transform maps the analog frequency axis to the digital frequency axis non‑linearly, causing frequency warping.
Correct answer is: Frequency warping
Q.50 In a digital filter, what is the role of the 'sinc' function in the ideal low‑pass filter?
It defines the impulse response in continuous time
It is used as a window function
It represents the frequency response of the filter
It provides the zero‑phase response
Explanation - The sinc function is the continuous‑time impulse response of an ideal low‑pass filter.
Correct answer is: It defines the impulse response in continuous time
Q.51 Which filter type is most suitable for real‑time audio equalization?
IIR filter
FIR filter
Notch filter
Band‑stop filter
Explanation - IIR filters provide efficient computation with relatively few coefficients, making them ideal for real‑time audio processing.
Correct answer is: IIR filter
Q.52 What is the effect of applying a Hamming window to a truncated sinc function?
Increases side lobe levels
Sharpens the main lobe
Reduces side lobe levels
No effect
Explanation - The Hamming window reduces side lobe amplitudes, improving the filter’s attenuation performance at the cost of a slightly wider main lobe.
Correct answer is: Reduces side lobe levels
Q.53 Which of the following is a property of a linear phase FIR filter?
Group delay is constant across frequencies
Impulse response is anti‑symmetric
Poles lie on the unit circle
It has infinite order
Explanation - A linear phase FIR filter has a constant group delay, preserving the waveform shape of the filtered signal.
Correct answer is: Group delay is constant across frequencies
Q.54 Which of the following best describes the term 'quantization error'?
The difference between analog and digital signals due to sampling
The error introduced by finite‑resolution representation of coefficients
The error in filter's frequency response due to rounding
Both B and C
Explanation - Quantization error arises from rounding the coefficients and intermediate values, affecting the filter’s performance.
Correct answer is: Both B and C
Q.55 In filter synthesis, what is the 'pole‑zero map' used for?
Visualizing the filter’s frequency response
Determining the filter’s stability
Designing the analog prototype
All of the above
Explanation - The pole‑zero map shows the locations of poles and zeros in the complex plane, indicating stability, frequency response, and guiding design.
Correct answer is: All of the above
Q.56 Which of the following is NOT a typical step in designing a digital low‑pass filter using the window method?
Choosing the desired cutoff frequency
Determining the filter order
Selecting the window function
Calculating the analog frequency response
Explanation - The window method directly designs a digital filter; an analog response is not required in this approach.
Correct answer is: Calculating the analog frequency response
Q.57 What is the primary cause of ripple in a Chebyshev filter?
Uniform coefficient values
Non‑linear phase response
Non‑uniform spacing of poles
Infinite impulse response
Explanation - Chebyshev filters intentionally place poles unevenly to achieve a ripple in the passband (Type I) or stopband (Type II).
Correct answer is: Non‑uniform spacing of poles
Q.58 Which type of filter is used to isolate a narrow band of frequencies around a center frequency?
Band‑stop filter
Band‑pass filter
High‑pass filter
Low‑pass filter
Explanation - A band‑pass filter allows frequencies within a specified range to pass while attenuating frequencies outside that range.
Correct answer is: Band‑pass filter
Q.59 In digital signal processing, what does the term 'aliasing' most directly refer to?
A type of filter
The loss of phase information
The superposition of higher frequencies onto lower ones due to insufficient sampling
A quantization technique
Explanation - Aliasing is the phenomenon where high‑frequency content folds back into lower frequencies when sampling below the Nyquist rate.
Correct answer is: The superposition of higher frequencies onto lower ones due to insufficient sampling
Q.60 Which of the following statements is true regarding zero‑phase filtering?
It introduces a linear phase shift
It can only be achieved with analog filters
It requires filtering in both forward and reverse directions
It is equivalent to a notch filter
Explanation - Zero‑phase filtering is accomplished by applying the filter forward and then backward, cancelling the phase shift.
Correct answer is: It requires filtering in both forward and reverse directions
Q.61 Which type of filter would you use to remove high‑frequency noise while preserving a low‑frequency signal component?
High‑pass filter
Band‑pass filter
Low‑pass filter
Notch filter
Explanation - A low‑pass filter allows low‑frequency signals to pass through while attenuating higher‑frequency noise components.
Correct answer is: Low‑pass filter
Q.62 What does the 'impulse response' of a filter represent?
The output of the filter to a sine wave input
The output of the filter to a unit impulse input
The frequency response of the filter
The transfer function
Explanation - The impulse response describes how the filter reacts to a single instantaneous input, fully characterizing its behavior.
Correct answer is: The output of the filter to a unit impulse input
Q.63 Which of the following is a major advantage of FIR filters?
Infinite memory requirement
Non‑causal operation
Linear phase capability
Higher computational cost
Explanation - FIR filters can be designed with symmetric coefficients to achieve a linear phase response, preserving waveform shape.
Correct answer is: Linear phase capability
Q.64 Which filter design approach involves transforming an analog prototype into a digital filter?
Impulse invariant method
Bilinear transform
Window method
Frequency sampling method
Explanation - The bilinear transform maps the s‑plane prototype to the z‑plane, creating a stable digital filter from an analog design.
Correct answer is: Bilinear transform
Q.65 In the design of a digital filter, why is the 'cutoff frequency' often specified in terms of the Nyquist frequency?
To simplify the transfer function
To ensure the filter remains causal
To express the cutoff as a fraction of the maximum representable frequency
Because the Nyquist frequency is irrelevant
Explanation - Specifying the cutoff as a fraction of the Nyquist frequency makes the design independent of the actual sampling rate.
Correct answer is: To express the cutoff as a fraction of the maximum representable frequency
Q.66 Which of the following is NOT an example of a real‑world application for digital filters?
Audio equalization
Image sharpening
Signal demodulation
Analog power supply regulation
Explanation - Digital filters process digital signals, while power supply regulation typically uses analog circuitry.
Correct answer is: Analog power supply regulation
Q.67 Which window function is known for its excellent sidelobe attenuation but a relatively wide main lobe?
Rectangular
Hamming
Blackman
Kaiser
Explanation - The Blackman window achieves very low sidelobe levels at the cost of a wider transition band.
Correct answer is: Blackman
Q.68 What is the purpose of a 'pole‑zero' representation in filter analysis?
To calculate the filter’s impulse response
To visualize the filter’s stability and frequency response
To determine the filter’s order
Both A and B
Explanation - The pole‑zero map shows the filter’s stability (poles) and frequency response characteristics (zeros).
Correct answer is: Both A and B
Q.69 Which of the following best describes the 'Butterworth filter'?
Has maximum passband ripple
Has a flat frequency response in the passband
Has a linear phase
Has a rectangular window
Explanation - The Butterworth filter is designed for a maximally flat magnitude response in its passband.
Correct answer is: Has a flat frequency response in the passband
Q.70 Which digital filter design method is especially useful when the desired filter shape is defined by a set of frequency points?
Window method
Frequency sampling method
Impulse invariant method
Least‑squares method
Explanation - The frequency sampling method designs a filter by directly specifying its frequency response at discrete frequencies.
Correct answer is: Frequency sampling method
Q.71 In the context of filter design, what does the term 'stopband' refer to?
The range of frequencies that the filter passes
The range of frequencies that the filter attenuates
The frequency at which the filter stops working
The frequency of the zero output
Explanation - The stopband is the portion of the spectrum where the filter’s output is significantly attenuated.
Correct answer is: The range of frequencies that the filter attenuates
Q.72 Which of the following is a characteristic of an IIR filter designed using the bilinear transform?
Infinite impulse response
No frequency warping
Linear phase
All of the above
Explanation - IIR filters have recursive coefficients leading to an infinite impulse response.
Correct answer is: Infinite impulse response
Q.73 Which of the following techniques can be used to reduce the effect of aliasing in a digital signal?
Increasing the sampling rate
Using a higher order filter
Applying a low‑pass anti‑aliasing filter before sampling
All of the above
Explanation - Increasing sampling rate, filter order, and anti‑aliasing filters all help mitigate aliasing effects.
Correct answer is: All of the above
Q.74 What is the significance of the 'Hamming window' in FIR filter design?
It maximizes the main‑lobe width
It provides a balance between main‑lobe width and side‑lobe levels
It eliminates all side‑lobes
It is used for IIR filters only
Explanation - The Hamming window offers a compromise between resolution (main‑lobe width) and attenuation (side‑lobe level).
Correct answer is: It provides a balance between main‑lobe width and side‑lobe levels
Q.75 In a digital filter, which of the following best describes the 'group delay'?
The difference between the input and output amplitudes
The time delay of the filter’s impulse response
The time delay per frequency component
The filter’s bandwidth
Explanation - Group delay indicates how different frequency components are delayed as they pass through the filter.
Correct answer is: The time delay per frequency component
Q.76 Which filter type is typically used to remove a narrow band of unwanted frequency, such as 60 Hz power line hum?
Band‑pass filter
Band‑stop filter
Notch filter
Low‑pass filter
Explanation - A notch filter is designed to attenuate a very narrow frequency band while leaving other frequencies largely unaffected.
Correct answer is: Notch filter
Q.77 Which of the following is a primary advantage of using a digital filter in an embedded system?
Unlimited memory usage
No quantization error
High flexibility with reconfiguration
Analog components are not needed
Explanation - Digital filters can be easily re‑programmed for different specifications, making them versatile for embedded applications.
Correct answer is: High flexibility with reconfiguration
Q.78 What is the main purpose of the 'sinc' function in ideal filter design?
To approximate the impulse response of an ideal low‑pass filter
To approximate the impulse response of an ideal high‑pass filter
To approximate the impulse response of an ideal band‑pass filter
None of the above
Explanation - The sinc function represents the impulse response of an ideal low‑pass filter in continuous time.
Correct answer is: To approximate the impulse response of an ideal low‑pass filter
Q.79 Which of the following is a drawback of the 'rectangular' window when used in FIR filter design?
High side‑lobe levels
Narrow main‑lobe width
Large passband ripple
Both A and B
Explanation - The rectangular window yields a narrow main lobe but suffers from high side‑lobe levels, causing significant leakage.
Correct answer is: Both A and B
Q.80 In a digital filter, what does the term 'pole' represent?
A frequency where the filter output is zero
A frequency where the filter output is infinite
A coefficient of the filter
The filter’s maximum amplitude
Explanation - Poles are points in the z‑plane where the denominator of the transfer function equals zero, potentially leading to infinite gain.
Correct answer is: A frequency where the filter output is infinite
Q.81 Which of the following is an example of a 'finite‑impulse‑response' (FIR) filter?
A filter with recursive coefficients
A filter whose impulse response terminates after a finite number of samples
A filter that uses infinite memory
A filter that has no zeros
Explanation - An FIR filter’s impulse response is zero after a certain number of samples, making its order finite.
Correct answer is: A filter whose impulse response terminates after a finite number of samples
Q.82 What does 'bandwidth' refer to in filter design?
The range of frequencies a filter allows to pass
The width of the filter’s transition band
The difference between cutoff frequencies in a band‑pass filter
All of the above
Explanation - Bandwidth can refer to the passband width, the transition band width, or the difference between upper and lower cutoff frequencies.
Correct answer is: All of the above
Q.83 Which of the following is a common method for reducing the effects of quantization in digital filters?
Increasing the filter order
Using floating‑point arithmetic
Applying a low‑pass filter before quantization
Both B and C
Explanation - Floating‑point arithmetic reduces quantization error, and pre‑filtering can reduce the dynamic range before quantization.
Correct answer is: Both B and C
Q.84 Which of the following statements best describes a 'linear‑phase' FIR filter?
It has a non‑linear frequency response
It maintains a constant phase shift across all frequencies
It has a constant group delay
Both B and C
Explanation - A linear‑phase filter ensures constant group delay, which preserves the waveform shape of the processed signal.
Correct answer is: Both B and C
Q.85 Which filter design approach is best for a real‑time audio system requiring minimal computational load?
High‑order FIR filter
Low‑order IIR filter
Band‑pass FIR filter
Band‑stop IIR filter
Explanation - Low‑order IIR filters achieve sharp transitions with fewer computations, making them ideal for real‑time audio processing.
Correct answer is: Low‑order IIR filter
Q.86 In the context of digital filter design, what does the term 'alias' most often refer to?
The frequency shift that occurs due to under‑sampling
A type of noise introduced by the filter
The difference between the analog and digital filter responses
The delay introduced by the filter
Explanation - Aliasing occurs when high‑frequency components are misrepresented as lower frequencies due to insufficient sampling.
Correct answer is: The frequency shift that occurs due to under‑sampling
Q.87 Which of the following is NOT a typical component of an analog low‑pass filter?
Resistor
Capacitor
Inductor
Microphone
Explanation - Resistors, capacitors, and inductors form the active elements of an analog filter; a microphone is an input device, not a filter component.
Correct answer is: Microphone
Q.88 Which filter type is best suited for removing a specific unwanted frequency from a signal?
Band‑pass filter
Band‑stop filter
Notch filter
High‑pass filter
Explanation - A notch filter attenuates a narrow band of frequencies while leaving the rest of the spectrum unchanged.
Correct answer is: Notch filter
Q.89 What does the term 'pole‑zero' map provide in filter analysis?
The filter’s frequency response
The filter’s stability characteristics
Both A and B
The filter’s memory usage
Explanation - A pole‑zero map visualizes how the filter’s poles and zeros influence its frequency response and stability.
Correct answer is: Both A and B
Q.90 Which of the following is a major advantage of using a fixed‑point DSP for embedded filter applications?
High precision floating‑point calculations
Lower power consumption
Unlimited memory usage
Automatic stability enforcement
Explanation - Fixed‑point DSPs consume less power and are often more cost‑effective for embedded real‑time processing.
Correct answer is: Lower power consumption
Q.91 Which filter design method uses a known analytic expression for the frequency response?
Window method
Frequency sampling method
Impulse invariant method
Least‑squares method
Explanation - The frequency sampling method constructs a filter by specifying its response at discrete frequency points based on analytic expressions.
Correct answer is: Frequency sampling method
Q.92 Which of the following best describes the 'cutoff frequency' in a filter?
The frequency where the filter’s gain is zero
The frequency where the filter’s gain drops to −3 dB
The frequency where the filter stops working
The frequency of the maximum output
Explanation - The −3 dB point is conventionally used to define the cutoff frequency of a filter.
Correct answer is: The frequency where the filter’s gain drops to −3 dB
Q.93 In a digital filter, the 'stopband attenuation' refers to:
The maximum attenuation in the passband
The minimum attenuation in the stopband
The bandwidth of the filter
The sampling rate
Explanation - Stopband attenuation measures how much the filter reduces unwanted frequencies in the stopband.
Correct answer is: The minimum attenuation in the stopband
Q.94 Which type of filter would you use to remove all frequencies above a certain threshold while keeping lower frequencies?
Low‑pass filter
High‑pass filter
Band‑pass filter
Band‑stop filter
Explanation - A low‑pass filter allows frequencies below the cutoff to pass and attenuates higher frequencies.
Correct answer is: Low‑pass filter
Q.95 What does the term 'group delay' represent in the context of digital filters?
The delay of the filter’s impulse response
The frequency response of the filter
The time delay for each frequency component
The filter’s stability margin
Explanation - Group delay is the derivative of the phase response with respect to frequency, indicating how each component is delayed.
Correct answer is: The time delay for each frequency component
Q.96 Which filter design technique is best suited for real‑time processing with minimal computational resources?
High‑order FIR filter
Low‑order IIR filter
Band‑pass FIR filter
Band‑stop IIR filter
Explanation - Low‑order IIR filters provide sharp frequency responses with fewer calculations, ideal for real‑time applications.
Correct answer is: Low‑order IIR filter
Q.97 Which of the following is a primary advantage of using a digital filter over an analog filter?
Unlimited memory
Fixed frequency response
Easy reconfiguration
No quantization error
Explanation - Digital filters can be reprogrammed for different specifications, allowing flexible adjustment of parameters.
Correct answer is: Easy reconfiguration
Q.98 Which of the following best describes the purpose of a notch filter?
To allow a narrow band of frequencies to pass
To attenuate a narrow band of frequencies
To eliminate all frequencies below a threshold
To amplify a narrow band of frequencies
Explanation - Notch filters are used to suppress a specific narrow frequency band, such as power‑line interference.
Correct answer is: To attenuate a narrow band of frequencies
Q.99 Which window function is known for its very low side‑lobe levels but a relatively broad main lobe?
Rectangular
Hamming
Blackman
Kaiser
Explanation - The Blackman window achieves excellent side‑lobe attenuation at the cost of a wider main lobe.
Correct answer is: Blackman
Q.100 Which of the following is a key benefit of an FIR filter having a linear phase response?
Zero group delay
Constant group delay across frequencies
No passband ripple
Infinite impulse response
Explanation - A linear phase FIR filter maintains a constant group delay, preserving the waveform shape of the signal.
Correct answer is: Constant group delay across frequencies
Q.101 Which of the following is NOT a typical design parameter for a digital filter?
Cutoff frequency
Filter order
Sampling rate
Color of the filter
Explanation - Filter design involves frequency parameters; the color of the filter is irrelevant.
Correct answer is: Color of the filter
Q.102 In the bilinear transform method, what does frequency warping refer to?
The change of frequency scale from analog to digital
The distortion of the filter’s phase
The stretching of the impulse response
The amplification of stopband frequencies
Explanation - Frequency warping occurs because the bilinear transform maps the analog frequency axis to the digital frequency axis in a non‑linear way.
Correct answer is: The change of frequency scale from analog to digital
Q.103 Which of the following best describes the impulse invariant method?
It converts a digital filter to an analog one
It preserves the analog impulse response at sampled times
It uses a window function to design a FIR filter
It is used for high‑order filter design
Explanation - The impulse invariant method samples the analog impulse response to produce a digital filter with a similar response at discrete times.
Correct answer is: It preserves the analog impulse response at sampled times
Q.104 Which type of filter is typically used to isolate a signal component around a center frequency?
Band‑stop filter
Band‑pass filter
High‑pass filter
Low‑pass filter
Explanation - A band‑pass filter passes frequencies around a specified center frequency while attenuating frequencies outside that band.
Correct answer is: Band‑pass filter
Q.105 Which of the following is a common application of a digital filter in audio processing?
Signal amplification
Noise reduction
Color correction
Image resizing
Explanation - Digital filters are widely used in audio to remove unwanted noise while preserving desired signal components.
Correct answer is: Noise reduction
Q.106 Which of the following statements best describes a Chebyshev Type I filter?
It has a flat passband and ripple in the stopband
It has ripple in the passband and a flat stopband
It has no ripple in either band
It is equivalent to a Butterworth filter
Explanation - Chebyshev Type I filters exhibit equiripple behavior in the passband while maintaining a flat stopband.
Correct answer is: It has ripple in the passband and a flat stopband
Q.107 What is the main purpose of a 'high‑pass filter' in signal processing?
To remove low‑frequency components from a signal
To remove high‑frequency components from a signal
To amplify low frequencies
To attenuate all frequencies
Explanation - A high‑pass filter allows high frequencies to pass while attenuating lower frequencies.
Correct answer is: To remove low‑frequency components from a signal
Q.108 Which filter design technique involves specifying a set of desired frequency points and computing the filter coefficients to fit those points?
Frequency sampling method
Window method
Impulse invariant method
Least‑squares method
Explanation - The frequency sampling method directly defines the filter’s response at discrete frequencies and calculates coefficients accordingly.
Correct answer is: Frequency sampling method
Q.109 Which of the following is a key property of an ideal low‑pass filter?
Zero phase response
Infinite impulse response
Flat magnitude response in the passband
None of the above
Explanation - An ideal low‑pass filter perfectly transmits all frequencies below the cutoff with no ripple in the passband.
Correct answer is: Flat magnitude response in the passband
Q.110 Which of the following best describes the concept of 'aliasing'?
The intentional shifting of a signal’s phase
The distortion of a signal due to inadequate sampling
The removal of high‑frequency components
The amplification of low‑frequency components
Explanation - Aliasing occurs when a signal is sampled below its Nyquist rate, causing high‑frequency components to fold into lower frequencies.
Correct answer is: The distortion of a signal due to inadequate sampling