Digital Signal Processing (DSP) MCQs January 8, 2026July 11, 2024 by u930973931_answers 50 min Score: 0 Attempted: 0/50 Subscribe 1. Which of the following is NOT a characteristic of digital signals compared to analog signals? (A) Ease of processing (B) Immunity to noise (C) Infinite resolution (D) Ease of storage 2. The process of converting a continuous signal into a discrete signal is known as: (A) Filtering (B) Quantization (C) Sampling (D) Modulation 3. Nyquist sampling theorem states that to avoid aliasing, the sampling rate should be: (A) Equal to the signal bandwidth (B) At least twice the signal bandwidth (C) Half of the signal bandwidth (D) Irrelevant to the signal bandwidth 4. The number of bits used to represent each sample in a digital signal affects: (A) The amplitude resolution (B) All of the above (C) The dynamic range (D) The frequency resolution 5. The process of converting a digital signal back into an analog signal is called: (A) Modulation (B) Demodulation (C) Digital-to-analog conversion (D) Sampling 6. In the context of digital filters, FIR stands for: (A) Fast Impulse Response (B) Frequency Invariant Response (C) Finite Impulse Response (D) Filtered Impulse Response 7. Which of the following filters is non-recursive? (A) Butterworth filter (B) IIR filter (C) FIR filter (D) Chebyshev filter 8. The Z-transform is used in DSP to: (A) Analyze digital filters in the frequency domain (B) Convert analog signals to digital signals (C) Quantize digital signals (D) None of the above 9. A signal has a sampling rate of 8000 Hz. What is its Nyquist frequency? (A) 2000 Hz (B) 8000 Hz (C) 16000 Hz (D) 4000 Hz 10. Which of the following techniques is used to reduce quantization errors in analog-to-digital conversion? (A) Sampling theorem (B) Aliasing (C) Nyquist criterion (D) Oversampling 11. The Discrete Fourier Transform (DFT) is used to convert: (A) Analog signals to digital signals (B) Digital signals to analog signals (C) Time-domain signals to frequency-domain signals (D) Continuous signals to discrete signals 12. Which window function is commonly used in FIR filter design to minimize the side lobes in the frequency domain? (A) Rectangular window (B) Blackman window (C) Bartlett window (D) Hamming window 13. The impulse response of a system can be found by taking the ________ of its transfer function. (A) Fourier transform (B) Z-transform (C) Laplace transform (D) Inverse Fourier transform 14. Which of the following statements about digital signal processing is true? (A) It requires more memory compared to analog processing. (B) It is less susceptible to noise compared to analog processing. (C) It can only process signals sampled at a low rate. (D) It is limited in its ability to handle complex algorithms. 15. A system that does not change its characteristics over time is known as: (A) Time-varying system (B) Time-invariant system (C) Linear system (D) Causal system 16. Which of the following is a common application of DSP? (A) Image processing (B) Speech recognition (C) All of the above (D) Radar systems 17. The process of converting an analog signal into a digital signal is called: (A) Modulation (B) Demodulation (C) Quantization (D) Sampling 18. Which of the following is NOT a type of digital filter? (A) Kalman (B) Butterworth (C) Chebyshev (D) Bessel 19. The primary advantage of digital filters over analog filters is: (A) They are cheaper to implement (B) They are easier to design (C) They have better phase response (D) They have better stop-band rejection 20. The main drawback of FIR filters compared to IIR filters is: (A) Higher computational complexity (B) Limited flexibility in frequency response (C) Ringing in the time domain (D) None of the above 21. In digital signal processing, decimation refers to: (A) Increasing the sampling rate (B) Converting digital signals to analog signals (C) Converting analog signals to digital signals (D) Decreasing the sampling rate 22. A DSP system with a quantization step size of 2 volts has how many quantization levels? (A) 2 (B) 4 (C) 16 (D) 8 23. Which of the following is used to reduce the computational complexity of the FFT algorithm? (A) Inverse FFT (B) Decimation-in-time algorithm (C) Radix-2 algorithm (D) Zero-padding 24. A digital signal has a maximum frequency component of 2000 Hz. What should be the minimum sampling rate to avoid aliasing? (A) 4000 Hz (B) 2000 Hz (C) 8000 Hz (D) 1000 Hz 25. Which of the following is an advantage of using the Hamming window in FIR filter design? (A) Maximum stop-band attenuation (B) Narrow main lobe (C) Low pass-band ripple (D) High computational efficiency 26. Quantization error in analog-to-digital conversion can be reduced by: (A) Decreasing the sampling rate (B) Decreasing the number of quantization levels (C) Increasing the sampling rate (D) Increasing the number of quantization levels 27. The process of shifting a signal in the time domain is equivalent to: (A) Convolution in the frequency domain (B) Differentiation in the frequency domain (C) Addition in the frequency domain (D) Multiplication by a complex exponential in the frequency domain 28. The primary difference between FIR and IIR filters is: (A) IIR filters have linear phase response, while FIR filters have nonlinear phase response. (B) IIR filters are always unstable, while FIR filters are stable. (C) FIR filters have a higher order than IIR filters. (D) FIR filters have a finite impulse response, while IIR filters have an infinite impulse response. 29. The process of recovering the original signal from its quantized version is called: (A) Demodulation (B) Reconstruction (C) Modulation (D) Decimation 30. Which of the following is a limitation of the discrete Fourier transform (DFT) compared to the continuous Fourier transform (CTFT)? (A) It assumes infinite periodic extension of the signal. (B) It requires more computational resources. (C) It can only be applied to time-domain signals. (D) It does not provide frequency resolution. 31. The process of extracting specific frequency bands from a signal is known as: (A) Modulation (B) Filtering (C) Demodulation (D) Sampling 32. The phase response of a digital filter describes: (A) How the filter affects the magnitude of different frequency components. (B) How the filter introduces distortion in the time domain. (C) How the filter delays different frequency components. (D) How the filter amplifies different frequency components. 33. Which of the following is a drawback of using a high-order FIR filter? (A) High pass-band ripple (B) Limited stop-band attenuation (C) None of the above (D) Increased computational complexity 34. In DSP terminology, the process of finding the time-domain output of a system given its impulse response and input signal is called: (A) Demodulation (B) Correlation (C) Modulation (D) Convolution 35. Which of the following statements about the FFT algorithm is true? (A) It reduces the number of computations compared to the direct DFT computation. (B) It computes the DFT in O(N^2) time complexity. (C) It is used primarily for analog signal processing. (D) It is slower than the DFT for large inputs. 36. A signal with a bandwidth of 4 kHz is sampled at 10 kHz. To avoid aliasing, what is the maximum allowable bandwidth of the input signal? (A) 4 kHz (B) 5 kHz (C) 2 kHz (D) 10 kHz 37. Which of the following is NOT a common window function used in FIR filter design? (A) Gibbs window (B) Hanning window (C) Kaiser window (D) Rectangular window 38. The main purpose of zero-padding in FFT computation is to: (A) Reduce the computational complexity (B) Improve frequency resolution (C) Decrease the amplitude of spectral leakage (D) Increase the time resolution 39. The process of converting a digital signal into an analog signal is called: (A) Modulation (B) Digital-to-analog conversion (C) Quantization (D) Sampling 40. Which of the following is a characteristic of a causal system? (A) The system can produce output without any input. (B) The output depends only on current and past inputs. (C) The system has a symmetric impulse response. (D) The system is linear but time-varying. 41. In DSP, the process of interpolating between samples of a signal is called: (A) Decimation (B) Reconstruction (C) Zero-padding (D) Aliasing 42. Which of the following is NOT a type of digital modulation technique? (A) Amplitude modulation (AM) (B) Pulse Code Modulation (PCM) (C) Frequency modulation (FM) (D) Phase modulation (PM) 43. A DSP system with a sampling rate of 20 kHz is used to process a signal with a maximum frequency component of 10 kHz. To avoid aliasing, what should be done? (A) Increase the sampling rate to 30 kHz (B) Use a low-pass filter with a cutoff frequency of 10 kHz (C) Decrease the sampling rate to 10 kHz (D) Use a high-pass filter with a cutoff frequency of 10 kHz 44. Which of the following statements about FIR filters is true? (A) They can have a recursive structure. (B) They are always stable. (C) They have infinite impulse response. (D) They are typically more computationally efficient than IIR filters. 45. The process of combining two signals to create a new signal is known as: (A) Filtering (B) Convolution (C) Modulation (D) Correlation 46. Which of the following transforms is commonly used to analyze signals in the frequency domain? (A) Fourier transform (B) Laplace transform (C) Z-transform (D) Hilbert transform 47. Which of the following statements about the amplitude resolution of a digital signal is true? (A) It depends on the sampling rate. (B) It is fixed by the number of bits used for quantization. (C) It is infinite. (D) It does not affect the signal quality. 48. The process of removing high-frequency components from a signal is called: (A) Filtering (B) Demodulation (C) Decimation (D) Modulation 49. Which of the following statements about the phase response of a digital filter is true? (A) It affects only the magnitude of the signal. (B) It is not important in digital signal processing. (C) It describes how the filter delays different frequency components. (D) It is linear for all digital filters. 50. Which of the following is an advantage of using the Chebyshev filter over other types of filters? (A) Maximum flatness in the pass-band (B) Linear phase response (C) Sharp roll-off in the stop-band (D) Minimum overshoot in the step response