Signal Processing Practice MCQs

Questions: 15 | Attempted: 0

1. A signal has a maximum frequency component of 4 kHz. To avoid aliasing and allow for a practical guard band, what is the standard sampling rate used in PCM (Pulse Code Modulation) for telephony?

A) 4 kHzB) 8 kHzC) 16 kHzD) 32 kHz

2. A system is considered 'Linear' if it satisfies which of the following properties?

A) Causality and StabilityB) Homogeneity and Additivity (Superposition)C) Time-invariance onlyD) Finite Impulse Response

3. During the Analog-to-Digital Conversion (ADC) process, what precisely is 'Quantization Error'?

A) The delay introduced by the digital signal processorB) The loss of signal caused by long copper wiresC) The unavoidable difference (noise) between the continuous, exact analog voltage value and the nearest discrete mathematical step value assigned to itD) The distortion caused by sampling a signal below the Nyquist rate

4. What is the primary technical purpose of applying A-law or μ-law companding to an analog voice signal before it is uniformly quantized and digitized?

A) To increase the overall bandwidth of the voice channelB) To encrypt the analog signal so it cannot be wiretappedC) To improve the Signal-to-Quantization-Noise Ratio (SQNR) by applying non-uniform amplification, giving smaller amplitude signals more discrete steps than larger amplitude signalsD) To convert the signal from an analog voltage to a digital optical pulse

5. According to the Nyquist-Shannon sampling theorem, what detrimental effect occurs if an analog baseband signal is digitized using a sampling rate that is strictly less than twice its highest frequency component?

A) Quantization NoiseB) Phase JitterC) AliasingD) Harmonic Distortion

6. The ITU-T G.711 standard defines two specific companding algorithms for PCM digital voice. Which algorithm is the standard in India and Europe, and which is the standard in North America and Japan?

A) A-Law in India/Europe and μ-Law in North America/JapanB) μ-Law in India/Europe and A-Law in North America/JapanC) ADPCM in India/Europe and PCM in North America/JapanD) G.729 in India/Europe and G.711 in North America/Japan

7. In standard telecommunications Pulse Code Modulation (PCM), if a voice signal is sampled at 8 kHz and each sample is quantized into 8 bits, what is the resulting uncompressed data rate of a single voice channel?

A) 8 KbpsB) 16 KbpsC) 32 KbpsD) 64 Kbps

8. Which fundamental mathematical operation is used in Digital Signal Processing (DSP) to calculate the output of a Linear Time-Invariant (LTI) system when given the input signal and the system's impulse response?

A) IntegrationB) DifferentiationC) ConvolutionD) Interpolation

9. What is the primary mathematical purpose of applying a Fourier Transform to a telecommunications signal?

A) To convert an analog signal into a digital signalB) To decompose a complex time-domain signal into its constituent sine and cosine frequencies, transforming it into the frequency domainC) To compress the file size of an audio recordingD) To amplify the voltage of a weak signal

10. In the practical implementation of an Analog-to-Digital Converter (ADC), what is the vital role of the 'Anti-Aliasing Filter' placed immediately before the sampling circuit?

A) It acts as a low-pass analog filter that completely strips away any frequencies higher than the Nyquist limit before the signal is sampled, physically preventing high-frequency noise from folding back into the basebandB) It boosts the high frequencies to make the audio sound clearerC) It converts the analog voltage into binary 1s and 0sD) It matches the impedance of the microphone to the amplifier

11. When designing digital filters, what is the primary structural difference between an IIR (Infinite Impulse Response) filter and an FIR (Finite Impulse Response) filter?

A) IIR filters only process analog signals, while FIR filters process digital signalsB) IIR filters utilize feedback loops (using previous outputs to calculate new outputs), whereas FIR filters do not use feedback, making them unconditionally stableC) IIR filters consume no power, while FIR filters consume massive amounts of powerD) IIR filters can only act as low-pass filters, while FIR filters act as high-pass filters

12. In digital quantization, what is the well-known '6 dB Rule' regarding the Signal-to-Quantization-Noise Ratio (SQNR)?

A) Adding 6 dB of power doubles the transmission distanceB) For every 1 bit added to the resolution of an ADC (e.g., upgrading from 8-bit to 9-bit), the SQNR improves by approximately 6 dBC) The noise floor of a digital signal is always exactly -6 dBmD) A signal must be at least 6 dB louder than the background noise to be digitized

13. Why is the Fast Fourier Transform (FFT) universally used in modern telecommunications (such as 4G/5G OFDM generation) instead of the standard Discrete Fourier Transform (DFT)?

A) FFT is an analog algorithm, making it compatible with older antennasB) FFT drastically reduces the computational complexity from O(N²) down to O(N log N), making real-time frequency processing mathematically possible on modern silicon chipsC) FFT requires less bandwidth than DFTD) FFT perfectly eliminates all background thermal noise

14. In the context of the Shannon-Hartley theorem, how is the absolute maximum channel capacity (C) related to the channel's bandwidth (B) and the Signal-to-Noise Ratio (S/N)?

A) C = B + (S/N)B) C = B * log2(1 + S/N)C) C = (S/N) / BD) C = B² * (S/N)

15. What is the primary advantage of processing signals digitally (DSP) rather than utilizing traditional analog circuitry (resistors, capacitors, and op-amps)?

A) Digital processing requires absolutely zero electrical powerB) Digital systems are perfectly reproducible, do not suffer from component aging or temperature drift, and allow for mathematically perfect filters that are physically impossible to build with analog componentsC) Digital signals travel faster than the speed of lightD) Digital processing guarantees zero latency from input to output