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Question 1Fundamentals of Audio and Video Engineering

A sine wave completes 500 full cycles in 2 seconds. What is its frequency?

  1. 1000 Hz
  2. 500 Hz
  3. 250 Hz Correct answer.
  4. 125 Hz

Explanation

Frequency is the number of cycles per second, so dividing 500 cycles by 2 seconds gives 250 Hz. Frequency is independent of amplitude, which is not specified here and does not affect the count of cycles.

Question 2Fundamentals of Audio and Video Engineering

What is the period of a 50 Hz waveform?

  1. 2 ms
  2. 50 ms
  3. 20 ms Correct answer.
  4. 500 ms

Explanation

Period is the reciprocal of frequency: 1 divided by 50 Hz equals 0.02 seconds, which is 20 milliseconds. This relationship holds for any periodic waveform regardless of shape.

Question 3Fundamentals of Audio and Video Engineering

Which statement correctly describes the relationship between wavelength, frequency, and the speed of sound?

  1. Wavelength equals the speed of sound divided by frequency Correct answer.
  2. Wavelength equals frequency divided by the speed of sound
  3. Wavelength equals the speed of sound multiplied by frequency
  4. Wavelength is independent of frequency at a fixed temperature

Explanation

The wave equation states that speed equals frequency times wavelength, so wavelength is speed divided by frequency. At a fixed speed of sound, higher frequencies therefore produce shorter wavelengths.

Question 4Fundamentals of Audio and Video Engineering

A sound pressure level increases from 80 dB SPL to 86 dB SPL. What is the approximate change in sound pressure?

  1. Increase of 6 percent
  2. Increase of 6 pascals
  3. Tripling of pressure
  4. Doubling of pressure Correct answer.

Explanation

A 6 dB increase corresponds to roughly doubling the sound pressure, because 20 log10(2) is approximately 6 dB. Decibels are logarithmic ratios, so a small dB change means a large linear change.

Question 5Fundamentals of Audio and Video Engineering

What does 0 dB SPL represent?

  1. A pressure of one pascal
  2. The threshold of pain
  3. Complete absence of air pressure variation
  4. The reference pressure of 20 micropascals Correct answer.

Explanation

0 dB SPL is defined against a reference sound pressure of 20 micropascals, roughly the threshold of hearing at mid frequencies. It is a reference point, not silence.

Question 6Fundamentals of Audio and Video Engineering

Which property of a sound wave primarily determines its perceived pitch?

  1. Peak amplitude
  2. Waveform symmetry
  3. Phase relative to another wave
  4. Fundamental frequency Correct answer.

Explanation

Pitch perception is most closely tied to the fundamental frequency of a periodic sound. Amplitude relates to loudness and waveform shape relates to timbre.

Question 7Fundamentals of Audio and Video Engineering

A complex tone contains harmonics at 200, 400, 600, and 800 Hz. What is its fundamental frequency?

  1. 100 Hz
  2. 200 Hz Correct answer.
  3. 600 Hz
  4. 400 Hz

Explanation

The harmonics are integer multiples of the fundamental, and the spacing between adjacent harmonics equals the fundamental. Here the spacing is 200 Hz, so the fundamental is 200 Hz.

Question 8Fundamentals of Audio and Video Engineering

Which type of microphone is most sensitive to sound arriving from one direction and rejects sound from the rear?

  1. Boundary
  2. Cardioid Correct answer.
  3. Figure-of-eight
  4. Omnidirectional

Explanation

A cardioid polar pattern has maximum sensitivity at the front and a null at the rear, making it useful for rejecting rearward sound. Omnidirectional and figure-of-eight patterns behave differently.

Question 9Fundamentals of Audio and Video Engineering

Why is phantom power typically specified as 48 volts?

  1. It is required to charge condenser capsules to full sensitivity
  2. It is a standard DC supply delivered over balanced microphone cables to power active electronics Correct answer.
  3. It is the peak voltage of a standard audio signal
  4. It matches the nominal line-level voltage used by mixing consoles

Explanation

Phantom power is a standardized DC supply, commonly 48 volts, sent over the same balanced cable as the audio signal to power active circuitry such as condenser microphones. The balanced nature of the connection lets the DC appear equally on both signal conductors.

Question 10Fundamentals of Audio and Video Engineering

A balanced audio connection primarily reduces which problem?

  1. Loudspeaker impedance mismatch
  2. High-frequency distortion within the source device
  3. Common-mode interference picked up along the cable Correct answer.
  4. Quantization error in digital conversion

Explanation

A balanced connection carries the signal on two conductors with opposite polarity and a separate shield, so interference coupled equally onto both conductors is cancelled at the differential input. This is why balanced lines are preferred for long runs.

Question 11Fundamentals of Audio and Video Engineering

Which connector is a standard balanced three-pin audio connector widely used on microphones?

  1. XLR Correct answer.
  2. Binding post
  3. TS jack
  4. RCA

Explanation

The XLR connector has three pins for hot, cold, and shield, supporting balanced microphone and line connections with a locking mechanism. RCA and TS jack connectors are typically unbalanced.

Question 12Fundamentals of Audio and Video Engineering

What is the primary function of an anti-aliasing filter in a digital audio system?

  1. To remove quantization noise after conversion
  2. To flatten the frequency response of the loudspeaker
  3. To convert digital samples back into an analog waveform
  4. To attenuate frequency content above half the sample rate before sampling Correct answer.

Explanation

An anti-aliasing filter limits the input bandwidth to below the Nyquist frequency, half the sample rate, so that high-frequency content cannot fold back into the audible band as aliasing. It sits before the sampler in the signal chain.

Question 13Fundamentals of Audio and Video Engineering

According to the Nyquist theorem, what is the minimum sample rate needed to represent a signal containing frequencies up to 20 kHz?

  1. 10 kHz
  2. 20 kHz
  3. 80 kHz
  4. 40 kHz Correct answer.

Explanation

The sample rate must be at least twice the highest frequency present, so 20 kHz content requires a minimum of 40 kHz sampling. Sampling below this causes aliasing.

Question 14Fundamentals of Audio and Video Engineering

What effect does increasing bit depth have in PCM audio?

  1. It raises the Nyquist frequency
  2. It removes the need for an anti-aliasing filter
  3. It lowers the quantization noise floor and increases dynamic range Correct answer.
  4. It reduces the file size for the same duration

Explanation

Each additional bit roughly adds 6 dB of dynamic range by reducing quantization error, so a higher bit depth yields a lower noise floor. Bit depth does not change the sampling bandwidth.

Question 15Fundamentals of Audio and Video Engineering

A 24-bit PCM system has an approximate theoretical dynamic range of how many decibels?

  1. 192 dB
  2. 96 dB
  3. 48 dB
  4. 144 dB Correct answer.

Explanation

Using the rule of roughly 6 dB per bit, 24 bits gives about 144 dB of theoretical dynamic range. A 16-bit system would give about 96 dB.

Question 16Fundamentals of Audio and Video Engineering

What is the purpose of dither when reducing bit depth?

  1. To increase the peak level of the signal
  2. To remove DC offset from the signal
  3. To raise the sample rate before conversion
  4. To randomize quantization error so it becomes low-level noise rather than correlated distortion Correct answer.

Explanation

Dither adds a small amount of noise before quantization so that the resulting error is decorrelated from the signal, converting harsh distortion into a benign hiss. It trades a slightly higher noise floor for cleaner low-level detail.

Question 17Fundamentals of Audio and Video Engineering

Which statement best distinguishes loudness from sound pressure level?

  1. Loudness and sound pressure level are identical quantities measured with different instruments
  2. Loudness is a subjective perception, while sound pressure level is an objective physical measurement Correct answer.
  3. Loudness is measured in pascals, while sound pressure level is measured in hertz
  4. Loudness depends only on frequency, while sound pressure level depends only on amplitude

Explanation

Sound pressure level is a physical quantity derived from measured pressure, whereas loudness is a perceptual quantity that also depends on frequency and duration. Equal SPL tones at different frequencies can be perceived as different loudness.

Question 18Fundamentals of Audio and Video Engineering

Two coherent sine waves of equal amplitude are summed with a phase difference of 180 degrees. What is the resulting amplitude?

  1. Zero Correct answer.
  2. Equal to the original amplitude
  3. Twice the original amplitude
  4. Half the original amplitude

Explanation

At 180 degrees of phase difference the two waveforms are equal and opposite at every instant, so they cancel completely. This is destructive interference at its maximum.

Question 19Fundamentals of Audio and Video Engineering

Which unit expresses absolute acoustic intensity in watts per square metre?

  1. Hz
  2. dB SPL
  3. Pa
  4. W/m² Correct answer.

Explanation

Acoustic intensity is power per unit area, so its unit is watts per square metre. Decibels express a ratio, pascals express pressure, and hertz express frequency.

Question 20Fundamentals of Audio and Video Engineering

An amplifier delivers 100 watts to a loudspeaker. If the input level is increased by 3 dB, what is the approximate new output power?

  1. 300 W
  2. 200 W Correct answer.
  3. 130 W
  4. 103 W

Explanation

A 3 dB increase in power corresponds to a doubling, so 100 watts becomes approximately 200 watts. The 3 dB rule reflects 10 log10(2) for power ratios.

Question 21Fundamentals of Audio and Video Engineering

What does a loudspeaker's sensitivity rating of 90 dB at 1 W and 1 m describe?

  1. The maximum power the loudspeaker can handle continuously
  2. The impedance presented to the amplifier at one metre
  3. The sound pressure level produced on axis at one metre for one watt of input Correct answer.
  4. The frequency response variation across the audio band

Explanation

Sensitivity states the on-axis sound pressure level measured at one metre when the loudspeaker is driven with one watt. It allows comparison of how efficiently different loudspeakers convert power into acoustic output.

Question 22Fundamentals of Audio and Video Engineering

Why does a loudspeaker's impedance vary with frequency?

  1. The voice coil has inductance and the mechanical resonance alters the electrical load Correct answer.
  2. The cable resistance changes with the audio signal
  3. The amplifier changes its output voltage with frequency
  4. The room absorbs different frequencies at different rates

Explanation

A loudspeaker is an electromechanical system: the voice coil inductance raises impedance at high frequencies, while the driver's mechanical resonance produces a peak at low frequencies. The result is a frequency-dependent load rather than a constant resistance.

Question 23Fundamentals of Audio and Video Engineering

In a mixing console, what does the term gain structure refer to?

  1. Managing signal levels through each stage to maximize headroom and minimize noise Correct answer.
  2. Assigning channels to output buses
  3. Setting the equalizer curves for each channel
  4. Choosing the sample rate for recording

Explanation

Gain structure is the practice of setting levels at each stage of the signal path so that signals stay well above the noise floor without clipping. Good gain structure preserves headroom throughout the chain.

Question 24Fundamentals of Audio and Video Engineering

What is the main advantage of a digital audio workstation using non-destructive editing?

  1. Edits permanently overwrite the source recording to save disk space
  2. Edits are applied only when exporting to a compressed format
  3. Edits bypass the need for any digital signal processing
  4. Edits are stored as instructions and the original audio file remains unaltered Correct answer.

Explanation

Non-destructive editing keeps the source file intact and records edits as parameters applied on playback or export. This allows changes to be revised or undone without degrading the original material.

Question 25Fundamentals of Audio and Video Engineering

A video signal with a frame rate of 25 frames per second is displayed on a monitor refreshing at 60 Hz. What problem can this cause?

  1. Loss of colour information in the video signal
  2. Judder caused by uneven frame distribution across refresh cycles Correct answer.
  3. Aliasing in the digital audio conversion stage
  4. Increased audio latency in the playback device

Explanation

When the frame rate does not divide evenly into the refresh rate, frames are displayed for unequal numbers of refresh cycles, producing visible judder. Matching frame and refresh rates, or using pulldown, reduces the effect.

Question 26Fundamentals of Audio and Video Engineering

A technician measures a signal level as +4 dBu. Which reference voltage does the 'u' in dBu indicate?

  1. 0.316 V RMS into any load
  2. 1.0 V peak-to-peak into 600 ohms
  3. 2.0 V RMS open circuit
  4. 0.775 V RMS across 600 ohms Correct answer.

Explanation

dBu is referenced to 0.775 V RMS, the voltage that dissipates 1 mW in a 600-ohm load. Unlike dBm, dBu does not require an actual 600-ohm termination, so the level is valid for modern high-impedance bridging inputs.

Question 27Fundamentals of Audio and Video Engineering

A room's reverberation time RT60 is defined as the time required for which condition to occur after the sound source stops?

  1. The sound pressure level to fall by 60 dB Correct answer.
  2. The sound pressure level to halve in absolute pressure
  3. The direct sound to reach the furthest listener
  4. The reverberant field to exceed the direct field

Explanation

RT60 measures the decay of sound pressure level by 60 dB after the source ceases, typically extrapolated from a 20 or 30 dB decay curve. It is a standard descriptor of how reverberant an enclosed space is at a given frequency band.

Question 28Fundamentals of Audio and Video Engineering

In a video system, what does the term 'genlock' allow multiple devices to do?

  1. Share a single timecode track across recordings
  2. Compress their output streams to a shared bitrate
  3. Distribute audio word clock over the video cable
  4. Synchronise their video timing to a common reference signal Correct answer.

Explanation

Genlock locks the sync generators of several video devices to a common reference so their frames and lines align. Without a shared reference, switching or compositing between sources can produce rolls, tears, or frame slips.

Question 29Fundamentals of Audio and Video Engineering

A cable run carries a line-level signal and picks up audible hum at 50 Hz. Which practice most directly reduces this magnetically induced interference?

  1. Use twisted-pair conductors with a low-impedance source and balanced input Correct answer.
  2. Increase the cable length to average out the induced voltage
  3. Raise the signal level after the interference has been picked up
  4. Replace the cable shield with a thicker dielectric

Explanation

Twisting the pair makes the loop area small and symmetric, so induced voltages cancel at the differential input, while a low source impedance keeps the interference from developing significant voltage. Balancing lets the receiver reject what remains as common-mode noise.

Question 30Fundamentals of Audio and Video Engineering

An audio interface is set to a 48 kHz sample rate. Which range of frequencies can it represent, according to the sampling theorem?

  1. Up to 12 kHz
  2. Up to 96 kHz
  3. Up to 48 kHz
  4. Up to 24 kHz Correct answer.

Explanation

The representable bandwidth extends to half the sample rate, so 48 kHz sampling supports content up to 24 kHz before aliasing. Practical converters roll off slightly below that limit with an anti-aliasing filter.

Question 31Fundamentals of Audio and Video Engineering

A loudspeaker with a nominal impedance of 8 ohms is replaced by a 4-ohm model of identical sensitivity. What happens to the power drawn from a solid-state amplifier delivering a fixed voltage?

  1. It stays the same
  2. It approximately doubles Correct answer.
  3. It approximately halves
  4. It falls to one quarter

Explanation

Power equals voltage squared divided by impedance, so halving the load impedance doubles the power for the same drive voltage. The amplifier must be able to supply the correspondingly higher current without current limiting.

Question 32Fundamentals of Audio and Video Engineering

A microphone preamplifier adds 40 dB of gain to a signal. By what factor is the voltage amplitude multiplied?

  1. 10
  2. 400
  3. 100 Correct answer.
  4. 40

Explanation

Voltage gain in decibels is 20 times the logarithm of the amplitude ratio, so 40 dB corresponds to a ratio of 100. The factor of 10 applies to power ratios, not voltage.

Question 33Fundamentals of Audio and Video Engineering

Why is a low-pass filter placed before the analogue-to-digital converter rather than after it?

  1. To flatten the frequency response of the converter
  2. To increase the effective bit depth of the conversion
  3. To remove content above half the sample rate before it can alias into the audio band Correct answer.
  4. To remove DC offset from the input signal

Explanation

Once sampling occurs, frequencies above the Nyquist limit fold back into the audible band and cannot be separated from legitimate content. Filtering in the analogue domain beforehand prevents that irreversible aliasing.

Question 34Fundamentals of Audio and Video Engineering

A digital audio workstation applies a fade using a destructive edit. What is the practical consequence for the operator?

  1. The audio file is duplicated automatically before the edit
  2. The fade can only be applied to the master bus
  3. The change is stored as metadata and can be undone at any time
  4. The change is written into the audio file and the original samples are lost Correct answer.

Explanation

Destructive editing permanently alters the stored sample data, so undoing requires a backup or re-import of the source. Non-destructive systems instead keep the original media and render edits as instructions.

Question 35Fundamentals of Audio and Video Engineering

In a video production, a 25 fps camera is recorded while a 50 Hz mains-driven light source flickers. What is the most likely visible artefact?

  1. A periodic loss of audio synchronisation
  2. A vertical colour shift across the frame
  3. A rolling horizontal band of brightness
  4. A steady exposure with no flicker because the rates are harmonically related Correct answer.

Explanation

When the flicker frequency is an integer multiple of the frame rate, each frame integrates the same portion of the light cycle, so exposure stays uniform. Problems arise when the rates are not harmonically related, as with 50 Hz lighting shot at 30 fps.

Question 36Fundamentals of Audio and Video Engineering

An amplifier is rated at 200 W continuous into 8 ohms. A programme peak requires 800 W for a few milliseconds. What does this indicate about the amplifier's headroom requirement?

  1. It needs about 3 dB of peak headroom above its continuous rating
  2. It needs about 6 dB of peak headroom above its continuous rating Correct answer.
  3. It needs about 12 dB of peak headroom above its continuous rating
  4. It needs no headroom because peaks are too short to matter

Explanation

A ratio of 800 W to 200 W is a factor of four in power, which equals 10 log10(4), or about 6 dB. Amplifiers must deliver these short peaks without clipping, so the peak capability above the continuous rating defines the usable headroom.

Question 37Fundamentals of Audio and Video Engineering

A stereo pair of microphones is spaced 3 metres apart in a reverberant hall. Which limitation most affects the resulting recording?

  1. Insufficient phantom power reaching both capsules
  2. Excessive proximity effect from the directional pattern
  3. Comb filtering and poor mono compatibility from the large time-of-arrival differences Correct answer.
  4. Loss of low-frequency extension below 200 Hz

Explanation

Widely spaced microphones capture the same source at different times, and summing the two channels to mono produces phase cancellations heard as comb filtering. The large spacing also weakens the coherence of the stereo image.

Question 38Fundamentals of Audio and Video Engineering

A digital console processes audio internally at 96 kHz and must output to a device locked at 48 kHz. Which process is required, and what risk does it carry?

  1. Sample rate conversion, which can introduce aliasing or filtering artefacts if poorly implemented Correct answer.
  2. Word clock distribution, which aligns the frame boundaries
  3. Bit depth truncation, which removes the least significant bits
  4. Quantisation, which maps continuous amplitudes to discrete steps

Explanation

Changing the sample rate requires resampling, either synchronously with a rational ratio or asynchronously with interpolation. Poorly designed converters can fold high-frequency content back into the band or impose audible filtering, so converter quality matters.

Question 39Fundamentals of Audio and Video Engineering

A control room measures 85 dBA at the listening position with pink noise. The engineer raises the monitor level by 10 dB. What is the approximate new sound pressure level, and what practical concern arises?

  1. About 95 dBA, which exceeds levels for extended safe listening without breaks Correct answer.
  2. About 87 dBA, because the A-weighting compresses large changes
  3. About 90 dBA, which remains within safe limits indefinitely
  4. About 105 dBA, because dBA scales non-linearly with level

Explanation

A 10 dB increase raises the level arithmetically on the decibel scale, giving roughly 95 dBA. Sustained exposure at that level exceeds recommended limits for long sessions, so duration and breaks become a genuine hearing-conservation concern.

Question 40Fundamentals of Audio and Video Engineering

A technician records a signal at a sample rate of 44.1 kHz. According to the Nyquist theorem, what is the highest frequency that can be represented without aliasing?

  1. 22.05 kHz Correct answer.
  2. 11.025 kHz
  3. 88.2 kHz
  4. 44.1 kHz

Explanation

The Nyquist theorem states that a sampled system can faithfully represent frequencies only up to half the sample rate, known as the Nyquist frequency. For a 44.1 kHz sample rate, half of this is 22.05 kHz, which is why the standard audio band extends to roughly 20 kHz. Frequencies above this limit must be removed by an anti-alias filter before conversion.

Question 41Audio Systems and Signal Processing

A loudspeaker is fed a 1 kHz sine wave and produces strong output at 2 kHz and 3 kHz in addition to the fundamental. Which measurement best characterizes this behavior?

  1. Signal-to-noise ratio
  2. Intermodulation distortion
  3. Crosstalk
  4. Total harmonic distortion Correct answer.

Explanation

Harmonic distortion is the generation of integer multiples of a single input tone, so energy at 2 kHz and 3 kHz from a 1 kHz stimulus is harmonic distortion. Intermodulation requires two or more input tones and produces sum and difference products. SNR and crosstalk describe noise floor and channel leakage, not harmonic generation.

Question 42Audio Systems and Signal Processing

Which condition must be satisfied for a digital audio system to avoid aliasing when capturing a signal?

  1. The anti-alias filter must have unity gain at Nyquist
  2. The bit depth must be at least 24 bits
  3. The quantizer must use dither
  4. The sample rate must exceed twice the highest frequency component present Correct answer.

Explanation

The Nyquist criterion requires the sampling frequency to be greater than twice the highest frequency in the input signal. Bit depth affects dynamic range and noise, not aliasing. Dither reduces quantization distortion, and filter gain at Nyquist does not by itself prevent aliasing.

Question 43Audio Systems and Signal Processing

A signal is sampled at 48 kHz. A 25 kHz tone is present at the converter input with no anti-alias filtering. At what frequency does it appear in the digital output?

  1. 48 kHz
  2. 1 kHz
  3. 25 kHz
  4. 23 kHz Correct answer.

Explanation

With a 48 kHz sample rate, Nyquist is 24 kHz, so a 25 kHz component folds around Nyquist and lands at 48 − 25 = 23 kHz. The alias is the mirror image across the Nyquist frequency. It does not stay at 25 kHz, and 48 kHz is the sample rate, not a signal frequency.

Question 44Audio Systems and Signal Processing

What is the primary effect of increasing bit depth from 16 to 24 bits in a PCM recording?

  1. It removes the need for an anti-alias filter
  2. It eliminates jitter in the converter clock
  3. It lowers the quantization noise floor and increases dynamic range Correct answer.
  4. It doubles the highest reproducible frequency

Explanation

Each additional bit adds roughly 6 dB of dynamic range, so 24-bit audio has a much lower quantization noise floor than 16-bit. Bit depth does not change the bandwidth limit, which is set by sample rate. It also does not replace anti-alias filtering or remove clock jitter.

Question 45Audio Systems and Signal Processing

Two loudspeakers reproduce the same 100 Hz tone but are wired with opposite polarity. What happens in the listening position equidistant from both?

  1. The tone shifts up in frequency
  2. The 100 Hz tone doubles in level
  3. The 100 Hz tone largely cancels Correct answer.
  4. The tone becomes amplitude modulated at 200 Hz

Explanation

Opposite polarity with equal path length means the two 100 Hz waves arrive in antiphase and cancel. Doubling in level would occur with matched polarity. Polarity reversal does not shift frequency or create modulation sidebands.

Question 46Audio Systems and Signal Processing

Which unit expresses sound pressure level referenced to 20 micropascals?

  1. dB SPL Correct answer.
  2. dBFS
  3. dBu
  4. dBm

Explanation

Sound pressure level is defined in decibels relative to 20 µPa, the nominal threshold of hearing. dBu and dBm are electrical references, and dBFS refers to full scale in digital systems.

Question 47Audio Systems and Signal Processing

A mixer sums two uncorrelated noise sources of equal level. What is the resulting increase in level?

  1. About 10 dB
  2. About 0 dB
  3. About 3 dB Correct answer.
  4. About 6 dB

Explanation

Uncorrelated sources add on a power basis, so two equal noise sources raise the level by about 3 dB. Correlated sources that add coherently would give about 6 dB. Zero dB would mean no addition at all.

Question 48Audio Systems and Signal Processing

Which phenomenon describes the perceived increase in loudness of a sound as its duration increases up to a few hundred milliseconds?

  1. Haas effect
  2. Comb filtering
  3. Temporal integration Correct answer.
  4. Masking

Explanation

Temporal integration describes how the auditory system sums energy over time, so a longer sound is perceived as louder up to a limit. Masking is the reduced audibility of one sound due to another. The Haas effect concerns localization of delayed arrivals, and comb filtering is a spectral interference pattern.

Question 49Audio Systems and Signal Processing

A 1 kHz tone at 80 dB SPL is presented with a 1.1 kHz tone at 60 dB. Which term describes the reduced audibility of the 1.1 kHz tone?

  1. Temporal masking
  2. Harmonic distortion
  3. Simultaneous masking Correct answer.
  4. Frequency aliasing

Explanation

When two sounds are present at the same time, the louder one can mask the weaker one, especially when their frequencies are close. This is simultaneous masking. Temporal masking involves sounds separated in time, aliasing is a sampling artifact, and harmonic distortion is nonlinear generation of new frequencies.

Question 50Audio Systems and Signal Processing

An audio system's measured level is 0.5 dB below a reference. Expressed as a linear amplitude ratio, this is closest to which value?

  1. 0.891
  2. 0.977
  3. 0.500
  4. 0.944 Correct answer.

Explanation

A level change in dB for amplitude is 20·log10(ratio). Solving for −0.5 dB gives 10^(−0.5/20) ≈ 0.944. A ratio of 0.977 corresponds to about −0.2 dB, and 0.891 corresponds to about −1 dB.