The Digital Audio Quiz! MCQ Trivia

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The Digital Audio Quiz! MCQ Trivia - Quiz


Do you think you can pass this quiz about digital audio? Digital audio is a technology utilized to record, store, manipulate, create, and reproduce sound using audio signals that have been encrypted in digital form. It also refers to the sequence of discreet samples that are taken from the analog waveform. Take on this quiz and look at how much you know about digital audio.


Questions and Answers
  • 1. 

    Digital audio is 

    • A.

      Continuous in time

    • B.

      Discrete in time

    • C.

      A waste of time

    Correct Answer
    B. Discrete in time
    Explanation
    The sampling process chops the audio up into discrete measurements, which are quantized in time by the sampling rate (E.G. 44.1kHz) and in amplitude by the number of bits (E.G. 16 bits) The sampling rate affects the bandwidth and the bit depth affects the dynamic range (noise level)

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  • 2. 

    The first component in the D/A signal train is an

    • A.

      Anti alliteration filter

    • B.

      Anti apartheid filter

    • C.

      Anti alienation filter

    • D.

      Anti aliasing filter

    Correct Answer
    D. Anti aliasing filter
    Explanation
    Aliasing is a phenomenon whereby false (alias) frequencies are created when there are not enough samples to fully represent the waveform. A visual form of aliasing is the illusion of wagon wheels going backward in Westerns. The sampling rate of film is around 30 frames per second.

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  • 3. 

    The purpose of the Anti-aliasing filter is to

    • A.

      Low pass filter the audio at a frequency related to the sampling rate

    • B.

      High pass filter the audio at the sampling rate

    Correct Answer
    A. Low pass filter the audio at a frequency related to the sampling rate
    Explanation
    In digital audio, the anti-aliasing filter usually filters out anything above 20kHz which ensures that there are at least 2 samples to encode the highest frequency if the sampling rate is above 40kHz. (e.g. 44.1 kHz)

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  • 4. 

    The cut off point of the filter has to be less than:-

    • A.

      One third of the sampling rate

    • B.

      The sampling rate

    • C.

      Half the sampling rate

    • D.

      Two thirds of the sampling rate

    Correct Answer
    C. Half the sampling rate
    Explanation
    Half the sampling rate is sometimes called the Nyquist frequency after the Swedish mathematician who demonstrated that it takes 2 samples to represent a frequency and that less than this will produce alias artifacts.

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  • 5. 

    This cut off frequency is named after a Swedish mathematician. It is called:-

    • A.

      The Nyquist frequency

    • B.

      The Blumlein frequency

    • C.

      The Einstein frequency

    • D.

      The Wittgenstein frequency

    Correct Answer
    A. The Nyquist frequency
    Explanation
    See the explanation for the previous question. Blumlein was an early sound engineer who demonstrated that a stereo recording could be made with 2 microphones in a suitable pattern. He is nothing to do with digital audio, however. Einstein was a famous theoretical mathematician and physicist and Wittgenstein was a philosopher.

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  • 6. 

    The sampling rate has a direct effect on:

    • A.

      The dynamic range of the audio

    • B.

      The bandwidth of the audio

    Correct Answer
    B. The bandwidth of the audio
    Explanation
    The greater the sample rate the higher the frequency that can be allowed into the system. If you sample at 32kHz your audio will have to be filtered to around 15 kHz. (e.g. FM radio)

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  • 7. 

    The amplitude measurement of each sample is rounded to the nearest bit. What is this process called

    • A.

      Quantisation

    • B.

      Sample and hold

    • C.

      Clocking

    • D.

      Compression

    Correct Answer
    A. Quantisation
    Explanation
    This rounding is called quantisation error and it is where most of the noise comes from in digital audio.

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  • 8. 

    What unwanted effect might quantization have if the bit depth is low?

    • A.

      Digital distortion

    • B.

      Quantisation noise

    • C.

      Alias noise

    Correct Answer
    B. Quantisation noise
    Explanation
    Quantization noise is the unwanted effect that can occur when the bit depth is low. When the bit depth is low, there are fewer levels available to represent the analog signal, resulting in a loss of detail and accuracy. This loss of detail leads to quantization errors, which manifest as quantization noise. This noise is perceived as a distortion or background noise in the digital signal, reducing the overall quality of the audio or image.

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  • 9. 

    Bit depth affects:

    • A.

      The dynamic range of the audio

    • B.

      The bandwidth of the audio

    Correct Answer
    A. The dynamic range of the audio
    Explanation
    The more numbers you have the louder you can record. Engineers usually go for at least 24 bits at the tracking stage even though it has to be re-clocked to 16 bits on transfer to CD.

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  • 10. 

    Which of the following might be considered professional sampling rates and bit depths?

    • A.

      16 bits 44.1 kHz

    • B.

      24 bits 96 kHz

    • C.

      16 bits 48 kHz

    • D.

      12 bits 32 kHz

    • E.

      24 bits 192 kHz

    Correct Answer(s)
    A. 16 bits 44.1 kHz
    B. 24 bits 96 kHz
    C. 16 bits 48 kHz
    E. 24 bits 192 kHz
    Explanation
    D Would yield very lo fi audio. A is the current minimum standard. B and E are becoming increasingly popular.

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  • 11. 

    Which of the following pathways represents the Digital audio process?

    • A.

      Anti aliasing filter, Sample and hold, (processing and editing in the digital domain) data recovery filter, DAC

    • B.

      Data recovery filter, Sample and hold, (processing and editing in the digital domain) anti aliasing filter, DAC

    • C.

      Sample and hold, Anti aliasing filter, (processing and editing in the digital domain) data recovery filter, DAC

    • D.

      Anti aliasing filter, Sample and hold, (processing and editing in the digital domain) DAC, data recovery filter

    Correct Answer
    D. Anti aliasing filter, Sample and hold, (processing and editing in the digital domain) DAC, data recovery filter
    Explanation
    The AAF first limits the bandwidth to something that the sampling rate can cope with, then the numbers are generated by sample and hold. The signal is now digital and can be processed and edited. After this, on playback, the numbers are converted back to a digital wave which is then filtered (Data recovery) to remove the spurious high frequencies and smooth the waveform to completely recover the original audio.

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