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Digital Audio

Subject : engineering
Instructions:
  • Answer 50 questions in 15 minutes.
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  • Match each statement with the correct term.
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This is a study tool. The 3 wrong answers for each question are randomly chosen from answers to other questions. So, you might find at times the answers obvious, but you will see it re-enforces your understanding as you take the test each time.
1. Twice as many samples as the highest frequency at minimum; Sampling rate totally controls frequency response






2. Fractional part of a floating- point number; Also called the mantissa; Defines precision






3. ADPCM; Pulse code modulation that produces a digital signal with a lower bit rate than standard PCM; Records only the difference between samples






4. Measure of the amplitude of a longitudinal wave






5. Series of dots and dashes representing the letters of the alphabet; Most common letters are represented by the shortest dots and dashes; Example of entropy coding






6. HD Audio format; Lossless Compression; 24- bit/96 kHz; 5.1 Surround or 24- bit / 192 kHz stereo sound






7. Based on psychoacoustics - these are the basis of frequency analysis for a perceptual codec;






8. Softest sound that can be heard by the average human ear (0 dB)






9. The act of a frequency swinging back and forth with a steady - uninterrupted rhythm






10. Deviation from a normal - steady pulse or tick of a clock that contributes to misrepresentation of a signal; Result of small timing irregularities that become magnified during the transmission of digital signals as the signals are passed from one dev






11. The continuous loss of signal strengths as a signal travels through a medium






12. Eight channel digital surround sound system by Dolby






13. Channels are processed one at a time and the results are stored on multiple CPU buffers that alternately send data as DAW requests the data for playback; First few seconds are relayed to the buffer prior to processing and playback and it continuously






14. Also known as equal loudness curves; Graph that indicates the average ear sensitivity to different frequencies at different SPL levels (as volume increases - these curves flatten out)






15. EDL; Final list of samples used in the audio editing process; Identified by time code






16. When recording you want the smallest buffer available; When mixing you want the largest buffer available






17. The elapsed time it takes for a packet of data to arrive at its destination; Lagging or pause of an audio signal as digital processing occurs; Can be managed utilizing several forms of 'audio monitoring'






18. Industry Standards: -6 dB Peak = -20 RMS Meter






19. Leaving space at beginning and ending of song for data crunching during mastering; Last step in mastering process






20. Perceptual coding technique that uses louder sounds of a similar frequency to decide what information is to be saved during data reduction






21. The difference between the analog value and the approximated digital value due to the 'rounding' that occurs while converting the analog signal to digital






22. (Amplitude Based) Amplitude: Voltage; Quantization; Bit Depth; Quantization Intervals; Quantization Noise; [Signal:Quantization Noise Ratio]; Dither; Dynamic Range






23. VBR; Most common & best data reduction technique; Codecs that encode data by determining how dense or sparse areas of the audio are; Can result in buffering issues






24. A method of representing real numbers using a mantissa and an exponent






25. Voltage -> Dither -> Anti-Aliasing (Low Pass Filter) -> Sample & Hold Circuit -> Successive Approximation/Quantizer --) 100111010 (PCM Audio File)






26. 16-Bit; 44.1 kHz; PCM; Stereo






27. The set of rules that computers use to move files from one computer to another on an internet






28. Unit of measurement that is equal to one billionth of a meter






29. Contains all even and odd harmonics associated with a fundamental tone - making it a rich source for modeling other sounds; Amplitude of each overtone decreases exponentially as a ratio of the harmonic's frequency to that of the fundamental






30. The difference in volume between the loudest and quietest sounds of a source






31. Very selective method of lowering buffer levels by halting different levels of audio processing






32. As sample rate is increased more room is created for a smoother slope of the attenuation band because Nyquist limit extends well beyond range of hearing with each increase






33. Waveform of a pure tone showing simple harmonic motion






34. 8- in/8- out on one cable; 25- pin D- sub connector






35. A sample- by- sample operation on two signals






36. Circuit that interprets the meaning of the symbols as they were chosen and arranged by the encode






37. Used when the reference pressure of a sound is 20 microPa (0.00002); Sound Pressure Level; Measure of amplitude






38. RAM holds in memory audio data before it is transferred to the memory controller; Certain amount of data is processed before large amounts of data are streamed to prevent latency






39. A network communications protocol that specifies how machines will exchange data; Uses a broadcast system in which one machine transmits its message on the communication medium and the other machines listen for messages directed to them






40. Amplitude meter that takes the square root of all instantaneous amplitudes and averages them to find a mean and squares that value; Useful with particularly complex waveforms






41. Only 2 digits used; The value of each place (ones - hundreds - etc.) are as follows from greatest to least: 128 - 64 - 32 - 16 - 8 - 4 - 2 - 1






42. The more bits allocated during quantization - the more accurate the measurement






43. Number of bits used to represent the smallest unit of information in an audio file; Greater bit depth = better quality audio






44. Process of building a complex tone by starting with the fundamental frequency and adding pure tone harmonics






45. Algorithm uses matrix of a mid/side microphone pair to determine a side signal & that signal is reduced then distributed as code in stereo






46. A digital filter's time domain output sequence when the input is a single sample is input






47. The loudest point of a Full Scale system






48. If a signal is sampled at a rate higher than twice the highest significant signal frequency and at evenly spaced intervals - then the samples contain all the information of the original signal






49. Based on Full Scale (dB/FS); -6dB represents a loss of one bit so account for this when calculating






50. Increases or decreases the digital signal so that the loudest sample is brought up to 0dBfs; Uses all bits from dynamic range and makes it even from track to track







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