Test your basic knowledge |

Digital Audio

Subject : engineering
Instructions:
  • Answer 50 questions in 15 minutes.
  • If you are not ready to take this test, you can study here.
  • Match each statement with the correct term.
  • Don't refresh. All questions and answers are randomly picked and ordered every time you load a test.

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






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






3. Unit of measurement that is equal to one millionth of a meter






4. Very quiet digital amplifier that produces a series of output pulses with the audio signal coded the same as the width of the output pulses; Pulses are used to represent wave forms and are either on or off; Intense signals have long pulses with short






5. High channel count; 64 channels on one cable; Coaxial cable with BNC connector or fiber optic with ST1 connector






6. Stores only one copy of a stereo signal and assigns it to both channels in order to save 50% of original bandwidth






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






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






9. More accuracy in low amplitudes and less in higher amplitudes






10. Process that begins with a fast FFT analysis of the spectra of two input signals - then the multiplication of like frequencies - and IFFT to finalize the process






11. 'Reconstructing' part of digital audio






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






13. French mathematician that noted that any complex sound can be broken down into a series of component pure tones






14. Accuracy with which an electronic system reproduces the sound or image of its input signal






15. Sony sigma- delta modulation based technology that bypasses the decimation and interpolation steps found in PCM converters






16. MPEG; Standardizing body of audio coding






17. 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)






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






19. Reduces A/D sample rate from the oversampled rate to nominal rate by turning series of one- bit samples into a series of multi- bit PCM samples; (ex. => 2.8MHz sample rate converted to 44.1kHz and simultaneously converts 1-Bit samples to multi- bit);






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






21. Describes various optical disc characteristics including the size and shape of the disc - the size of pits - the speed at which the disc spins - and a multitude of aspects regarding the specifications of the player itself






22. Six channel (five speakers and one subwoofer for bass) digital surround sound system by Dolby






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






24. Measurement at regular intervals of the amplitude of a varying waveform (in order to convert it to digital form); There must be a minimum of 2 samples for each cycle in a waveform






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






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






27. Overtones that contribute to the timbre of a sound and make up a complex waveform's physical characteristics






28. Increases D/A sample rate from nominal rate to oversampling rate by turning series multi- bit PCM samples into 1- bit samples; (ex. => 44.1kHz sample rate converted to 2.8MHz and simultaneously converts multi- bit samples to 1- bit); Low Pass Filter






29. Used as the main disc from which other discs are made; Composed of ground glass with a very fine photoresistor layer; An imaging laser burns pit and land patterns in preparation for duplication






30. Toshiba developed digital audio interface utilizes fiber optics as a transmission medium.






31. ADAT Optcal; 8- in/8- out on two cables; Fiber- optic - TOSLINK connector






32. Computer program or device capable of encoding and/or decoding a digital data stream with the end result being a reduced file size






33. Describes acceptable data - performances both offered and essential for a disc player - and the complete user experience






34. Measure of sound pressure over the frequency spectrum - for which a listener perceives a constant loudness when presented with pure steady tones






35. How Loud (Y-Axis) & How Fast (X-Axis)






36. Number or variable that represents the number of times the base of a power is used as a factor; Defines magnitude






37. Digital Word -> Series of Resistors (each with assigned charges) -> Sample- and-Hold Circuit -> Anti-Imaging Filter (Smoothing Filter) -> Reconstructed Sample






38. Governs the frequency response of a digital system; The highest- frequency component that can be captured with a sampling rate; always 1/2 of sampling rate; Also called the limiting frequency






39. 4.38 GB; SS/SL






40. The frequency above or below which attenuation begins in a filter circuit






41. Ultra low- latency - 512- channel (on a gigabit network) - less flexible AoE format; Routed like audio cables...not network cables






42. Signal conversions are mixed with playback tracks resulting in near-zero latency






43. ABR; Codecs that encode data by determining how dense or sparse areas of the audio are while also keeping bit rate within specified limits to avoid rebuffering






44. Improper (false) sampling of high frequencies that cause much lower frequencies to be produced when the audio is reconstructed; Foldover happens at higher frequencies as harmonics reach levels that exceed the Nyquist limit.






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






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






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






48. Inner ear component that attaches to the stapes and helps to decrease the amplitude of vibrations; Causes the masking phenomenon






49. Multi-Bit Words; (Pulse Code Modulation)






50. Occurs as data is assembled into meaningful bits or information and as left & right channels are separated