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






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






3. Allows for an internal sample rate at multiples of the input and output rates; Alleviates the need for steep 'brickwall' filters; Often combined with internal '1- bit' processing; Increases smoothing effect






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






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






6. 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






7. Samples are duplicated and the playback sampling rate correspondingly increased; Significantly raises the Nyquist limit to a range well beyond human hearing; Processing 'pushes' the distortion resulting from quantization error into these higher frequ






8. 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






9. 4.38 GB; SS/SL






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






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






12. 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






13. Mixing data and control characters in a single operation






14. (Time Based) Frequency: Aliasing; Anti-Aliasing Filter; Sample Rate; Nyquist Limit






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






16. Circuit that seizes voltage values with each tick of an A/D device's internal clock






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






18. Allowance of noise floor below that which is required for the final product






19. Twice as many samples as the highest frequency at minimum; Sampling rate totally controls frequency response






20. In order to play multiple channels at one time data is buffered as read to disk; Latency happens between play command & beginning of playback






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






22. 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);






23. Sample Rate x Bit Depth x # of Channels






24. Lossless Format; Can hold up to 25GB on a single- layer disc and 50GB on a dual- layer disc






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






26. 8- in/8- out on one cable; 25- pin D- sub connector






27. Electromagnetic receptor that detects the radiation known as visible light






28. 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






29. 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.






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






31. Psychoacoustic model of data reduction used for general audio compression that aims to transmit only features perceptible to the human ear; Useful for mastering streaming media






32. A frequency specified for a filter (digital or electronic) the marks the point at which the frequency content of a signal is altered +/- 3dB






33. 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






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






35. Smallest interval measurable by a scientific instrument; Defined by bit rate (sample rate x bit depth)






36. CBR; Codecs encodes data at a constant rate regardless of density of the audio file






37. Decibels Full Scale






38. Having a repeated succession of waves or curves as in a sound waveform






39. Method used in digital recording and reproduction in which a signal is sampled at various points and the resulting value is translated into binary numbers






40. Reduces the percieved distortion due to quantization error; Low level white noise source is introduced to make the profile of the quantization 'noise' more irregular; Useful when reducing the number of bits per word in a signal (i.e. when converting






41. Method of sampling data at a higher resolution (higher sample rate) as a means of reducing harmonic content during D/A conversion; (x2) oversampling gets rid of all odd harmonic content






42. CobraNet; EtherSound; Dante; AVB (currently under development)






43. The number of compressions or rarefactions in one second; The higher the frequency the more compressions & rarefactions per second; Measured in Hertz






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






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






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






47. 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






48. Difference in brightness between land and pit on a CD Physical Format






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






50. A time regulator that makes all samples and bits to align when working with interconnected digital devices; Basically a signal that all of the digital devices refer to when operating.