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






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






3. Measure of the amplitude of a longitudinal wave






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






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






6. DAW's and software must buffer when converting or bouncing and this latency can add up if not monitored






7. Sample Rate x Bit Depth x # of Channels






8. 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);






9. 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






10. Visual graph that shows how loud a sound is at different frequencies






11. 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






12. Measures the highest levels of a signal being recorded or mixed; Monitors for clipping - which occurs at 0dBFS); Does not always reflect perceived volume of signal






13. Uses entropy coding as the basis; Computer data compression algorithm that packages files such as .ZIP & .RAR






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






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






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






17. 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.






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






19. Same as 'aliasing'






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






21. 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






22. Most significant lossless coding technique in current use; Measure of disorder in which long strings of data are represented by short symbols and uses the shortest symbols to represent the most common repetitive audio data maximizing data reduction






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






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






25. 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






26. 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






27. 15.9 GB; DS/DL






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






29. Level above which audible sounds are painful (125 - 130 db)






30. EBU






31. 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






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






33. 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)






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






35. Reference voltage determined by summing the voltage values of a predetermined number of previous samples; Numbers of 1 in row = waveform peak; Numbers of 0 in row = waveform trough; Basis of Sony's Direct Stream Digital (DSD)






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






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






38. 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






39. Method of representing an acoustic quantity with a series of binary numbers; Can have only specific individually distinct values






40. A drive that can read and write on optical media that hold up to 50 GB on two layers; 24- bit/96 kHz for 8-Channel; 24- bit/192 kHz for 6-Channel






41. Data is transmitted over fiber optic lines; Uses a TOSLINK connecter instead of an RCA type; Can transmit multi- channel audio; Not susceptible to ground hum and loops; Able to support far higher rates of data transfer over greater distances than coa






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






43. 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






44. Discrete incremental distinctions made between the value of one sample and the next; Breaks down bit depth into a series of evenly spaced intervals






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






46. High Pressure - Part of a longitudinal wave where the particles of the medium are close together






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






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






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






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