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. Digital (binary) measurements of how long each pulse is either on or off; Width of increasing voltage or decreasing voltage is assigned a 1 or 0 respectively






2. 7.95 GB; SS/DL






3. 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






4. Single- pin RCA cable or fiber- optic TOSLINK connector used for digital transfer; 75O coaxial - 2- channel unbalanced; 'Consumer' format of AES3






5. 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'






6. Eight channel digital surround sound system by Dolby






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






8. 15.9 GB; DS/DL






9. 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






10. Sony and Philips optical disc format; Utilizes sigma delta DSD to offer higher resolution; 1- bit; 2.8224 MHz; 6-Channel






11. Snippets of time in which frequency analysis takes place in a perceptual codec






12. Each bit in the bit depth is equal to a _____ increase in dynamic range






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






14. Ratio of magnitude of the analytical signal to the magnitude of the background noise signal






15. Joint-Stereo Technique; Since the human brain is unable to localize sounds at high frequencies well sounds above 9 kHz threshold are encoded in mono






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






17. 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






18. 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






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






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






21. 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






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






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






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






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






26. 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






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






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






29. Sample Rate x Bit Depth x # of Channels






30. The mathematics - algorithms - and the techniques used to manipulate signals after they have been converted to digital form






31. 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);






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






33. Number of bits per second processed when sampling sound; (Sampling Rate x Bit Depth) = Resolution






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






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






36. Data reduction technique that does not effect quality of original audio; No effect on original quality; Typically around 50% reduction; Exact reconstruction of digital code for the audio signal






37. The process of reducing the space required to store data by efficiently encoding the content.






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






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






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






41. 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






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






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






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






45. 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






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






47. Data transmission protocol over which computer network traffic travels; Poorly suited to real- time transmission but numerous attempts have been made to harness the technology because of its flexible routing; Uses simple Cat5 cabling; Resists interfe






48. 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






49. 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






50. 12.33 GB; DS/ML