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. The frequency range that is allowed through a filter






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






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






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






5. 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






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






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






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






9. 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






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






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






12. 'Reconstructing' part of digital audio






13. 8.75 GB; DS/SL






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






15. Serial Copy Management System; main difference between AES3 & S/PIDF






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






17. 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






18. 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






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






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






21. 1.) Bit Rate x Sample Rate (you'll get b/sec) 2.) Multiply by 60 if converting seconds to minutes 3.) Divide by 8 to convert bits to Bytes and get B/min 4.) Divide by 1 -024 to get KB/min and keep doing it until you get desired bit rate specification






22. Defines the number of sample per second taken from a continuous signal to make a discrete signal; Governs the frequency response of digital audio






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






24. Built into DAWs; Bits are added when signals are mixed together to avoid clipping






25. 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






26. Low Pressure; Part in a longitudinal wave where the particles are spread apart






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






28. 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






29. 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






30. 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






31. 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






32. 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






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






34. 'Capturing' part of digital audio; Never captures a signal perfectly






35. Same as 'aliasing'






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






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






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






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






40. Eliminates frequencies above the Nyquist limit from becoming samples; Occurs prior to quantization






41. 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






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






43. 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






44. 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






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






46. 12cm plastic disc; 1.2mm thick; One- sided; Red Laser; 1.6 microns between tracks; 125 nanometer pits






47. The art of deciding where to place a processor in signal flow based on how that processor will be influenced by the other processors in the path






48. Subtract place values from the decimal number and place ones or zeros in the correct places






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






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