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






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






3. 4.38 GB; SS/SL






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






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






6. 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






7. 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






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






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






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






11. 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






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






13. Sum of all harmonics; Sum of sine and cosine waves which have frequencies f - 2f - 3f - 4f...






14. 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.






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






16. Roughly around 1 -130 ft/s






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






18. Father of modern information theory; Solidified the Nyquist Theory by adding the concept that bits per second (binary representation of audio signals) must be at equal intervals to accurately represent data






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






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






21. 12.33 GB; DS/ML






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






23. 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






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






25. 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'






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






27. 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






28. Removes high frequency images and noise and smoothes the stair case output coming from of the sample and hold circuit; Also called a SMOOTHING FILTER






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






30. 15.9 GB; DS/DL






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






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






33. Mixing data and control characters in a single operation






34. 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






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






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






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






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






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. Decibels Full Scale






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






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






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






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






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






46. 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






47. 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






48. 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)






49. 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






50. 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