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. More accuracy in low amplitudes and less in higher amplitudes






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






3. Specific set of instructions for carrying out a data reduction technique that determines how to 'save' binary data information efficiently






4. Used when the reference pressure of a sound is 20 microPa (0.00002); Sound Pressure Level; Measure of amplitude






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






6. Take up half as many bits (50%); Algorithm can keep same quality by lossing the stereo track






7. 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






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






9. 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);






10. Digital and analog processing capability is combined on a single microchip allowing for 1- bit resolution at high sample rates






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






12. Roughly around 1 -130 ft/s






13. 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






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






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






16. 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






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






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






19. 4.38 GB; SS/SL






20. The loudest point of a Full Scale system






21. 1.) Taking a series of evenly- spaced measurements 2.) Signal contains no frequency components higher than half the sample rate






22. 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






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






24. Very quiet digital amplifier that produces a series of output pulses with the audio signal coded the same as the width of the output pulses; Pulses are used to represent wave forms and are either on or off; Intense signals have long pulses with short






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






26. Number or variable that represents the number of times the base of a power is used as a factor; Defines magnitude






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






28. The amount of energy at each wavelength






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






30. 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






31. 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






32. 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






33. AAC (Advanced Audio Coding); MP3; RA; WMA; OGG Vorbis; Dolby Digital/AC-3; DTS; ADPCM






34. 7.95 GB; SS/DL






35. The continuous loss of signal strengths as a signal travels through a medium






36. Data reduction technique that selectively removes original information in order to significantly reduce the file size; Some data is lost; Files can be reduced up to 99% in size (90% with no perceived sound quality loss); Bit rate effects the perceive






37. 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






38. Same as 'aliasing'






39. 15.9 GB; DS/DL






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






41. Measuring equipment in A/D conversion that processes voltage and provides a value for that voltage






42. Contains all even and odd harmonics associated with a fundamental tone - making it a rich source for modeling other sounds; Amplitude of each overtone decreases exponentially as a ratio of the harmonic's frequency to that of the fundamental






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






44. The act of a frequency swinging back and forth with a steady - uninterrupted rhythm






45. 'Reconstructing' part of digital audio






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






47. The frequency range that is allowed through a filter






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






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. 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







Sorry!:) No result found.

Can you answer 50 questions in 15 minutes?


Let me suggest you:

Browse all subjects

Browse all tests

Most popular tests



Major Subjects

Tests & Exams


AP
CLEP
DSST
GRE
SAT
GMAT

Browse all subjects

Browse all tests

Most popular tests