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






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






3. 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






4. Playback; I/O Connections; CPU (Streaming); Conversion from DAW or Software






5. 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






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






7. Mixing data and control characters in a single operation






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






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






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






11. 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'






12. 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.






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






14. Apple Lossless; Windows Media Lossless; DTS HD Master Audio; Dolby True HD; FLAC






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






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






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






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






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






20. 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






21. 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






22. 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






23. 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






24. 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






25. 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






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






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






28. Found that aliasing was always a problem no matter how fast you sample; Less data recorded but more accurate; 2 samples per wave length.






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






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






31. Splits the input signal and mixes it with an analog copy so that no latency is present






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






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






34. The ability of a digital system to perform complex DSP without running into problems with overflow or loss of resolution






35. Roughly around 1 -130 ft/s






36. If a signal is sampled at a rate higher than twice the highest significant signal frequency and at evenly spaced intervals - then the samples contain all the information of the original signal






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






38. Digital Word -> Series of Resistors (each with assigned charges) -> Sample- and-Hold Circuit -> Anti-Imaging Filter (Smoothing Filter) -> Reconstructed Sample






39. 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






40. 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






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






42. 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






43. Waveform of a pure tone showing simple harmonic motion






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






45. 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






46. 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






47. Very selective method of lowering buffer levels by halting different levels of audio processing






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






49. Measure of the amplitude of a longitudinal wave






50. Improper (false) sampling of high frequencies that cause much lower frequencies to be produced when the audio is reconstructed; Foldover happens at higher frequencies as harmonics reach levels that exceed the Nyquist limit.







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