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Digital Audio

Subject : engineering
Instructions:
  • Answer 50 questions in 15 minutes.
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  • Match each statement with the correct term.
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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. Sony and Philips optical disc format; Utilizes sigma delta DSD to offer higher resolution; 1- bit; 2.8224 MHz; 6-Channel






2. The loudest point of a Full Scale system






3. Ultra low- latency - 512- channel (on a gigabit network) - less flexible AoE format; Routed like audio cables...not network cables






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






5. 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






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






7. Signal voltage is relayed to a register from sample- and - hold circuit; Holds reference frequencies in binary form that decrease in value; Finds approximated value & assigns binary number accordingly






8. Represents the amplitude component of the digital sampling process; Technique of incrementing a continuous analog event into a discrete set of binary digits (bits)






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






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






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






12. 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






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






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






15. Full Scale; Type of metering that measures level in digital recording system; Recording and Mixing levels should NEVER exceed 0dB FS in digital audio or clipping will occur






16. 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






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






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






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






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






21. A network communications protocol that specifies how machines will exchange data; Uses a broadcast system in which one machine transmits its message on the communication medium and the other machines listen for messages directed to them






22. 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






23. 15.9 GB; DS/DL






24. 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)






25. 1st commercially successful AoE format for the transmission of digital audio - video - and control signals over 64- channel 100Mbps Ethernet networks






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






27. 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






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






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






30. Waveform of a pure tone showing simple harmonic motion






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






32. Also known as equal loudness curves; Graph that indicates the average ear sensitivity to different frequencies at different SPL levels (as volume increases - these curves flatten out)






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






34. A frequency specified for a filter (digital or electronic) the marks the point at which the frequency content of a signal is altered +/- 3dB






35. 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






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. High Pressure - Part of a longitudinal wave where the particles of the medium are close together






38. 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






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






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






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






42. Branch of psychology concerned with the subjective perception of sound






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






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






45. 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






46. 8.75 GB; DS/SL






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






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






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






50. 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







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