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. 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






2. Single- pin RCA cable or fiber- optic TOSLINK connector used for digital transfer; 75O coaxial - 2- channel unbalanced; 'Consumer' format of AES3






3. The loudest point of a Full Scale system






4. 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






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






6. 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






7. 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






8. 'Reconstructing' part of digital audio






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






11. Based on psychoacoustics - these are the basis of frequency analysis for a perceptual codec;






12. Root Mean Square; Refers to taking the square root of all instantaneous amplitudes; Takes the average of those squares; (-6 Peak Level is approximately equal to -20 RMS)






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






14. 7.95 GB; SS/DL






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






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






17. 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






18. 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






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






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






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






22. 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






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






24. 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)






25. 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






26. 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






27. 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






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






29. Most significant lossless coding technique in current use; Measure of disorder in which long strings of data are represented by short symbols and uses the shortest symbols to represent the most common repetitive audio data maximizing data reduction






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






31. Stores only one copy of a stereo signal and assigns it to both channels in order to save 50% of original bandwidth






32. EBU






33. French mathematician that noted that any complex sound can be broken down into a series of component pure tones






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






35. Multi-Bit Words; (Pulse Code Modulation)






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






37. Roughly around 1 -130 ft/s






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






39. Same as 'aliasing'






40. A situation where a calculated value cannot fit into the number of digits reserved for it






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






42. Occurs as data is assembled into meaningful bits or information and as left & right channels are separated






43. 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






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






45. Discrete incremental distinctions made between the value of one sample and the next; Breaks down bit depth into a series of evenly spaced intervals






46. 8- in/8- out on one cable; 25- pin D- sub connector






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






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






49. 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






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