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






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






3. 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






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






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






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






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






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






10. Number of bits used to represent the smallest unit of information in an audio file; Greater bit depth = better quality audio






11. Electromagnetic receptor that detects the radiation known as visible light






12. 8.75 GB; DS/SL






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






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






15. 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






16. 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






17. 8- in/8- out on one cable; 25- pin D- sub connector






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






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






20. Eight channel digital surround sound system by Dolby






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






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






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






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






25. 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






26. Channels are processed one at a time and the results are stored on multiple CPU buffers that alternately send data as DAW requests the data for playback; First few seconds are relayed to the buffer prior to processing and playback and it continuously






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






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






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






30. 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)






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






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






33. The frequency range that is allowed through a filter






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






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






36. 'Capturing' part of digital audio; Never captures a signal perfectly






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






38. Waveform of a pure tone showing simple harmonic motion






39. 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






40. Data transmission protocol over which computer network traffic travels; Poorly suited to real- time transmission but numerous attempts have been made to harness the technology because of its flexible routing; Uses simple Cat5 cabling; Resists interfe






41. Signal that uses variable voltage to create continuous waves resulting in an inexact transmission






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






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






44. 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






45. 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






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






47. 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






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






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






50. 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)