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. 12.33 GB; DS/ML






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






3. 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






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






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






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






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






8. Decibels Full Scale






9. 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






10. 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






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






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






13. More accuracy in low amplitudes and less in higher amplitudes






14. 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






15. The art of deciding where to place a processor in signal flow based on how that processor will be influenced by the other processors in the path






16. Unit of measurement that is equal to one billionth of a meter






17. 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






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






19. 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.






20. 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)






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






22. Each bit in the bit depth is equal to a _____ increase in dynamic range






23. 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






24. 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






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






26. 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);






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






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






29. 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






30. Eliminates frequencies above the Nyquist limit from becoming samples; Occurs prior to quantization






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






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






33. Method of representing an acoustic quantity with a series of binary numbers; Can have only specific individually distinct values






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






35. 15.9 GB; DS/DL






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






37. AES






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






39. 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






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






41. 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






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






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






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






45. Allowance of noise floor below that which is required for the final product






46. (AES/EBU); 110O - 2- channel balanced digital audio cable with an XLR connection; NOT a mic cable!!






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






48. 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






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






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