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






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






3. In order to play multiple channels at one time data is buffered as read to disk; Latency happens between play command & beginning of playback






4. 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






5. 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);






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






7. 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






8. EDL; Final list of samples used in the audio editing process; Identified by time code






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






10. 12cm plastic disc; 1.2mm thick; One- sided; Red Laser; 1.6 microns between tracks; 125 nanometer pits






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






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






13. 15.9 GB; DS/DL






14. Same as 'aliasing'






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






16. 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






17. 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






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






19. 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






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






21. 12.33 GB; DS/ML






22. (Amplitude Based) Amplitude: Voltage; Quantization; Bit Depth; Quantization Intervals; Quantization Noise; [Signal:Quantization Noise Ratio]; Dither; Dynamic Range






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






24. Data reduction technique that selectively removes original information in order to significantly reduce the file size; Some data is lost; Files can be reduced up to 99% in size (90% with no perceived sound quality loss); Bit rate effects the perceive






25. 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






26. The frequency range that is allowed through a filter






27. Eight channel digital surround sound system by Dolby






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






29. 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






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






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






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






33. Industry Standards: -6 dB Peak = -20 RMS Meter






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






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






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






39. 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






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






41. Deviation from a normal - steady pulse or tick of a clock that contributes to misrepresentation of a signal; Result of small timing irregularities that become magnified during the transmission of digital signals as the signals are passed from one dev






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






43. Have odd numbered harmonics






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






45. Number of bits per second processed when sampling sound; (Sampling Rate x Bit Depth) = Resolution






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






47. 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






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






49. Psychoacoustic model of data reduction used for general audio compression that aims to transmit only features perceptible to the human ear; Useful for mastering streaming media






50. Decibels Full Scale