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. AES






2. 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






3. Sample Rate x Bit Depth x # of Channels






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






5. 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






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






7. Measure of the amplitude of a longitudinal wave






8. 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






9. 15.9 GB; DS/DL






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






11. 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






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






13. Roughly around 1 -130 ft/s






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






15. 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






16. Waveform of a pure tone showing simple harmonic motion






17. 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






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






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






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






21. 12cm plastic disc; 1.2mm thick; One- sided; Red Laser; 1.6 microns between tracks; 125 nanometer pits






22. 12.33 GB; DS/ML






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






24. Amplitude meter that takes the square root of all instantaneous amplitudes and averages them to find a mean and squares that value; Useful with particularly complex waveforms






25. 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






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






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






28. 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






29. 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






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






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






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






33. 'Reconstructing' part of digital audio






34. 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






35. 16-Bit; 44.1 kHz; PCM; Stereo






36. Circuit that seizes voltage values with each tick of an A/D device's internal clock






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






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






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






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






41. 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)






42. 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






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






44. 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






45. 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.






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






47. 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






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






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






50. Sony and Philips optical disc format; Utilizes sigma delta DSD to offer higher resolution; 1- bit; 2.8224 MHz; 6-Channel