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Digital Audio

Subject : engineering
Instructions:
  • Answer 50 questions in 15 minutes.
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  • 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 digital filter's time domain output sequence when the input is a single sample is input






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






3. Eight channel digital surround sound system by Dolby






4. Playback; I/O Connections; CPU (Streaming); Conversion from DAW or Software






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






6. The loudest point of a Full Scale system






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






8. 'Reconstructing' part of digital audio






9. A drive that can read and write on optical media that hold up to 50 GB on two layers; 24- bit/96 kHz for 8-Channel; 24- bit/192 kHz for 6-Channel






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






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






12. Fractional part of a floating- point number; Also called the mantissa; Defines precision






13. 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






14. 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






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. 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






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






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






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






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






21. Ultra low- latency - 512- channel (on a gigabit network) - less flexible AoE format; Routed like audio cables...not network cables






22. 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






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






24. Describes acceptable data - performances both offered and essential for a disc player - and the complete user experience






25. 16-Bit; 44.1 kHz; PCM; Stereo






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. ADPCM; Pulse code modulation that produces a digital signal with a lower bit rate than standard PCM; Records only the difference between samples






28. The amount of energy at each wavelength






29. 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






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






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






32. Increases or decreases the digital signal so that the loudest sample is brought up to 0dBfs; Uses all bits from dynamic range and makes it even from track to track






33. Subtract place values from the decimal number and place ones or zeros in the correct places






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






35. 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)






36. 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






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






38. 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






39. 15.9 GB; DS/DL






40. 4.38 GB; SS/SL






41. 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






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






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






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






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






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






47. 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






48. 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






49. A method of representing real numbers using a mantissa and an exponent






50. 12.33 GB; DS/ML