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. As sample rate is increased more room is created for a smoother slope of the attenuation band because Nyquist limit extends well beyond range of hearing with each increase






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






3. AES






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. More aggressive lossy data reduction techniques that require further manipulation of the stereo field; Examples are 'Intensity' & 'M-S'






6. 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






7. Twice as many samples as the highest frequency at minimum; Sampling rate totally controls frequency response






8. Rate at which energy is drawn from a source that produces a flow of electricity in a circuit; Expressed in volts






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






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






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






12. Digital and analog processing capability is combined on a single microchip allowing for 1- bit resolution at high sample rates






13. Measure of the amplitude of a longitudinal wave






14. 8.75 GB; DS/SL






15. MPEG; Standardizing body of audio coding






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






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






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






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






20. Snippets of time in which frequency analysis takes place in a perceptual codec






21. The amount of energy at each wavelength






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






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






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






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






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






27. 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






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






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






30. 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






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






32. The act of a frequency swinging back and forth with a steady - uninterrupted rhythm






33. 15.9 GB; DS/DL






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






35. The elapsed time it takes for a packet of data to arrive at its destination; Lagging or pause of an audio signal as digital processing occurs; Can be managed utilizing several forms of 'audio monitoring'






36. 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






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






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






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






40. 1st commercially successful AoE format for the transmission of digital audio - video - and control signals over 64- channel 100Mbps Ethernet networks






41. 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






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






43. Sample Rate x Bit Depth x # of Channels






44. 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);






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. 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






47. Have odd numbered harmonics






48. 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






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






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