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






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






3. How Loud (Y-Axis) & How Fast (X-Axis)






4. 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






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






6. The process of reducing the space required to store data by efficiently encoding the content.






7. Root Mean Square; Refers to taking the square root of all instantaneous amplitudes; Takes the average of those squares; (-6 Peak Level is approximately equal to -20 RMS)






8. Have odd numbered harmonics






9. 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






10. Same as 'aliasing'






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






12. Series of dots and dashes representing the letters of the alphabet; Most common letters are represented by the shortest dots and dashes; Example of entropy coding






13. 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






14. Lossless Format; Can hold up to 25GB on a single- layer disc and 50GB on a dual- layer disc






15. The amount of energy at each wavelength






16. Mixing data and control characters in a single operation






17. 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






18. 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






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






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






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






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






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






24. 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.






25. 12.33 GB; DS/ML






26. Reduces the percieved distortion due to quantization error; Low level white noise source is introduced to make the profile of the quantization 'noise' more irregular; Useful when reducing the number of bits per word in a signal (i.e. when converting






27. A frequency specified for a filter (digital or electronic) the marks the point at which the frequency content of a signal is altered +/- 3dB






28. Discrete incremental distinctions made between the value of one sample and the next; Breaks down bit depth into a series of evenly spaced intervals






29. Measure of the amplitude of a longitudinal wave






30. 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






31. 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);






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






33. 8- in/8- out on one cable; 25- pin D- sub connector






34. 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






35. 15.9 GB; DS/DL






36. The frequency range that is allowed through a filter






37. Describes various optical disc characteristics including the size and shape of the disc - the size of pits - the speed at which the disc spins - and a multitude of aspects regarding the specifications of the player itself






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






39. The more bits allocated during quantization - the more accurate the measurement






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






41. 16-Bit; 44.1 kHz; PCM; Stereo






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






43. 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






44. A digital filter's time domain output sequence when the input is a single sample is input






45. ADAT Optcal; 8- in/8- out on two cables; Fiber- optic - TOSLINK connector






46. The difference between the analog value and the approximated digital value due to the 'rounding' that occurs while converting the analog signal to digital






47. 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






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






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






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