Digital audio

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Digital audio refers to audio signals stored in a digital format. Specifically, the term encompasses the following:

  1. Audio conversion:
    1. Analogue to digital conversion (ADC) - the capture and digitisation of an analogue audio signal, a process known as sampling.
    2. Digital to analog conversion (DAC) - the conversion of digital audio to a line signal for playback or distribution.
  2. Audio signal processing - processing the digital signal in some way, such as to apply equalisation, reverberation, or to perform sample rate conversion.
  3. Storage, retrieval, and transmission of digital information in an audio format such as CD, MP3, Ogg Vorbis, etc.

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The digital paradigm

Digital technology has emerged for the simple reason that analogue signals cannot be copied or transmitted perfectly, while digital signals can be. With analog technology, information resolution (ie. 'quality') is lost with each generation of reproduction.

Digital audio was implemented as an answer to a desire to use media multiple ways. The preservation of quality is important in a professional environment where signals could possibly be used through cable, mixing desks and processing equipment.

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Overview of digital audio

Sound inherently begins as an analogue signal and in order for the benefits of digital audio to be realised, the conversion process must be of sufficiently high quality in order to be worthwhile.

In an audio context, "sufficiently high quality" means that the reproduced digital signal should sound identical to the original analogue signal. In other words, the limits of the human auditory system govern the technical requirements of the conversion process.

The generally accepted frequency response of human hearing is from 20 Hz - 20 kHz. According to Nyquist, the maximum bandwidth that can be represented by a digital signal less is half that of the sample rate. This leads to a required sample rate of at least 40 kHz. In practise, a slightly higher sample rate is needed to allow for a practical anti-aliasing filter.

In the early days of digital audio, the only practical storage device with sufficient bandwidth and storage space was a video recorder and these were adapted to store the digital signal, usually by interfacing said video recorder to a PCM adaptor. Some simple maths shows that it is possible to use either 525/60 NTSC or 625/50 PAL video with a sampling rate of 44.1 kHz, a sample rate which persisted with the introduction of CD.

16-bit digital audio was adopted as the broadcast standard because it offers 96 decibels (dB) of dynamic range, enough to match the quality of broadcast analogue. Modern systems do not suffer from the earlier constraints of bandwidth and storage space; 96 kHz and 192 kHz sample rates and 24-bit samples are now common.

Pulse-code Modulation (PCM) is by far the most common way of representing a digital signal. It is simple, easy to reconstruct and is not compressed. A PCM representation of an analogue signal is generated by measuring (sampling) the instantaneous amplitude of the analogue signal, and then quantising the result to the nearest bit.

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Digital audio technologies

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Digital audio interfaces

Audio signals can also be carried losslessly over general-purpose buses such as USB or FireWire.

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References

  • Borwick, John, ed., 1994: Sound Recording Practice (Oxford: Oxford University Press)
  • Ifeachor, Emmanuel C., and Jervis, Barrie W., 2002: Digital Signal Processing: A Practical Approach (Harlow, England: Pearson Education Limited)
  • Rabiner, Lawrence R., and Gold, Bernard, 1975: Theory and Application of Digital Signal Processing (Englewood Cliffs, New Jersey: Prentice-Hall, Inc.)
  • Watkinson, John, 1994: The Art of Digital Audio (Oxford: Focal Press)
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See also

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External links

it:Audio digitale

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