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5_Analogue recording vs digital.doc
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Digital fundamentals

A digital recorder firstly requires the input of an analog signal; this signal may come directly from a microphone pre-amp, but any analog audio signal can be converted. Measurements of the signal intensity are then made at regular intervals (sampling) by the analog-to-digital converter. At each sampling point, the signal must be assigned a specific intensity from a set range of values (quantization). For doing this, the original sound wave can now be described using only numbers—as digital information. Each sample can be given an ordinal number which signifies the point in relative time that it represents, and the magnitude of the sample is an analog of pressure at the microphone (Watkinson 1994) (or, for an artificial sound signal, the pressure that would be at the microphone to correspond to that sound.) When the original signal is converted into numbers (usually binary numbers, 1's and 0's, called 'bits') further additions of noise and distortion, provided they are not great enough to cause digital errors, can be rejected at every stage of processing; this is what is referred to as the regenerative nature of digital signals. Digital errors, called bit errors in binary digital systems, are events of noise and/or distortion which cause one number (or bit) to appear more like another number than like the number they started out as. As long as a digital symbol appears closer to being what it began as than to anything else, it can be regenerated. When raw digital errors cannot be avoided, error correction coding can allow some of them to be detected and fixed. Error correction, essential when transferring digital audio over noisy channels, helps to eliminate bit errors by comparing extra data against the main data to detect limited numbers of digital errors, figure out which digital symbols (numbers) were changed, and change them back. When playing back a digital recording, the digital information is converted back into a continuous, analog signal by a digital-to-analog converter. This electronic signal is then amplified and converted back into a sound wave by a loudspeaker (just as would be done with the analog signal produced by an analog machine from an analog recording).

Noise performance

For electronic audio signals, sources of noise include (unavoidable) mechanical, electrical and thermal noise in the recording and playback cycle (from mechanical transducers (microphones, loudspeakers), amplifiers, recording equipment, the mastering process, reproduction equipment, etc). Whether an audio signal is, at some stage, converted into a digital form will affect how much effective noise is added, due to the partial immunity to noise that the digital regenerative effect provides. The actual process of digital conversion will always add some noise, however small in intensity; the bulk of this in a high-quality system is quantization noise, which cannot be theoretically avoided, but some will also be electrical, thermal, etc. noise from the analog-to-digital converted device.

The amount of noise that a piece of audio equipment adds to the original signal can be quantified. Mathematically, this can be expressed by means of the signal to noise ratio (SNR). Sometimes the maximum possible dynamic range of the system is quoted instead. In a digital system, the number of quantization levels, in binary systems determined by and typically stated in terms of the number of bits, will have a bearing on the level of noise and distortion added to that signal. The 16-bit digital system of Red Book audio CD has 216= 65,536 possible signal amplitudes, theoretically allowing for an SNR of 98 dB (Sony Europe 2001) and dynamic range of 96 dB.

In order to meet the theoretical maximum performance of a 16 bit digital system, for a 0.5 V peak to peak input line signal, a PCM (pulse code modulation) quantizer would require an equivalent minimum input sensitivity of just 7.629 microvolts. For an analog recorder, this is equivalent to a 15.3 ppm sensitivity for the whole recording system and medium.[citation needed] With digital systems, the quality of reproduction depends on the analog-to-digital and digital-to-analog conversion steps, and does not depend on the quality of the recording medium, provided it is adequate to retain the digital values without an excessive error rate (exceeding the capacity of any error correction mechanisms used in the system).

Typically anything below 14 bits can lead to perceptibly reduced sound quality, with 80 dB of SNR considered as an informal "minimum" for Hi-Fi audio. However, it is uncommon to find digital media specified for less than 14 bits, except for older 12-bit PCM Camcorder audio (or DAT in long-play, 32 kHz mode) and the output from older or lower-cost computer software, sound cards/circuitry, consoles and games (typically 8 bit as a minimum and standard, though trick sample output methods for generally non-PCM hardware [e.g. FM synthesis cards including the Adlib card] gave SNR performances closer to that of an ideal "6" or "4" bit PCM digital converter).

Each additional quantization bit theoretically adds 6 dB in possible dynamic range, e.g. 24 x 6 = 144 dB for 24 bit quantization, 126 dB for 21-bit, and 120 dB for 20-bit.