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The simplest way to digitally encode audio signals is [[pulse-code modulation]] (PCM), which is used on [[audio CDs]], [[Digital Audio Tape|DAT]] recordings, and so on. Digitization transforms continuous signals into discrete ones by sampling a signal's amplitude at uniform intervals and [[rounding]] to the nearest value representable with the available [[Audio bit depth|number of bits]]. This process is fundamentally inexact, and involves two errors: ''[[discretization error]],'' from sampling at intervals, and ''[[quantization error]],'' from rounding.
The more bits used to represent each sample, the finer the granularity in the digital representation, and thus the smaller the quantization error. Such ''quantization errors'' may be thought of as a type of noise, because they are effectively the difference between the original source and its binary representation. With PCM, the audible effects of these errors can be mitigated with [[dither]] and by using enough bits to ensure that the noise is low enough to be masked either by the signal itself or by other sources of noise. A high quality signal is possible, but at the cost of a high [[bitrate]] (e.g., over 700 [[kbit/s]] for one channel of CD audio). In effect, many bits are wasted in encoding masked portions of the signal because PCM makes no assumptions about how the human ear hears.
Sub-band coding is used for example in the [[G.722]] codec. It uses sub-band adaptive differential pulse code modulation (SB-[[ADPCM]]) within a bit rate of 64 kbit/s. In the SB-ADPCM technique, the frequency band is split into two sub-bands (higher and lower) and the signals in each sub-band are encoded using ADPCM.
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