Sub-band coding: Difference between revisions

Revision as of 00:53, 13 March 2007

Revision as of 05:56, 15 March 2007

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==Encoding audio signals==

The simplest way to encode audio signals is [[Pulse-code modulation]] (PCM), which is used on music[[audio CDs]], [[Digital Audio Tape|DAT]] recordings, and so on. Like all digitization, PCM adds noise to the signal, which is generally undesirable. The fewer bits used in digitization, the more noise gets added. The way to keep this noise from being a problem is to use enough bits to ensure that the noise is always low enough to be masked either by the signal or by other sources of noise. This produces a high quality signal, but at a high bit rate[[bitrate]] (over 700k bps700[[Kbps]] for one channel of CD audio). A lot of those bits are encoding masked portions of the signal, and are being wasted.

There are more clever ways of digitizing an audio signal, which can save some of that wasted bandwidth. A classic method is nonlinear PCM, such as [[mu-law]] encoding (named after a perceptual curve in auditory perception research). This is like PCM on a logarithmic scale, and the effect is to add noise that is proportional to the signal strength. Sun's [[Au file format]] for sound is a popular example of mu-law encoding. Using 8-bit mu-law encoding would cut our one channel of CD audio down to about 350k bps, which is better but still pretty high, and is often audibly poorer quality than the original (this scheme doesn't really model masking effects).