In
telephony, a standard audio signal for a single phone call is encoded as 8000 analog samples per second, of 8 bits each, giving a 64 kbit/s digital signal known as
DS0. The default
signal compression encoding on a DS0 is either
μ-law (mu-law) PCM (North America and Japan) or
A-law PCM (Europe and most of the rest of the world). These are logarithmic compression systems where a 13- or 14-bit linear PCM sample number is mapped into an 8-bit value. This system is described by international standard
G.711. Where circuit costs are high and loss of voice quality is acceptable, it sometimes makes sense to compress the voice signal even further. An ADPCM algorithm is used to map a series of 8-bit μ-law (or a-law) PCM samples into a series of 4-bit ADPCM samples. In this way, the capacity of the line is doubled. The technique is detailed in the
G.726 standard. ADPCM techniques are used in
voice over IP communications. In the early 1990s, ADPCM was also used by
Interactive Multimedia Association to develop the legacy audio codecs ADPCM DVI, IMA ADPCM, and DVI4. ==Split-band or subband ADPCM==