Adaptive DPCM (ADPCM) is a variant of DPCM which in term is a form of PCM, DPCM varies the size of the quantization step, to allow further reduction of the required bandwidth for a given signal-to-noise ratio (SNR).
Delta modulation (differential PCM), another variant, uses one bit per sample.
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-lawPCM (Europe and most of the rest of the world). These are logarithmic compression systems where a 12 or 13 bit linear PCM sample number is mapped into an 8 bit value. This system is described by international standard G.711. An alternative proposal for a floating point representation, with 5 bit mantissa and 3 bit radix, was abandoned.
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 u-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.
Later it was found that even further compression was possible and additional standards were published. Some of these international standards describe systems and ideas which are covered by privately owned patents and thus use of these standards requires payments to the patent holders.
Some ADPCM techniques are used in Voice over IP communications.