A way to reduce the bandwidth needed by digitized voice channels by half (to 32 Kb from 64 Kb).
A method of speech encoding that calculates the difference between two consecutive coded speech samples, allowing voice signals to be encoded in half the space that standard PCM requires.
Differential pulse code modulation that also uses adaptive quantizing; an audio coding algorithm providing a modest degree of compression together with good quality.
A standard that codes for digitized speech. It samples the sound waves 8,000 times a second; the sample representing the difference between two adjacent samples.
A technique for compressing audio data by storing the difference between signals rather than the actual signal. Usually abbreviated as ADPCM.
An audio-compression technique.
A ITU-TS standard technique for voice encoding and compression. It allows an analog to be carried within a 32Kbit/s digital channel.
A form for the digital coding voice signals, typically at 32 Kbps.
Encoding technique (ITU-T) that allows analog voice signals to be carried on a 32K bps digital channel. Sampling is done at 8 kHz with 3 or 4 bits used to describe the difference between adjacent samples.
A method of sampling and converting analog signals to digital signals. Similar to DPCM except that when a wide difference occurs between two successive samples of a signal, it uses a sophisticated algorithm to code the difference.
A PCM encoding technique that uses only 4 bits to encode an analog sample. By adapting the quantizing range to the difference between the two samples, it achieves the same voice quality as 8-bit PCM, but with 32 Kbps instead of 64 Kbps.