Adaptive DPCM
Encyclopedia
Adaptive differential pulse-code modulation (ADPCM) is a variant of differential pulse-code modulation (DPCM) that varies the size of the quantization step, to allow further reduction of the required bandwidth for a given signal-to-noise ratio
.
Typically, the adaptation to signal statistics in ADPCM consists simply of an adaptive scale factor before quantizing the difference in the DPCM encoder.
ADPCM was developed in the early 1970s at Bell Labs
for voice coding, by P. Cummiskey, N. S. Jayant, and James L. Flanagan.
, 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.
Some ADPCM techniques are used in Voice over IP
communications. ADPCM was also used by Interactive Multimedia Association
for development of legacy audio codec known as ADPCM DVI, IMA ADPCM or DVI4, in the early 1990s.
is a ITU-T
standard wideband speech codec
operating at 48, 56 and 64 kbit/s, based on subband coding with two channels and ADPCM coding of each. Before the digitization process, it catches the analog signal and divides it in frequency bands with QMF
filters (quadrature mirror filters) to get two subbands of the signal. When the ADPCM bitstream of each subband is obtained, the results are multiplexed and the next step is storage or transmission of the data. The decoder has to perform the reverse process, that is, demultiplex and decode each subband of the bitstream and recombine them.
Referring to the coding process, in some applications as voice coding, the subband that includes the voice is coded with more bits than the others. It is a way to reduce the file size.
Signal-to-noise ratio
Signal-to-noise ratio is a measure used in science and engineering that compares the level of a desired signal to the level of background noise. It is defined as the ratio of signal power to the noise power. A ratio higher than 1:1 indicates more signal than noise...
.
Typically, the adaptation to signal statistics in ADPCM consists simply of an adaptive scale factor before quantizing the difference in the DPCM encoder.
ADPCM was developed in the early 1970s at Bell Labs
Bell Labs
Bell Laboratories is the research and development subsidiary of the French-owned Alcatel-Lucent and previously of the American Telephone & Telegraph Company , half-owned through its Western Electric manufacturing subsidiary.Bell Laboratories operates its...
for voice coding, by P. Cummiskey, N. S. Jayant, and James L. Flanagan.
In telephony
In telephonyTelephony
In telecommunications, telephony encompasses the general use of equipment to provide communication over distances, specifically by connecting telephones to each other....
, 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
Signal compression
In telecommunication, the term signal compression has the following meanings:In analog systems, reduction of the dynamic range of a signal by controlling it as a function of the inverse relationship of its instantaneous value relative to a specified reference level.Signal compression is usually...
encoding on a DS0 is either μ-law (mu-law)
Mu-law algorithm
The µ-law algorithm is a companding algorithm, primarily used in the digital telecommunication systems of North America and Japan. Companding algorithms reduce the dynamic range of an audio signal...
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
G.711
G.711 is an ITU-T standard for audio companding. It is primarily used in telephony. The standard was released for usage in 1972. Its formal name is Pulse code modulation of voice frequencies. It is required standard in many technologies, for example in H.320 and H.323 specifications. It can also...
. 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
G.726
G.726 is an ITU-T ADPCM speech codec standard covering the transmission of voice at rates of 16, 24, 32, and 40 kbit/s. It was introduced to supersede both G.721, which covered ADPCM at 32 kbit/s, and G.723, which described ADPCM for 24 and 40 kbit/s. G.726 also introduced a new...
standard.
Some ADPCM techniques are used in Voice over IP
Voice over IP
Voice over Internet Protocol is a family of technologies, methodologies, communication protocols, and transmission techniques for the delivery of voice communications and multimedia sessions over Internet Protocol networks, such as the Internet...
communications. ADPCM was also used by Interactive Multimedia Association
Interactive Multimedia Association
The Interactive Multimedia Association was an industry association which developed a set of audio algorithms. The most important is the ADPCM algorithm which is in use by Apple and Microsoft....
for development of legacy audio codec known as ADPCM DVI, IMA ADPCM or DVI4, in the early 1990s.
Split-band or subband ADPCM
G.722G.722
G.722 is a ITU-T standard 7 kHz wideband speech codec operating at 48, 56 and 64 kbit/s. It was approved by ITU-T in November 1988. Technology of the codec is based on sub-band ADPCM ....
is a ITU-T
ITU-T
The ITU Telecommunication Standardization Sector is one of the three sectors of the International Telecommunication Union ; it coordinates standards for telecommunications....
standard wideband speech codec
Codec
A codec is a device or computer program capable of encoding or decoding a digital data stream or signal. The word codec is a portmanteau of "compressor-decompressor" or, more commonly, "coder-decoder"...
operating at 48, 56 and 64 kbit/s, based on subband coding with two channels and ADPCM coding of each. Before the digitization process, it catches the analog signal and divides it in frequency bands with QMF
Quadrature mirror filter
In digital signal processing, a quadrature mirror filter is a filter most commonly used to implement a filter bank that splits an input signal into two bands...
filters (quadrature mirror filters) to get two subbands of the signal. When the ADPCM bitstream of each subband is obtained, the results are multiplexed and the next step is storage or transmission of the data. The decoder has to perform the reverse process, that is, demultiplex and decode each subband of the bitstream and recombine them.
Referring to the coding process, in some applications as voice coding, the subband that includes the voice is coded with more bits than the others. It is a way to reduce the file size.