Adaptive DPCM
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Adaptive differential pulse-code modulation (ADPCM) is a variant of
differential pulse-code modulation Differential pulse-code modulation (DPCM) is a signal encoder that uses the baseline of pulse-code modulation (PCM) but adds some functionalities based on the prediction of the samples of the signal. The input can be an analog signal or a digital ...
(DPCM) that varies the size of the quantization step, to allow further reduction of the required data bandwidth for a given
signal-to-noise ratio Signal-to-noise ratio (SNR or S/N) is a measure used in science and engineering that compares the level of a desired signal to the level of background noise. SNR is defined as the ratio of signal power to the noise power, often expressed in de ...
. 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 for
speech coding Speech coding is an application of data compression of digital audio signals containing speech. Speech coding uses speech-specific parameter estimation using audio signal processing techniques to model the speech signal, combined with generic d ...
by P. Cummiskey, Nikil S. Jayant and
James L. Flanagan James Loton Flanagan (August 26, 1925 – August 25, 2015) was an American electrical engineer. He was Rutgers University's vice president for research until 2004. He was also director of Rutgers' Center for Advanced Information Processing and t ...
at
Bell Labs Nokia Bell Labs, originally named Bell Telephone Laboratories (1925–1984), then AT&T Bell Laboratories (1984–1996) and Bell Labs Innovations (1996–2007), is an American industrial research and scientific development company owned by mul ...
in 1973.


In telephony

In
telephony Telephony ( ) is the field of technology involving the development, application, and deployment of telecommunication services for the purpose of electronic transmission of voice, fax, or data, between distant parties. The history of telephony is i ...
, 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 Digital Signal 0 (DS0) is a basic digital signaling rate of 64 kilobits per second (kbit/s), corresponding to the capacity of one analog voice-frequency-equivalent communication channel. The DS0 rate, and its equivalents E0 in the E-carrier system ...
. The default
signal compression Signal compression is the use of various techniques to increase the quality or quantity of signal parameters transmitted through a given telecommunications channel. Types of signal compression include: * Bandwidth compression *Data compression *Dy ...
encoding on a DS0 is either μ-law (mu-law) PCM (North America and Japan) or
A-law An A-law algorithm is a standard companding algorithm, used in European 8-bit PCM digital communications systems to optimize, i.e. modify, the dynamic range of an analog signal for digitizing. It is one of two versions of the G.711 standar ...
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 Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet t ...
communications. In the early 1990s, ADPCM was also used by
Interactive Multimedia Association The Interactive Multimedia Association (IMA) 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 Microsoft Corporation is an American multina ...
to develop the legacy audio codecs ADPCM DVI, IMA ADPCM, and DVI4.


Split-band or subband ADPCM

G.722 is an
ITU-T The ITU Telecommunication Standardization Sector (ITU-T) is one of the three sectors (divisions or units) of the International Telecommunication Union (ITU). It is responsible for coordinating standards for telecommunications and Information Co ...
standard wideband speech
codec A codec is a device or computer program that encodes or decodes a data stream or signal. ''Codec'' is a portmanteau of coder/decoder. In electronic communications, an endec is a device that acts as both an encoder and a decoder on a signal or ...
operating at 48, 56 and 64 kbit/s, based on
subband coding In signal processing, sub-band coding (SBC) is any form of transform coding that breaks a signal into a number of different frequency bands, typically by using a fast Fourier transform, and encodes each one independently. This decomposition i ...
with two channels and ADPCM coding of each. Before the digitization process, it catches the analog signal and divides it in frequency bands with quadrature mirror filters (QMF) 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.


Software

The Windows Sound System supported ADPCM in WAV files. The
FFmpeg FFmpeg is a free and open-source software project consisting of a suite of libraries and programs for handling video, audio, and other multimedia files and streams. At its core is the command-line ffmpeg tool itself, designed for processing of vid ...
audio codecs supporting ADPCM are ''adpcm_ima_qt'', ''adpcm_ima_wav'', ''adpcm_ms'', ''adpcm_swf'' and ''adpcm_yamaha''.


See also

*
Audio coding format An audio coding format (or sometimes audio compression format) is a content representation format for storage or transmission of digital audio (such as in digital television, digital radio and in audio and video files). Examples of audio coding ...
*
Audio data compression In information theory, data compression, source coding, or bit-rate reduction is the process of encoding information using fewer bits than the original representation. Any particular compression is either lossy or lossless. Lossless compressi ...
*
Pulse-code modulation Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals. It is the standard form of digital audio in computers, compact discs, digital telephony and other digital audio applications. In a PCM stream, the ...
(PCM)


References

{{Compression Methods Speech codecs Audio codecs