View source for Adaptive differential pulse-code modulation

{{Short description|Technique used to encode voices in telephony}}
'''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 data 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.<ref>{{cite book | title = Principles of Digital Audio | author = Ken C. Pohlmann | publisher = McGraw-Hill Professional | year = 2005 | isbn = 978-0-07-144156-8 | url = https://books.google.com/books?id=VZw6z9a03ikC&q=adpcm+adaptive+differential+pulse-code+modulation&pg=PA122 }}</ref>
ADPCM was developed for [[speech coding]] by P. Cummiskey, [[Nikil Jayant|Nikil S. Jayant]] and [[James L. Flanagan]] at [[Bell Labs]] in 1973.<ref>{{cite journal |last1=Cummiskey |first1=P. |last2=Jayant |first2=Nikil S. |author2-link=Nikil Jayant |last3=Flanagan |first3=James L. |author3-link=James L. Flanagan |title=Adaptive quantization in differential PCM coding of speech |journal=[[The Bell System Technical Journal]] |date=September 1973 |volume=52 |issue=7 |pages=1105–1118 |doi=10.1002/j.1538-7305.1973.tb02007.x}}</ref>
==In telephony==
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&nbsp;kbit/s digital signal known as [[DS0]]. The default [[signal compression]] encoding on a DS0 is either [[Mu-law algorithm|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.<ref>[https://www.cs.columbia.edu/~hgs/audio/dvi/ Recommended Practices for Enhancing Digital Audio Compatibility in Multimedia Systems] – legacy IMA ADPCM specification, Retrieved on 2009-07-06.</ref>
==Split-band or subband ADPCM==
{{anchor|SB-ADPCM}}
[[G.722]]<ref>[http://www.itu.int/rec/recommendation.asp?type=folders&lang=e&parent=T-REC-G.722 ITU-T G.722 page]. ITU-T Recommendation G.722 (11/88), "7&nbsp;kHz audio-coding within 64&nbsp;kbit/s".</ref> is an [[ITU-T]] standard wideband speech [[codec]] operating at 48, 56 and 64&nbsp;kbit/s, based on [[subband coding]] with two channels and ADPCM coding of each.<ref>{{cite book | title = Digital Compression for Multimedia |author1=Jerry D. Gibson |author2=Toby Berger |author3=Tom Lookabaugh | publisher = Morgan Kaufmann | year = 1998 | isbn = 978-1-55860-369-1 | url = https://books.google.com/books?id=aqQ2Ry6spu0C&q=G.722+adpcm+subband+split&pg=PA265 }}</ref> Before the digitization process, it catches the analog signal and divides it in frequency bands with [[quadrature mirror filter]]s (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.
[[Image:Adpcm en.svg]]
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.<ref>{{cite web
|url=http://support.microsoft.com/kb/89879
|url-status=dead
|archive-url=https://web.archive.org/web/20131231000305/http://support.microsoft.com/kb/89879
|archive-date=2013-12-31
|title=Differences Between PCM/ADPCM Wave Files Explained
|publisher=[[Microsoft Knowledge Base]]
|date=2011-09-24
|work=KB 89879 Revision 3.0
|access-date=2013-12-30}}</ref>
{{As of|2024|10|31}}, [[FFmpeg]] include 50 built-in ADPCM decoders and 16 encoders, some catering to niche purposes. For instance, "ADPCM Westwood Studios IMA" (<code>adpcm_ima_ws</code>) encodes and decodes the audio of the old [[Command & Conquer]] video games.<ref>{{cite web
|url=http://ffmpeg.org/general.html#Audio-Codecs
|title=General Documentation |at=Audio Codecs
|website=FFmpeg.org <!-- automatically updated daily, therefore no date -->
|access-date=2024-10-31}}</ref><ref>{{cite web
|url=https://github.com/FFmpeg/FFmpeg/blob/87068b9600daa522e3f45b5501ecd487a3c0be57/libavcodec/adpcmenc.c#L998-L1034
|title=adpcmenc.c at 87068b9600daa522e3f45b5501ecd487a3c0be57
|at = Lines 998–1034
|work=FFmpeg/FFmpeg repo
|via=[[GitHub]]
|date=2024-10-31
}}</ref>
The DSP in the [[GameCube]] supports ADPCM encoding on 64 simultaneous audio channels.
==See also==
*[[Audio coding format]]
*[[Audio data compression]]
*[[Pulse-code modulation]] (PCM)
==References==
<references />
{{Compression Methods}}
[[Category:Speech codecs]]
[[Category:Audio codecs]]