Pulse-code modulation: Difference between revisions

Revision as of 09:15, 1 September 2022

Revision as of 21:17, 5 September 2022

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'''Pulse-code modulation''' ('''PCM''') is a method used to [[Digital signal (signal processing)|digitally]] represent sampled [[analog signal]]s. It is the standard form of [[digital audio]] in computers, [[compact disc]]s, [[digital telephony]] and other digital audio applications. In a PCM [[Stream (computing)|stream]], the [[amplitude]] of the analog signal is [[Sampling (signal processing)|sampled]] regularly at uniform intervals, and each sample is [[Quantization (signal processing)|quantized]] to the nearest value within a range of digital steps.

'''Linear pulse-code modulation''' ('''LPCM''') is a specific type of PCM in which the quantization levels are linearly uniform.<ref name=LOC1>{{Cite"LOC_LPCM" web|url=http://www.digitalpreservation.gov/formats/fdd/fdd000011.shtml |title=Linear Pulse Code Modulated Audio (LPCM) |publisher=[[Library of Congress]]|date=2008-02-19 }}</ref> This is in contrast to PCM encodings in which quantization levels vary as a function of amplitude (as with the [[A-law|A-law algorithm]] or the [[μ-law|μ-law algorithm]]). Though ''PCM'' is a more general term, it is often used to describe data encoded as LPCM.

A PCM stream has two basic properties that determine the stream's fidelity to the original analog signal: the [[sampling rate]], which is the number of times per second that samples are taken; and the [[Audio bit depth|bit depth]], which determines the number of possible digital values that can be used to represent each sample.

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Common sample depths for LPCM are 8, 16, 20 or 24 bits per [[sample (signal)|sample]].<ref name="rfc2586" /><ref name="rfc4856" /><ref name="rfc3190" /><ref>{{cite document |url=http://tools.ietf.org/html/rfc3108#page-62 |title=RFC 3108 – Conventions for the use of the Session Description Protocol (SDP) for ATM Bearer Connections |date=May 2001 |access-date=2010-03-16|last1=Mostafa |first1=Mohamed |last2=Kumar |first2=Rajesh }}</ref>

LPCM encodes a single sound channel. Support for multichannel audio depends on file format and relies on synchronization of multiple LPCM streams.<ref name=LOC1LOC_LPCM/><ref>{{Cite web|publisher=Library of Congress |url=httphttps://www.digitalpreservationloc.gov/preservation/digital/formats/fdd/fdd000016.shtml |title=PCM, Pulse Code Modulated Audio |access-date=20092022-0709-1805}}</ref> While two channels (stereo) is the most common format, systems can support up to 8 audio channels (7.1 surround)<ref name="rfc4856"/><ref name="rfc3190"/> or more.

Common sampling frequencies are 48 [[hertz|kHz]] as used with [[DVD]] format videos, or 44.1&nbsp;kHz as used in CDs. Sampling frequencies of 96&nbsp;kHz or 192&nbsp;kHz can be used on some equipment, but the benefits have been debated.<ref>{{Cite web|last=Christopher|first=Montgometry|title=24/192 Music Downloads, and why they do not make sense|url=http://people.xiph.org/~xiphmont/demo/neil-young.html|url-status=dead|archive-url=https://web.archive.org/web/20140906115306/http://people.xiph.org/~xiphmont/demo/neil-young.html|archive-date=2014-09-06|access-date=2013-03-16|publisher=Chris "Monty" Montgomery}}</ref>

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Some forms of PCM combine signal processing with coding. Older versions of these systems applied the processing in the analog ___domain as part of the analog-to-digital process; newer implementations do so in the digital ___domain. These simple techniques have been largely rendered obsolete by modern transform-based [[audio compression (data)|audio compression]] techniques, such as [[modified discrete cosine transform]] (MDCT) coding.

* Linear PCM (LPCM) is PCM with linear quantization.<ref>{{cite web |url=http://www.digitalpreservation.gov/formats/fdd/fdd000011.shtml |title=Linear Pulse Code Modulated Audio (LPCM) |publisher=The Library of Congress |access-date=March 21, 2010|datename=2008-02-19"LOC_LPCM" }}</ref>

* [[DPCM|Differential PCM]] (DPCM) encodes the PCM values as differences between the current and the predicted value. An algorithm predicts the next sample based on the previous samples, and the encoder stores only the difference between this prediction and the actual value. If the prediction is reasonable, fewer bits can be used to represent the same information. For audio, this type of encoding reduces the number of bits required per sample by about 25% compared to PCM.

* [[Adaptive differential pulse-code modulation]] (ADPCM) is a variant of DPCM that varies the size of the quantization step, to allow further reduction of the required bandwidth for a given [[signal-to-noise ratio]].

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* [http://wiki.multimedia.cx/index.php?title=PCM PCM description on MultimediaWiki]

* [http://www.ralph-miller.org/ Ralph Miller] and Bob Badgley invented multi-level PCM independently in their work at Bell Labs on [[SIGSALY]]: {{US patent|3912868}} filed in 1943: N-ary Pulse Code Modulation.

* [httphttps://www.digitalpreservationloc.gov/preservation/digital/formats/fdd/fdd000016.shtml Information about PCM]: A description of PCM with links to information about subtypes of this format (for example [[linear pulse-code modulation]]), and references to their specifications.

* [httphttps://www.digitalpreservationloc.gov/preservation/digital/formats/fdd/fdd000011.shtml Summary of LPCM] – Contains links to information about implementations and their specifications.

* [http://www.tactilemedia.com/info/MCI_Control_Info.html How to control internal/external hardware using Microsoft's Media Control Interface] – Contains information about, and specifications for the implementation of LPCM used in WAV files.

* [http://tools.ietf.org/html/rfc4856 RFC 4856 – Media Type Registration of Payload Formats in the RTP Profile for Audio and Video Conferences] – audio/L8 and audio/L16 (March 2007)