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| extension = .L16, .WAV, .AIFF, .AU, .PCM<ref name="rfc2586">{{cite journal|first1=Harald Tveit |last1=Alvestrand |last2=Salsman |first2=James |url=http://tools.ietf.org/html/rfc2586 |title=RFC 2586 – The Audio/L16 MIME content type |date=May 1999 |publisher=The Internet Society |doi=10.17487/RFC2586 |access-date=2010-03-16|url-access=subscription }}</ref>
| mime = audio/L16, audio/L8,<ref name="rfc4856">{{cite journal|first=S. |last=Casner |url=http://tools.ietf.org/html/rfc4856#page-17 |title=RFC 4856 – Media Type Registration of Payload Formats in the RTP Profile for Audio and Video Conferences – Registration of Media Type audio/L8 |date=March 2007 |publisher=The IETF Trust |doi=10.17487/RFC4856 |access-date=2010-03-16}}</ref> audio/L20, audio/L24<ref name="rfc3190">{{cite journal |last1=Bormann |first1=C. |last2=Casner |first2=S. |last3=Kobayashi |first3=K. |last4=Ogawa |first4=A.
|url=http://tools.ietf.org/html/rfc3190 |title=RFC 3190 – RTP Payload Format for 12-bit DAT Audio and 20- and 24-bit Linear Sampled Audio |date=January 2002 |publisher=The Internet Society |doi=10.17487/RFC3190 |access-date=2010-03-16|doi-access=free }}</ref><ref>{{cite web |url=https://www.iana.org/assignments/media-types/audio/ |title=Audio Media Types |publisher=Internet Assigned Numbers Authority |access-date=2010-03-16}}</ref>
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==Standard sampling precision and rates==
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 journal |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 |doi=10.17487/RFC3108 |url-access=subscription }}</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=LOC_LPCM/><ref>{{Cite web|publisher=Library of Congress |url=https://www.loc.gov/preservation/digital/formats/fdd/fdd000016.shtml |title=PCM, Pulse Code Modulated Audio |date=April 6, 2022 |access-date=2022-09-05}}</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 [[kHz]] as used with [[DVD]] format videos, or 44.1 kHz as used in CDs. Sampling frequencies of 96 kHz or 192 kHz can be used on some equipment, but the [[High-resolution audio#Controversy|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>
==Limitations==
The [[Nyquist–Shannon sampling theorem]] shows PCM devices can operate without introducing distortions within their designed frequency bands if they provide a sampling frequency at least twice that of the highest frequency contained in the input signal. For example, in [[telephony]], the usable [[voice frequency]] band ranges from approximately 300 to 3400 [[Hz]]
Regardless, there are potential sources of impairment implicit in any PCM system:
* Choosing a discrete value that is near but not exactly at the analog signal level for each sample leads to [[quantization error]]. When [[dither]]ing is used to compensate for this, it introduces additional noise.<ref group=note>Quantization error swings between -''q''/2 and ''q''/2. In the ideal case (with a fully linear ADC and signal level >> ''q'') it is [[uniform distribution (continuous)|uniformly distributed]] over this interval, with zero mean and variance of ''q''<sup>2</sup>/12.</ref>
* Between samples no measurement of the signal is made; the sampling theorem guarantees non-ambiguous representation and recovery of the signal only if it has no energy at frequency ''f<sub>s</sub>''/2 or higher (one half the sampling frequency, known as the [[Nyquist frequency]]); higher frequencies will not be correctly represented or recovered and add aliasing distortion to the signal below the Nyquist frequency.
* As samples are dependent on time, an accurate clock is required for accurate reproduction. If either the encoding or decoding clock is not stable, these imperfections will directly affect the output quality of the device.<ref group=note>A slight difference between the encoding and decoding clock frequencies is not generally a major concern; a small constant error is not noticeable. Clock error does become a major issue if the clock contains significant [[jitter]], however.</ref>
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