Audio coding format: Difference between revisions

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{{short description|Digitally coded format for audio signals}}

[[File:Opus quality comparison colorblind compatible.svg|thumb|Comparison of coding efficiency between popular audio formats]]

An '''audio coding format'''<ref>The term "audio coding" can be seen in e.g. the name [[Advanced Audio Coding]], and is analogous to the term [[video coding format|video coding]]</ref> (or sometimes '''audio compression format''') is a content representationencoded 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 formats include [[MP3]], [[Advanced Audio Coding|AAC]], [[Vorbis]], [[FLAC]], and [[Opus (audio format)|Opus]]. A specific software or hardware implementation capable of [[Data_compression#Audio|audio compression]] and decompression to/from a specific audio coding format is called an ''[[audio codec]]''; an example of an audio codec is [[LAME]], which is one of several different codecs which implements encoding and decoding audio in the [[MP3]] audio coding format in software.

Some audio coding formats are documented by a detailed [[technical specification]] document known as an '''audio coding specification'''. Some such specifications are written and approved by [[standardization organizationsorganization]]s as [[technical standardsstandard]]s, and are thus known as an '''audio coding standard'''. The term "standard" is also sometimes used for [[de facto standard|''de facto'' standards]] as well as formal standards.

Audio content encoded in a particular audio coding format is normally encapsulated within a [[container format]]. As such, the user normally doesn't have a raw [[Advanced Audio Coding|AAC]] file, but instead has a .m4a [[audio file format|audio file]], which is a [[MPEG-4 Part 14]] container containing AAC-encoded audio. The container also contains Matadata[[metadata]] such as title and other tags, and perhaps an index for fast seeking.<ref>{{Cite web | url=http://superuser.com/questions/357686/where-is-synchronization-information-stored-in-container-formats | title=Video – Where is synchronization information stored in container formats?}}</ref> A notable exception is [[MP3]] files, which are raw audio coding without a container format. De facto standards for adding metadata tags such as title and artist to MP3s, such as [[ID3]], are hacks[[Hack (computer science)#In computer science|hack]]s which work by appending the tags to the MP3, and then relying on the MP3 player to recognize the chunk as malformed audio coding and therefore skip it. In video files with audio, the encoded audio content is bundled with video (in a [[video coding format]]) inside a [[multimedia container format]].

An audio coding format does not dictate all algorithms[[algorithm]]s used by a [[codec]] implementing the format. An important part of how lossy audio compression works is by removing data in ways humans can't hear, according to a [[psychoacoustic model]]; the implementer of an encoder has some freedom of choice in which data to remove (according to their psychoacoustic model).

==Lossless, lossy, and uncompressed audio coding formats==

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[[File:Placa-audioPC-925.jpg|right|thumb|Solidyne 922: The world's first commercial audio bit compression [[sound card]] for PC, 1990]]

In 1950, [[Bell Labs]] filed the patent on [[differential pulse-code modulation]] (DPCM).<ref name="DPCM">{{US patent reference|inventor=C. Chapin Cutler|title=Differential Quantization of Communication Signals|number=2605361|A-Datum=1950-06-29|issue-date=1952-07-29}}</ref> [[Adaptive DPCM]] (ADPCM) was introduced by P. Cummiskey, [[Nikil Jayant|Nikil S. Jayant]] and [[James L. Flanagan]] at [[Bell Labs]] in 1973.<ref>{{cite journal|doi=10.1002/j.1538-7305.1973.tb02007.x|url=https://ieeexplore.ieee.org/document/6770730|title=Adaptive Quantization in Differential PCM Coding of Speech|year=1973|last1=Cummiskey|first1=P.|last2=Jayant|first2=N. S.|last3=Flanagan|first3=J. L.|journal=Bell System Technical Journal|volume=52|issue=7|pages=1105–1118|url-access=subscription}}</ref><ref>{{cite journal |last1=Cummiskey |first1=P. |last2=Jayant |first2=Nikil S. |last3=Flanagan |first3=J. L. |title=Adaptive quantization in differential PCM coding of speech |journal=The Bell System Technical Journal |date=1973 |volume=52 |issue=7 |pages=1105–1118 |doi=10.1002/j.1538-7305.1973.tb02007.x |issn=0005-8580}}</ref>

[[Perceptual coding]] was first used for [[speech coding]] compression, with [[linear predictive coding]] (LPC).<ref name="Schroeder2014">{{cite book |last1=Schroeder |first1=Manfred R. |title=Acoustics, Information, and Communication: Memorial Volume in Honor of Manfred R. Schroeder |date=2014 |publisher=Springer |isbn=9783319056609 |chapter=Bell Laboratories |page=388 |chapter-url=https://books.google.com/books?id=d9IkBAAAQBAJ&pg=PA388}}</ref> Initial concepts for LPC date back to the work of [[Fumitada Itakura]] ([[Nagoya University]]) and Shuzo Saito ([[Nippon Telegraph and Telephone]]) in 1966.<ref>{{cite journal |last1=Gray |first1=Robert M. |title=A History of Realtime Digital Speech on Packet Networks: Part II of Linear Predictive Coding and the Internet Protocol |journal=Found. Trends Signal Process. |date=2010 |volume=3 |issue=4 |pages=203–303 |doi=10.1561/2000000036 |url=https://ee.stanford.edu/~gray/lpcip.pdf |issn=1932-8346|doi-access=free }}</ref> During the 1970s, [[Bishnu S. Atal]] and [[Manfred R. Schroeder]] at [[Bell Labs]] developed a form of LPC called [[adaptive predictive coding]] (APC), a perceptual coding algorithm that exploited the masking properties of the human ear, followed in the early 1980s with the [[code-excited linear prediction]] (CELP) algorithm which achieved a significant compression ratio for its time.<ref name="Schroeder2014"/> Perceptual coding is used by modern audio compression formats such as [[MP3]]<ref name="Schroeder2014"/> and [[Advanced Audio Codec|AAC]].

[[Discrete cosine transform]] (DCT), developed by [[Nasir Ahmed (engineer)|Nasir Ahmed]], T. Natarajan and [[K. R. Rao]] in 1974,<ref name="DCT">{{cite journal |author1=Nasir Ahmed |author1-link=N. Ahmed |author2=T. Natarajan |author3=Kamisetty Ramamohan Rao |journal=IEEE Transactions on Computers |title=Discrete Cosine Transform |volume=C-23 |issue=1 |pages=90–93 |date=January 1974 |doi=10.1109/T-C.1974.223784 |s2cid=149806273 |url=https://www.ic.tu-berlin.de/fileadmin/fg121/Source-Coding_WS12/selected-readings/Ahmed_et_al.__1974.pdf |access-date=2019-10-20 |archive-date=2016-12-08 |archive-url=https://web.archive.org/web/20161208075733/https://www.ic.tu-berlin.de/fileadmin/fg121/Source-Coding_WS12/selected-readings/Ahmed_et_al.__1974.pdf |url-status=dead }}</ref> provided the basis for the [[modified discrete cosine transform]] (MDCT) used by modern audio compression formats such as MP3<ref name="Guckert">{{cite web |last1=Guckert |first1=John |title=The Use of FFT and MDCT in MP3 Audio Compression |url=http://www.math.utah.edu/~gustafso/s2012/2270/web-projects/Guckert-audio-compression-svd-mdct-MP3.pdf |website=[[University of Utah]] |date=Spring 2012 |access-date=14 July 2019}}</ref> and AAC. MDCT was proposed by J. P. Princen, A. W. Johnson and A. B. Bradley in 1987,<ref>{{cite book|doi=10.1109/ICASSP.1987.1169405|chapter-url=https://ieeexplore.ieee.org/document/1169405|chapter=Subband/Transform coding using filter bank designs based on time ___domain aliasing cancellation|title=ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing|year=1987|last1=Princen|first1=J.|last2=Johnson|first2=A.|last3=Bradley|first3=A.|volume=12|pages=2161–2164|s2cid=58446992}}</ref> following earlier work by Princen and Bradley in 1986.<ref>{{cite journal|doi=10.1109/TASSP.1986.1164954|url=https://ieeexplore.ieee.org/document/1164954|title=Analysis/Synthesis filter bank design based on time ___domain aliasing cancellation|year=1986|last1=Princen|first1=J.|last2=Bradley|first2=A.|journal=IEEE Transactions on Acoustics, Speech, and Signal Processing|volume=34|issue=5|pages=1153–1161}}</ref> The MDCT is used by modern audio compression formats such as [[Dolby Digital]],<ref name="Luo">{{cite book |last1=Luo |first1=Fa-Long |title=Mobile Multimedia Broadcasting Standards: Technology and Practice |date=2008 |publisher=[[Springer Science & Business Media]] |isbn=9780387782638 |page=590 |url=https://books.google.com/books?id=l6PovWat8SMC&pg=PA590}}</ref><ref>{{cite journal |last1=Britanak |first1=V. |title=On Properties, Relations, and Simplified Implementation of Filter Banks in the Dolby Digital (Plus) AC-3 Audio Coding Standards |journal=IEEE Transactions on Audio, Speech, and Language Processing |date=2011 |volume=19 |issue=5 |pages=1231–1241 |doi=10.1109/TASL.2010.2087755|bibcode=2011ITASL..19.1231B |s2cid=897622 }}</ref> [[MP3]],<ref name="Guckert"/> and [[Advanced Audio Coding]] (AAC).<ref name=brandenburg>{{cite web|url=http://graphics.ethz.ch/teaching/mmcom12/slides/mp3_and_aac_brandenburg.pdf|title=MP3 and AAC Explained|last=Brandenburg|first=Karlheinz|year=1999|url-status=live|archive-url=https://web.archive.org/web/20170213191747/https://graphics.ethz.ch/teaching/mmcom12/slides/mp3_and_aac_brandenburg.pdf|archive-date=2017-02-13}}</ref>

==List of lossy formats==

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| rowspan="2" | 36–54%{{refn|group=n|name=MarketShareNote|The report combines AC-3 & E-AC-3 and separates [[Dolby Atmos]] from its market share calculation. Dolby Atmos can be encoded either lossily with E-AC-3/[[Dolby AC-4|AC-4]]<ref>{{Cite web |date=2023-05-23 |title=Does Dolby AC-4 support Dolby Atmos? |url=https://professionalsupport.dolby.com/s/article/Does-Dolby-AC-4-support-Dolby-Atmos |access-date=2024-11-08 |website=Dolby Professional Support}}</ref> or losslessly with [[Dolby TrueHD]]. [[Music streaming service|Music]] and [[Video on demand|video streaming]] providers typically use Dolby Digital Plus augmented with Dolby Atmos, whereas [[Music download|digital downloads]] and [[Blu-ray|Blu-ray discs]] typically use Dolby TrueHD augmented with Dolby Atmos.<ref>{{Cite web |date=2023-05-03 |title=Just wait until you hear lossless Dolby Atmos Music |url=https://www.digitaltrends.com/home-theater/lossless-spatial-audio-dolby-atmos-music/ |access-date=2024-11-08 |website=Digital Trends |language=en}}</ref>}}

| rowspan="2" |37–61%{{refn|group=n|name=MarketShareNote}}

| <ref name="Luo" /><ref name="Britanak2011">{{cite journal |last1=Britanak |first1=V. |title=On Properties, Relations, and Simplified Implementation of Filter Banks in the Dolby Digital (Plus) AC-3 Audio Coding Standards |journal=IEEE Transactions on Audio, Speech, and Language Processing |date=2011 |volume=19 |issue=5 |pages=1231–1241 |doi=10.1109/TASL.2010.2087755|bibcode=2011ITASL..19.1231B |s2cid=897622 }}</ref>

|-

|[[Dolby Digital|Dolby Digital Plus]] (E-AC-3)

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===Speech===

* Linear predictive{{further|Speech coding (LPC)}}

** Adaptive predictive coding (APC)

* [[Linear predictive coding]] (LPC)

** [[Adaptive predictive coding]] (APC)

** [[Code-excited linear prediction]] (CELP)

** [[Algebraic code-excited linear prediction]] (ACELP)

** [[Relaxed code-excited linear prediction]] (RCELP)

** [[Low-delay CELP]] (LD-CELP)

** [[Adaptive Multi-Rate audio codec|Adaptive Multi-Rate]] (used in [[GSM]] and [[3GPP]])

** [[Codec 2]] (noted for its lack of patent restrictions)

** [[Speex]] (noted for its lack of patent restrictions)

* [[Modified discrete cosine transform]] (MDCT)

** [[AAC-LD]]

** [[CELT|Constrained Energy Lapped Transform]] (CELT)

** [[Opus (audio format)|Opus]] (mostly for real-time applications)

== List of lossless formats ==

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* TTA (True Audio Lossless)

* [[WavPack]] (WavPack lossless)

* [[Windows Media Audio 9 Lossless|WMA Lossless]] (Windows Media Lossless)

==See also==

* [[Comparison of audio coding formats]]

* [[Data compression#Audio]]

* [[Audio file format]]

* [[List of audio compression formats]]

== Notes ==

<references group="n" />

==References==

<references/>

{{Compression formats}}

[[Category:Audio file formats| ]]