Editing Digital audio

{{Short description|Technology that records, stores, and reproduces sound}}
{{redirect-distinguish|Digital Audio|Digital Audio (magazine){{!}}''Digital Audio'' (magazine)}}
{{redirect|Digital music|modern music composed by digital or electronic means|Computer music|and|Electronic music}}
{{Use American English|date=December 2024}}

[[File:Zoom H4n audio recording levels.jpg|thumb|Audio levels display on a digital audio recorder ([[Zoom H4n]])]]

'''Digital audio''' is a representation of sound recorded in, or converted into, [[digital signal (signal processing)|digital form]]. In digital audio, the [[sound wave]] of the [[audio signal]] is typically encoded as numerical [[sampling (signal processing)|samples]] in a continuous sequence. For example, in [[CD audio]], samples are taken 44,100 [[Hertz|times per second]], each with 16-bit [[audio bit depth|resolution]]. Digital audio is also the name for the entire technology of [[sound recording and reproduction]] using audio signals that have been encoded in digital form. Following significant advances in digital audio technology during the 1970s and 1980s, it gradually replaced [[comparison of analog and digital recording|analog audio technology]] in many areas of [[audio engineering]], [[record production]] and [[telecommunications]] in the 1990s and 2000s.

In a digital audio system, an [[analog signal|analog electrical signal]] representing the sound is converted with an [[analog-to-digital converter]] (ADC) into a digital signal, typically using [[pulse-code modulation]] (PCM). This digital signal can then be recorded, edited, modified, and copied using [[computer]]s, audio playback machines, and other digital tools. For playback, a [[digital-to-analog converter]] (DAC) performs the reverse process, converting a digital signal back into an analog signal, which is then sent through an [[audio power amplifier]] and ultimately to a [[loudspeaker]].

Digital audio systems may include [[audio compression (data)|compression]], [[computer data storage|storage]], [[digital signal processing|processing]], and [[data transmission|transmission]] components. Conversion to a digital format allows convenient manipulation, storage, transmission, and retrieval of an audio signal. Unlike analog audio, in which making copies of a recording results in [[generation loss]] and degradation of signal quality, digital audio allows an infinite number of copies to be made without any degradation of signal quality.

==Overview==
[[File:4-bit-linear-PCM.svg|300px|right|thumb|A sound wave, in red, represented digitally, in blue (after [[sampling (signal processing)|sampling]] and 4-bit [[quantization (signal processing)|quantization]]).]]

Digital audio technologies are used in the recording, manipulation, mass-production, and distribution of sound, including recordings of [[song]]s, instrumental pieces, [[podcast]]s, sound effects, and other sounds. Modern [[Music download|online music distribution]] depends on digital recording and [[Audio compression (data)|data compression]]. The availability of music as data files, rather than as physical objects, has significantly reduced the costs of distribution as well as making it easier to share copies.<ref name="Janssens">{{cite journal|last=Janssens|first=Jelle|year=2009|title=The Music Industry on (the) Line? Surviving Music Piracy in a Digital Era|journal= European Journal of Crime, Criminal Law and Criminal Justice|volume=77|issue=96|pages=77–96|doi=10.1163/157181709X429105|author2=Stijn Vandaele|author3=Tom Vander Beken|hdl=1854/LU-608677|url=https://biblio.ugent.be/publication/608677 |hdl-access=free}}</ref> Before digital audio, the music industry distributed and sold music by selling physical copies in the form of [[Phonograph record|records]] and [[cassette tape]]s. With digital audio and online distribution systems such as [[iTunes]], companies sell digital sound files to consumers, which the consumer receives over the Internet. Popular streaming services such as [[Apple Music]], [[Spotify]], or [[YouTube]], offer temporary access to the digital file, and are now the most common form of music consumption.<ref>{{Cite journal |last1=Liikkanen |first1=Lassi A. |last2=Åman |first2=Pirkka |date=May 2016 |title=Shuffling Services: Current Trends in Interacting with Digital Music |url=https://academic.oup.com/iwc/article-lookup/doi/10.1093/iwc/iwv004 |journal=Interacting with Computers |language=en |volume=28 |issue=3 |pages=352–371 |doi=10.1093/iwc/iwv004 |issn=0953-5438|url-access=subscription }}</ref>

An analog audio system converts physical waveforms of sound into electrical representations of those waveforms by use of a [[transducer]], such as a [[microphone]]. The sounds are then stored on an analog medium such as [[magnetic tape]], or transmitted through an analog medium such as a [[telephone line]] or [[Radio broadcasting|radio]]. The process is reversed for reproduction: the electrical audio signal is [[amplifier|amplified]] and then converted back into physical waveforms via a [[loudspeaker]]. Analog audio retains its fundamental wave-like characteristics throughout its storage, transformation, duplication, and amplification.

[[Analog signal|Analog audio signal]]s are susceptible to noise and distortion, due to the innate characteristics of electronic circuits and associated devices. Disturbances in a [[digital system]] do not result in error unless they are so large as to result in a symbol being misinterpreted as another symbol or disturbing the sequence of symbols. It is, therefore, generally possible to have an entirely error-free digital audio system in which no noise or distortion is introduced between conversion to digital format and conversion back to analog.{{efn|Anti-alias filtering and optional digital signal processing may degrade the audio signal via passband ripple, non-linear phase shift, numeric precision quantization noise or time distortion of transients. However, these potential degradations can be limited by careful digital design.<ref>{{cite web|last1=Story|first1=Mike|title=A Suggested Explanation For (Some Of) The Audible Differences Between High Sample Rate And Conventional Sample Rate Audio Material |date=September 1997|url=http://sdg-master.com:80/lesestoff/aes97ny.pdf |publisher=dCS Ltd|archive-date=28 November 2009|archive-url=https://web.archive.org/web/20091128021651/http://sdg-master.com:80/lesestoff/aes97ny.pdf|url-status=live}}</ref>}}

A digital audio signal may be encoded for correction of any errors that might occur in the storage or transmission of the signal. This technique, known as [[channel coding]], is essential for broadcast or recorded digital systems to maintain bit accuracy. [[Eight-to-fourteen modulation]] is the channel code used for the audio [[compact disc]] (CD).

===Conversion process===
[[File:A-D-A Flow.svg|thumb|alt=Analog to Digital to Analog conversion|The lifecycle of sound from its source, through an ADC, digital processing, a DAC, and finally as sound again.]]

If an audio signal is analog, a digital audio system starts with an ADC that converts an analog signal to a digital signal.{{efn|Some audio signals such as those created by [[Synthesizer|digital synthesis]] originate entirely in the digital domain, in which case analog to digital conversion does not take place.}} The ADC runs at a specified [[sampling rate]] and converts at a known bit resolution. [[CD audio]], for example, has a sampling rate of 44.1&nbsp;[[kHz]] (44,100&nbsp;samples per second), and has 16-bit [[Audio bit depth|resolution]] for each [[stereo]] channel. Analog signals that have not already been [[bandlimited]] must be passed through an [[anti-aliasing filter]] before conversion, to prevent the [[Aliasing|aliasing distortion]] that is caused by audio signals with frequencies higher than the [[Nyquist frequency]] (half the sampling rate).

A digital audio signal may be stored or transmitted. Digital audio can be stored on a CD, a [[digital audio player]], a [[hard drive]], a [[USB flash drive]], or any other digital [[data storage device]]. The digital signal may be altered through [[digital signal processing]], where it may be [[audio filter|filter]]ed or have [[audio signal processing|effect]]s applied. [[Sample-rate conversion]] including [[upsampling]] and [[downsampling]] may be used to change signals that have been encoded with a different sampling rate to a common sampling rate prior to processing. Audio data compression techniques, such as [[MP3]], [[Advanced Audio Coding]] (AAC), [[Opus (audio format)|Opus]], [[Ogg Vorbis]], or [[FLAC]], are commonly employed to reduce the file size. Digital audio can be carried over [[digital audio interface]]s such as [[AES3]] or [[MADI]]. Digital audio can be carried over a network using [[audio over Ethernet]], [[audio over IP]] or other [[streaming media]] standards and systems.

For playback, digital audio must be converted back to an analog signal with a DAC. According to the [[Nyquist–Shannon sampling theorem]], with some practical and theoretical restrictions,<!--there's jitter, device nonlinearities and tradeoffs in antialiasing filter design; quantization noise is introduced--> a band-limited version of the original analog signal can be accurately reconstructed from the digital signal.

During conversion, audio data can be embedded with a [[digital watermark]] to prevent piracy and unauthorized use. Watermarking is done using a direct-sequence spread-spectrum (DSSS) method. The audio information is then modulated by a pseudo-noise (PN) sequence, then shaped within the frequency domain and put back in the original signal. The strength of the embedding determines the strength of the watermark on the audio data.<ref>{{Cite journal |last1=Seok |first1=Jongwon |last2=Hong |first2=Jinwoo |last3=Kim |first3=Jinwoong |date=2002-06-01 |title=A Novel Audio Watermarking Algorithm for Copyright Protection of Digital Audio |journal=ETRI Journal |language=en |volume=24 |issue=3 |pages=181–189 |doi=10.4218/etrij.02.0102.0301 |s2cid=3008374 |issn=1225-6463|doi-access=free }}</ref>

==History==
===Coding===
{{Main|Audio coding format|Audio data compression}}

[[Pulse-code modulation]] (PCM) was invented by British scientist [[Alec Reeves]] in 1937.<ref>{{citation |url=http://www.bbc.co.uk/programmes/b00zs7v5 |publisher=BBC |title=Genius Unrecognised |date=2011-03-27 |access-date=2011-03-30}}</ref> In 1950, [[C. Chapin Cutler]] of [[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> a [[data compression]] algorithm. [[Adaptive DPCM]] (ADPCM) was introduced by P. Cummiskey, [[Nikil Jayant|Nikil S. Jayant]] and [[James L. Flanagan]] at Bell Labs in 1973.<ref>P. Cummiskey, Nikil S. Jayant, and J. L. Flanagan, "Adaptive quantization in differential PCM coding of speech", ''Bell Syst. Tech. J.'', vol. 52, pp. 1105—1118, Sept. 1973</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.<ref name="Schroeder2014"/>

[[Discrete cosine transform]] (DCT) coding, a [[lossy compression]] method first proposed by [[Nasir Ahmed (engineer)|Nasir Ahmed]] in 1972,<ref name="Ahmed">{{cite journal |last=Ahmed |first=Nasir |author-link=N. Ahmed |title=How I Came Up With the Discrete Cosine Transform |journal=[[Digital Signal Processing (journal)|Digital Signal Processing]] |date=January 1991 |volume=1 |issue=1 |pages=4–5 |doi=10.1016/1051-2004(91)90086-Z |bibcode=1991DSP.....1....4A |url=https://www.scribd.com/doc/52879771/DCT-History-How-I-Came-Up-with-the-Discrete-Cosine-Transform|url-access=subscription }}</ref><ref name="DCT">{{cite journal |author1=Nasir 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}}</ref> provided the basis for the [[modified discrete cosine transform]] (MDCT), which was developed by J. P. Princen, A. W. Johnson and A. B. Bradley in 1987.<ref>J. P. Princen, A. W. Johnson und A. B. Bradley: ''Subband/transform coding using filter bank designs based on time domain aliasing cancellation'', IEEE Proc. Intl. Conference on Acoustics, Speech, and Signal Processing (ICASSP), 2161–2164, 1987.</ref> The MDCT is the basis for most [[audio coding standards]], such as [[Dolby Digital]] (AC-3),<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> MP3 ([[MPEG]] Layer III),<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><ref name="Schroeder2014"/> AAC, [[Windows Media Audio]] (WMA), Opus and [[Vorbis]] ([[Ogg]]).<ref name="Luo"/>

===Recording===
{{Main|Digital recording}}
[[File:Reel to reel Hitachi I (1972).JPG|thumb|upright|Analog [[reel-to-reel tape recorder]]]]
[[File:Sony PCM-7030 of DR 20111102a-crop.jpg|thumb|Sony professional [[digital audio tape]] (DAT) recorder PCM-7030]]
[[File:REAPER_Digital_Audio_Workstation.jpg|thumb|[[Digital audio workstation]]]]

PCM was used in [[telecommunications]] applications long before its first use in commercial broadcast and recording. Commercial digital recording was pioneered in Japan by [[NHK]] and [[Nippon Columbia]] and their [[Denon]] brand, in the 1960s. The first commercial digital recordings were released in 1971.<ref name="Fine"/>

The [[BBC]] also began to experiment with digital audio in the 1960s. By the early 1970s, it had developed a 2-channel recorder, and in 1972 it deployed a digital audio transmission system that linked their broadcast center to their remote transmitters.<ref name="Fine">{{cite journal |url=http://www.aes.org/aeshc/pdf/fine_dawn-of-digital.pdf |access-date=2010-05-02 |journal=ARSC Journal |year=2008 |editor=Barry R. Ashpole |first=Thomas |last=Fine |title=The Dawn of Commercial Digital Recording}}</ref>

The first 16-bit PCM recording in the [[United States]] was made by [[Thomas Stockham]] at the [[Santa Fe Opera]] in 1976, on a [[Soundstream]] recorder. An improved version of the Soundstream system was used to produce several classical recordings by [[Telarc]] in 1978. The [[3M]] digital [[multitrack recorder]] in development at the time was based on BBC technology. The first all-digital album recorded on this machine was [[Ry Cooder]]'s ''[[Bop till You Drop]]'' in 1979. British record label [[Decca Records|Decca]] began development of its own 2-track digital audio recorders in 1978 and released the first European digital recording in 1979.<ref name="Fine"/>

Popular professional digital multitrack recorders produced by Sony/Studer ([[Digital Audio Stationary Head|DASH]]) and Mitsubishi ([[ProDigi]]) in the early 1980s helped to bring about digital recording's acceptance by the major record companies.  Machines for these formats had their own transports built-in as well, using [[reel-to-reel]] tape in either 1/4", 1/2", or 1" widths, with the audio data being recorded to the tape using a multi-track stationary tape head. [[PCM adaptor]]s allowed for stereo digital audio recording on a conventional NTSC or PAL [[video tape recorder]].

The 1982 introduction of the CD by [[Philips]] and [[Sony]] popularized digital audio with consumers.<ref name="Fine"/>

[[ADAT]] became available in the early 1990s, which allowed eight-track [[44,100 Hz|44.1]] or [[48,000 Hz|48&nbsp;kHz]] recording on S-VHS cassettes, and [[DTRS]] performed a similar function with Hi8 tapes.

Formats like ProDigi and DASH were referred to as '''SDAT''' (stationary-head digital audio tape) formats, as opposed to formats like the PCM adaptor-based systems and [[Digital Audio Tape]] (DAT), which were referred to as '''RDAT''' (rotating-head digital audio tape) formats,  due to their helical-scan process of recording.

Like the DAT cassette, ProDigi and DASH machines also accommodated the obligatory 44.1&nbsp;kHz sampling rate, but also 48&nbsp;kHz on all machines, and eventually a 96&nbsp;kHz sampling rate.  They overcame the problems that made typical analog recorders unable to meet the bandwidth (frequency range) demands of digital recording by a combination of higher tape speeds, narrower head gaps used in combination with metal-formulation tapes, and the spreading of data across multiple parallel tracks.

Unlike analog systems, modern [[digital audio workstation]]s and [[audio interface]]s allow as many channels in as many different sampling rates as the computer can effectively run at a single time. [[Avid Audio]] and [[Steinberg]] released the first digital audio workstation software programs in 1989.<ref name=":0">{{Cite journal |last=Reuter |first=Anders |date=2022-03-15 |title=Who let the DAWs Out? The Digital in a New Generation of the Digital Audio Workstation |url=https://www.tandfonline.com/doi/full/10.1080/03007766.2021.1972701 |journal=Popular Music and Society |language=en |volume=45 |issue=2 |pages=113–128 |doi=10.1080/03007766.2021.1972701 |s2cid=242779244 |issn=0300-7766|url-access=subscription }}</ref> Digital audio workstations make multitrack recording and mixing much easier for large projects which would otherwise be difficult with analog equipment.
{{clear}}

===Telephony===
{{Main|Digital telephony}}

The rapid development and wide adoption of PCM [[digital telephony]] was enabled by [[metal–oxide–semiconductor]] (MOS) [[switched capacitor]] (SC) circuit technology, developed in the early 1970s.<ref name="Allstot">{{cite book |last1=Allstot |first1=David J. |chapter=Switched Capacitor Filters |editor-last1=Maloberti |editor-first1=Franco |editor-last2=Davies |editor-first2=Anthony C. |title=A Short History of Circuits and Systems: From Green, Mobile, Pervasive Networking to Big Data Computing |date=2016 |publisher=[[IEEE Circuits and Systems Society]] |isbn=9788793609860 |pages=105–110 |chapter-url=https://ieee-cas.org/sites/default/files/a_short_history_of_circuits_and_systems-_ebook-_web.pdf |access-date=2019-11-29 |archive-date=2021-09-30 |archive-url=https://web.archive.org/web/20210930151716/https://ieee-cas.org/sites/default/files/a_short_history_of_circuits_and_systems-_ebook-_web.pdf |url-status=dead }}</ref> This led to the development of PCM codec-filter chips in the late 1970s.<ref name="Allstot"/><ref name="Gibson26">{{cite book |last1=Floyd |first1=Michael D. |last2=Hillman |first2=Garth D. |chapter=Pulse-Code Modulation Codec-Filters |title=The Communications Handbook |edition=2nd |date=8 October 2018 |orig-year=1st pub. 2000 |pages=26-1, 26-2, 26-3 |publisher=[[CRC Press]] |isbn=9781420041163 |chapter-url=https://books.google.com/books?id=Tokk5bZxB0MC&pg=SA26-PA1}}</ref> The [[silicon-gate]] [[CMOS]] (complementary MOS) PCM codec-filter chip, developed by [[David A. Hodges]] and W.C. Black in 1980,<ref name="Allstot"/> has since been the industry standard for digital telephony.<ref name="Allstot"/><ref name="Gibson26"/> By the 1990s, [[telecommunication network]]s such as the [[public switched telephone network]] (PSTN) had been largely [[digitized]] with [[VLSI]] (very [[large-scale integration]]) CMOS PCM codec-filters, widely used in [[electronic switching system]]s for [[telephone exchanges]], user-end [[modems]] and a range of [[digital transmission]] applications such as the [[integrated services digital network]] (ISDN), [[cordless telephones]] and [[cell phones]].<ref name="Gibson26"/>

==Technologies ==
Digital audio is used in [[broadcasting]] of audio. Standard technologies include [[Digital audio broadcasting]] (DAB), [[Digital Radio Mondiale]] (DRM), [[HD Radio]] and [[In-band on-channel]] (IBOC).

Digital audio in recording applications is stored on audio-specific technologies including CD, DAT, [[Digital Compact Cassette]] (DCC) and [[MiniDisc]]. Digital audio may be stored in a standard [[audio file format]]s and stored on a [[Hard disk recorder]], [[Blu-ray]] or [[DVD-Audio]]. Files may be played back on smartphones, computers or [[MP3 player]]. Digital audio resolution is measured in [[audio bit depth]]. Most digital audio formats use either 16-bit, 24-bit, and 32-bit resolution.

==See also==
*[[Digital audio editor]]
*[[Digital synthesizer]]
*[[Frequency modulation synthesis]]
*[[Sound chip]]
*[[Sound card]]
*[[Audio interface|Audio Interface]]
*[[Quantization (signal processing)|Quantization]]
*[[Sampling (signal processing)|Sampling]]
*[[Multitrack recording]]
*[[Digital audio workstation]]

==Notes==
{{Notelist}}

==References==
{{Reflist}}

==Further reading==
*Borwick, John, ed., 1994: ''Sound Recording Practice'' (Oxford: Oxford University Press)
*Bosi, Marina, and Goldberg, Richard E., 2003: ''Introduction to Digital Audio Coding and Standards'' (Springer)
*Ifeachor, Emmanuel C., and Jervis, Barrie W., 2002: ''Digital Signal Processing: A Practical Approach'' (Harlow, England: Pearson Education Limited)
*Rabiner, Lawrence R., and Gold, Bernard, 1975: ''Theory and Application of Digital Signal Processing'' (Englewood Cliffs, New Jersey: Prentice-Hall, Inc.)
*Watkinson, John, 1994: ''The Art of Digital Audio'' (Oxford: Focal Press)

==External links==
{{commons category|Digital audio}}
*{{cite web |author=Monty Montgomery |publisher=evolver.fm |date=2012-10-24 |title=Guest Opinion: Why 24/192 Music Downloads Make No Sense |url=http://evolver.fm/2012/10/04/guest-opinion-why-24192-music-downloads-make-no-sense/ |access-date=2012-12-07 |archive-date=2012-12-10 |archive-url=https://web.archive.org/web/20121210132914/http://evolver.fm/2012/10/04/guest-opinion-why-24192-music-downloads-make-no-sense/ |url-status=dead }}
*{{cite web |title=Coding High Quality Digital Audio |author=J. ROBERT STUART |url=http://www.meridian.co.uk/ara/coding2.pdf |access-date=2012-12-07 |archive-url=https://web.archive.org/web/20070627075502/http://www.meridian.co.uk/ara/coding2.pdf |archive-date=2007-06-27 |url-status=dead }}
*{{cite web |author=Dan Lavry |title=Sampling Theory For Digital Audio |url=http://lavryengineering.com/pdfs/lavry-sampling-theory.pdf |access-date=2012-12-07 |url-status=live |archive-url=https://web.archive.org/web/20120916040445/http://lavryengineering.com/pdfs/lavry-sampling-theory.pdf |archive-date=2012-09-16 }}
{{Spoken Wikipedia|En-Digital audio-article.ogg|date=2016-03-12}}

{{Audio broadcasting}}
{{Digital systems}}
{{Music technology}}

[[Category:Digital audio| ]]