# Digital audio
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{{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](/source/Zoom_H4n))]]
'''Digital audio''' is a representation of sound recorded in, or converted into, [digital form](/source/digital_signal_(signal_processing)). In digital audio, the [sound wave](/source/sound_wave) of the [audio signal](/source/audio_signal) is typically encoded as numerical [samples](/source/sampling_(signal_processing)) in a continuous sequence. For example, in [CD audio](/source/CD_audio), [samples](/source/Sampling_(signal_processing)) are taken 44,100 [times per second](/source/Hertz), each with 16-bit [resolution](/source/audio_bit_depth). Digital audio is also the name for the entire technology of [sound recording and reproduction](/source/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 [analog audio technology](/source/comparison_of_analog_and_digital_recording) in many areas of [audio engineering](/source/audio_engineering), [record production](/source/record_production) and [telecommunications](/source/telecommunications) in the 1990s and 2000s.
In a digital audio system, an [analog electrical signal](/source/analog_signal) representing the sound is converted with an [analog-to-digital converter](/source/analog-to-digital_converter) (ADC) into a digital signal, typically using [pulse-code modulation](/source/pulse-code_modulation) (PCM). This digital signal can then be recorded, edited, modified, and copied using [computer](/source/computer)s, audio playback machines, and other digital tools. For playback, a [digital-to-analog converter](/source/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](/source/audio_power_amplifier) and ultimately to a [loudspeaker](/source/loudspeaker).
Digital audio systems may include [compression](/source/audio_compression_(data)), [storage](/source/computer_data_storage), [processing](/source/digital_signal_processing), and [transmission](/source/data_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](/source/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](/source/sampling_(signal_processing)) and 4-bit [quantization](/source/quantization_(signal_processing))).]]
Digital audio technologies are used in the recording, manipulation, mass-production, and distribution of sound, including recordings of [song](/source/song)s, instrumental pieces, [podcast](/source/podcast)s, sound effects, and other sounds. Modern [online music distribution](/source/Music_download) depends on digital recording and [data compression](/source/Audio_compression_(data)). 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.[{{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}}] Before digital audio, the music industry distributed and sold music by selling physical copies in the form of [records](/source/Phonograph_record) and [cassette tape](/source/cassette_tape)s. With digital audio and online distribution systems such as [iTunes](/source/iTunes), companies sell digital sound files to consumers, which the consumer receives over the Internet. Popular streaming services such as [Apple Music](/source/Apple_Music), [Spotify](/source/Spotify), or [YouTube](/source/YouTube), offer temporary access to the digital file, and are now the most common form of music consumption.[{{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 }}]
An analog audio system converts physical waveforms of sound into electrical representations of those waveforms by use of a [transducer](/source/transducer), such as a [microphone](/source/microphone). The sounds are then stored on an analog medium such as [magnetic tape](/source/magnetic_tape), or transmitted through an analog medium such as a [telephone line](/source/telephone_line) or [radio](/source/Radio_broadcasting). The process is reversed for reproduction: the electrical audio signal is [amplified](/source/amplifier) and then converted back into physical waveforms via a [loudspeaker](/source/loudspeaker). Analog audio retains its fundamental wave-like characteristics throughout its storage, transformation, duplication, and amplification.
[Analog audio signals](/source/Analog_signal) are susceptible to noise and distortion, due to the innate characteristics of electronic circuits and associated devices. Disturbances in a [digital system](/source/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.[{{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}}]}}
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](/source/channel_coding), is essential for broadcast or recorded digital systems to maintain bit accuracy. [Eight-to-fourteen modulation](/source/Eight-to-fourteen_modulation) is the channel code used for the audio [compact disc](/source/compact_disc) (CD).
===Conversion process===
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 [digital synthesis](/source/Synthesizer), 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](/source/sampling_rate) and converts at a known bit resolution. [CD audio](/source/CD_audio), for example, has a sampling rate of 44.1 [kHz](/source/kHz) (44,100 samples per second), and has 16-bit [resolution](/source/Audio_bit_depth) for each [stereo](/source/stereo) channel. Analog signals that have not already been [bandlimited](/source/bandlimited) must be passed through an [anti-aliasing filter](/source/anti-aliasing_filter) before conversion, to prevent the [aliasing distortion](/source/Aliasing) that is caused by audio signals with frequencies higher than the [Nyquist frequency](/source/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](/source/digital_audio_player), a [hard drive](/source/hard_drive), a [USB flash drive](/source/USB_flash_drive), or any other digital [data storage device](/source/data_storage_device). The digital signal may be altered through [digital signal processing](/source/digital_signal_processing), where it may be [filtered](/source/audio_filter) or have [effect](/source/audio_signal_processing)s applied. [Sample-rate conversion](/source/Sample-rate_conversion), including [upsampling](/source/upsampling) and [downsampling](/source/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](/source/MP3), [Advanced Audio Coding](/source/Advanced_Audio_Coding) (AAC), [Opus](/source/Opus_(audio_format)), [Ogg Vorbis](/source/Ogg_Vorbis), or [FLAC](/source/FLAC), are commonly employed to reduce the file size. Digital audio can be carried over [digital audio interface](/source/digital_audio_interface)s such as [AES3](/source/AES3) or [MADI](/source/MADI). Digital audio can be carried over a network using [audio over Ethernet](/source/audio_over_Ethernet), [audio over IP](/source/audio_over_IP) or other [streaming media](/source/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](/source/Nyquist%E2%80%93Shannon_sampling_theorem), with some practical and theoretical restrictions, 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](/source/digital_watermark) to prevent piracy and unauthorized use. Watermarking is done using a [direct-sequence spread-spectrum](/source/direct-sequence_spread-spectrum) (DSSS) method. The audio information is then modulated by a [pseudo-noise](/source/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.[{{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 }}]
==History==
===Coding===
{{Main|Audio coding format|Audio data compression}}
[Pulse-code modulation](/source/Pulse-code_modulation) (PCM) was invented by British scientist [Alec Reeves](/source/Alec_Reeves) in 1937.[{{citation |url=https://www.bbc.co.uk/programmes/b00zs7v5 |publisher=BBC |title=Genius Unrecognised |date=2011-03-27 |access-date=2011-03-30}}] In 1950, [C. Chapin Cutler](/source/C._Chapin_Cutler) of [Bell Labs](/source/Bell_Labs) filed the patent on [differential pulse-code modulation](/source/differential_pulse-code_modulation) (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}}] a [data compression](/source/data_compression) algorithm. [Adaptive DPCM](/source/Adaptive_DPCM) (ADPCM) was introduced by P. Cummiskey, [Nikil S. Jayant](/source/Nikil_Jayant) and [James L. Flanagan](/source/James_L._Flanagan) at Bell Labs in 1973.[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][{{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 |bibcode=1973BSTJ...52.1105C |issn=0005-8580}}]
[Perceptual coding](/source/Perceptual_coding) was first used for [speech coding](/source/speech_coding) compression, with [linear predictive coding](/source/linear_predictive_coding) (LPC).[{{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}}] Initial concepts for LPC date back to the work of [Fumitada Itakura](/source/Fumitada_Itakura) ([Nagoya University](/source/Nagoya_University)) and Shuzo Saito ([Nippon Telegraph and Telephone](/source/Nippon_Telegraph_and_Telephone)) in 1966.[{{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 }}] During the 1970s, [Bishnu S. Atal](/source/Bishnu_S._Atal) and [Manfred R. Schroeder](/source/Manfred_R._Schroeder) at Bell Labs developed a form of LPC called [adaptive predictive coding](/source/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](/source/code-excited_linear_prediction) (CELP) algorithm.[{{cite journal |last=Ahmed |first=Nasir |author-link=N. Ahmed |title=How I Came Up With the Discrete Cosine Transform |journal=[Digital Signal Processing](/source/Digital_Signal_Processing_(journal)) |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 }}][{{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 |bibcode=1974ITCmp.100...90A |s2cid=149806273 |url=https://www.ic.tu-berlin.de/fileadmin/fg121/Source-Coding_WS12/selected-readings/Ahmed_et_al.__1974.pdf}}] provided the basis for the [modified discrete cosine transform](/source/modified_discrete_cosine_transform) (MDCT), which was developed by J. P. Princen, A. W. Johnson and A. B. Bradley in 1987.[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.] The MDCT is the basis for most [audio coding standards](/source/audio_coding_standards), such as [Dolby Digital](/source/Dolby_Digital) (AC-3),[{{cite book |last1=Luo |first1=Fa-Long |title=Mobile Multimedia Broadcasting Standards: Technology and Practice |date=2008 |publisher=[Springer Science & Business Media](/source/Springer_Science_%26_Business_Media) |isbn=9780387782638 |page=590 |url=https://books.google.com/books?id=l6PovWat8SMC&pg=PA590}}] MP3 ([MPEG](/source/MPEG) Layer III),[{{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](/source/University_of_Utah) |date=Spring 2012 |access-date=14 July 2019}}][{{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}}]
The first 16-bit PCM recording in the [United States](/source/United_States) was made by [Thomas Stockham](/source/Thomas_Stockham) at the [Santa Fe Opera](/source/Santa_Fe_Opera) in 1976, on a [Soundstream](/source/Soundstream) recorder. An improved version of the Soundstream system was used to produce several classical recordings by [Telarc](/source/Telarc) in 1978. The [3M](/source/3M) digital [multitrack recorder](/source/multitrack_recorder) in development at the time was based on BBC technology. The first all-digital album recorded on this machine was [Ry Cooder](/source/Ry_Cooder)'s ''[Bop till You Drop](/source/Bop_till_You_Drop)'' in 1979. British record label [Decca](/source/Decca_Records) began development of its own 2-track digital audio recorders in 1978 and released the first European digital recording in 1979.[{{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 }}] 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](/source/digital_telephony) was enabled by [metal–oxide–semiconductor](/source/metal%E2%80%93oxide%E2%80%93semiconductor) (MOS) [switched capacitor](/source/switched_capacitor) (SC) circuit technology, developed in the early 1970s.[{{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](/source/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 }}] This led to the development of PCM codec-filter chips in the late 1970s.[{{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](/source/CRC_Press) |isbn=9781420041163 |chapter-url=https://books.google.com/books?id=Tokk5bZxB0MC&pg=SA26-PA1}}] The [silicon-gate](/source/silicon-gate) [CMOS](/source/CMOS) (complementary MOS) PCM codec-filter chip, developed by [David A. Hodges](/source/David_A._Hodges) and W.C. Black in 1980,