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{{short description|Electronic manipulation of audio signals}}
{{Redirect|Audio processor|audio processing chips|Sound chip}}
{{More citations needed|date=June 2021}}

'''Audio signal processing''' is a subfield of [[signal processing]] that is concerned with the electronic manipulation of [[audio signal]]s. Audio signals are electronic representations of [[sound wave]]s—[[longitudinal wave]]s which travel through air, consisting of compressions and rarefactions. The energy contained in audio signals or [[sound power level]] is typically measured in [[decibel]]s.  As audio signals may be represented in either [[Digital signal (signal processing)|digital]] or [[analog signal|analog]] format, processing may occur in either domain.  Analog processors operate directly on the electrical signal, while digital processors operate mathematically on its digital representation.

== History ==
The motivation for audio signal processing began at the beginning of the 20th century with inventions like the [[telephone]], [[phonograph]], and [[radio]] that allowed for the transmission and storage of audio signals. Audio processing was necessary for early [[radio broadcasting]], as there were many problems with [[studio-to-transmitter link]]s.<ref>{{cite book|last=Atti|first=Andreas Spanias, Ted Painter, Venkatraman|title=Audio signal processing and coding|year=2006|publisher=John Wiley & Sons|location=Hoboken, NJ|isbn=0-471-79147-4|pages=464|url=https://books.google.com/books?id=Z_z-OQbadPIC|edition=[Online-Ausg.]}}</ref> The theory of signal processing and its application to audio was largely developed at [[Bell Labs]] in the mid 20th century. [[Claude Shannon]] and [[Harry Nyquist]]'s early work on [[communication theory]], [[Nyquist–Shannon sampling theorem|sampling theory]] and [[pulse-code modulation]] (PCM) laid the foundations for the field. In 1957, [[Max Mathews]] became the first person to [[Synthesizer|synthesize audio]] from a [[computer]], giving birth to [[computer music]].

Major developments in [[Digital audio|digital]] [[audio coding]] and [[audio data compression]] include [[differential pulse-code modulation]] (DPCM) by [[C. Chapin Cutler]] at Bell Labs in 1950,<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> [[linear predictive coding]] (LPC) by [[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 |archive-url=https://ghostarchive.org/archive/20221009/https://ee.stanford.edu/~gray/lpcip.pdf |archive-date=2022-10-09 |url-status=live |issn=1932-8346|doi-access=free }}</ref> [[adaptive DPCM]] (ADPCM) 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> [[discrete cosine transform]] (DCT) coding 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 |archive-url=https://ghostarchive.org/archive/20221009/https://www.ic.tu-berlin.de/fileadmin/fg121/Source-Coding_WS12/selected-readings/Ahmed_et_al.__1974.pdf |archive-date=2022-10-09 |url-status=live}}</ref> and [[modified discrete cosine transform]] (MDCT) coding by J. P. Princen, A. W. Johnson and A. B. Bradley at the [[University of Surrey]] 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> LPC is the basis for [[perceptual coding]] and is widely used in [[speech coding]],<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> while MDCT coding is widely used in modern [[audio coding 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 |archive-url=https://ghostarchive.org/archive/20221009/http://www.math.utah.edu/~gustafso/s2012/2270/web-projects/Guckert-audio-compression-svd-mdct-MP3.pdf |archive-date=2022-10-09 |url-status=live |website=[[University of Utah]] |date=Spring 2012 |access-date=14 July 2019}}</ref> 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>

== Types ==

=== Analog ===
{{further|Analog signal processing}}

An analog audio signal is a continuous signal represented by an electrical voltage or current that is ''analogous'' to the sound waves in the air. Analog signal processing then involves physically altering the continuous signal by changing the voltage or current or charge via [[electrical circuits]].

Historically, before the advent of widespread [[digital electronics|digital technology]], analog was the only method by which to manipulate a signal. Since that time, as computers and software have become more capable and affordable, digital signal processing has become the method of choice. However, in music applications, analog technology is often still desirable as it often produces [[Nonlinear system|nonlinear responses]] that are difficult to replicate with digital filters.

== Digital ==
{{further|Digital signal processing}}

A digital representation expresses the audio waveform as a sequence of symbols, usually [[binary numbers]].  This permits signal processing using [[digital circuits]] such as [[digital signal processor]]s, [[microprocessor]]s and general-purpose computers.  Most modern audio systems use a digital approach as the techniques of digital signal processing are much more powerful and efficient than analog domain signal processing.<ref>{{Cite book |title=Digital Audio Signal Processing |first=Udo |last=Zölzer |publisher=John Wiley and Sons |year=1997 |isbn=0-471-97226-6}}</ref>

== Applications ==

Processing methods and application areas include [[audio storage|storage]], [[audio data compression|data compression]], [[music information retrieval]], [[speech processing]], [[acoustic location|localization]], [[detection theory|acoustic detection]], [[transmission (telecom)|transmission]], [[noise cancellation]], [[acoustic fingerprint]]ing, [[sound recognition]], [[synthesizer|synthesis]], and enhancement (e.g. [[Equalization (audio)|equalization]], [[audio filter|filtering]], [[audio level compression|level compression]], [[echo]] and [[reverb]] removal or addition, etc.).

=== Audio broadcasting ===
{{see also | Broadcasting}}
Audio signal processing is used when broadcasting audio signals in order to enhance their fidelity or optimize for bandwidth or latency. In this domain, the most important audio processing takes place just before the transmitter. The audio processor here must prevent or minimize [[overmodulation]], compensate for non-linear transmitters (a potential issue with [[medium wave]] and [[shortwave]] broadcasting), and adjust overall [[loudness]] to the desired level.

=== Active noise control ===

[[Active noise control]] is a technique designed to reduce unwanted sound. By creating a signal that is identical to the unwanted noise but with the opposite polarity, the two signals cancel out due to [[destructive interference]].

=== Audio synthesis ===
{{see also|Synthesizer}}

Audio synthesis is the electronic generation of audio signals. A musical instrument that accomplishes this is called a synthesizer. Synthesizers can either [[Physical modelling synthesis|imitate sounds]] or generate new ones. Audio synthesis is also used to generate human [[speech]] using [[speech synthesis]].

===Audio effects===
{{Main|Effects unit}}
Audio effects alter the sound of a [[musical instrument]] or other audio source. Common effects include [[Distortion (music)|distortion]], often used with electric guitar in [[electric blues]] and [[rock music]]; [[Dynamics (music)|dynamic]] effects such as [[volume pedal]]s and [[Audio compressor|compressors]], which affect loudness; [[Linear filter|filters]] such as [[wah-wah pedal]]s and [[graphic equalizer]]s, which modify frequency ranges; [[modulation]] effects, such as [[Chorus effect|chorus]], [[flanger]]s and [[Phaser (effect)|phasers]]; [[Pitch (music)|pitch]] effects such as [[Pitch shifter (audio processor)|pitch shifters]]; and time effects, such as [[reverb]] and [[Delay (audio effect)|delay]], which create echoing sounds and emulate the sound of different spaces.

Musicians, [[audio engineer]]s and record producers use effects units during live performances or in the studio, typically with electric guitar, bass guitar, [[electronic keyboard]] or [[electric piano]]. While effects are most frequently used with [[Electric instrument|electric]] or [[Electronic musical instrument|electronic]] instruments, they can be used with any audio source, such as [[Acoustic music|acoustic]] instruments, drums, and vocals.<ref>{{Cite book|last1=Horne|first1=Greg|url=https://books.google.com/books?id=cHALQ_CO5P0C|title=Complete Acoustic Guitar Method: Mastering Acoustic Guitar c|publisher=Alfred Music|year=2000|isbn=9781457415043|page=92}}</ref><ref>{{Cite book|last1=Yakabuski|first1=Jim|url=https://books.google.com/books?id=QwcLdjCCXHkC|title=Professional Sound Reinforcement Techniques: Tips and Tricks of a Concert Sound Engineer|publisher=Hal Leonard|year=2001|isbn=9781931140065|page=139}}</ref>

===Computer audition===
{{excerpt|Computer audition}}

== See also ==
* [[Sound card]]
* [[Sound effect]]

== References ==
{{Reflist}}

== Further reading ==
*{{Cite book | last = Rocchesso | first = Davide | title = Introduction to Sound Processing | date = March 20, 2003| url = http://dsp-book.narod.ru/spv.pdf }}
*{{Cite journal |last1=Wilmering |first1=Thomas |last2=Moffat |first2=David |last3=Milo |first3=Alessia |last4=Sandler |first4=Mark B. |title=A History of Audio Effects |journal=Applied Sciences |date=2020 |volume=10 |issue=3 |page=791 |doi=10.3390/app10030791 |doi-access=free |hdl=10026.1/15335 |hdl-access=free }}

{{Audio broadcasting}}
{{Music production}}

{{DEFAULTSORT:Audio Signal Processing}}
[[Category:Audio electronics]]
[[Category:Signal processing]]