Editing Digital audio (section)

==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>