Editing Digital audio (section)

===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>