Audio coding standard in the context of Vorbis


Audio coding standard in the context of Vorbis

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⭐ Core Definition: Audio coding standard

An audio coding format, or audio compression format, is a encoded format of digital audio, such as in digital television, digital radio and in audio and video files. Examples of audio coding formats include MP3, AAC, Vorbis, FLAC, and Opus. A specific software or hardware implementation capable of audio compression and decompression to/from a specific audio coding format is called an audio codec; an example of an audio codec is LAME, which is one of several different codecs which implements encoding and decoding audio in the MP3 audio coding format in software.

Some audio coding formats are documented by a detailed technical specification document known as an audio coding specification. Some such specifications are written and approved by standardization organizations as technical standards, and are thus known as an audio coding standard. The term "standard" is also sometimes used for de facto standards as well as formal standards.

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Audio coding standard in the context of Dolby Digital

Dolby Digital, originally synonymous with Dolby AC-3 (see below), is the name for a family of audio compression technologies developed by Dolby Laboratories. Called Dolby Stereo Digital until 1995, it uses lossy compression (except for Dolby TrueHD). The first use of Dolby Digital was to provide digital sound in cinemas from 35 mm film prints. It has since also been used for TV broadcast, radio broadcast via satellite, digital video streaming, DVDs, Blu-ray discs and game consoles.

Dolby AC-3 was the original version of the Dolby Digital codec. The basis of the Dolby AC-3 multi-channel audio coding standard is the modified discrete cosine transform (MDCT), a lossy audio compression algorithm. It is a modification of the discrete cosine transform (DCT) algorithm, which was proposed by Nasir Ahmed in 1972 for image compression. The DCT was adapted into the MDCT by J.P. Princen, A.W. Johnson and Alan B. Bradley at the University of Surrey in 1987.

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Audio coding standard in the context of Advanced Audio Coding

Advanced Audio Coding (AAC) is an audio coding standard for lossy digital audio compression. It was developed by Dolby, AT&T, Fraunhofer and Sony, originally as part of the MPEG-2 specification but later improved under MPEG-4. AAC was designed to be the successor of the MP3 format (MPEG-2 Audio Layer III) and generally achieves higher sound quality than MP3 at the same bit rate. AAC encoded audio files are typically packaged in an MP4 container most commonly using the filename extension .m4a.

The basic profile of AAC (both MPEG-4 and MPEG-2) is called AAC-LC (Low Complexity). It is widely supported in the industry and has been adopted as the default or standard audio format on products including Apple's iTunes Store, Nintendo's Wii, DSi and 3DS and Sony's PlayStation 3. It is also further supported on various other devices and software such as iPhone, iPod, PlayStation Portable and Vita, PlayStation 5, Android and older cell phones, digital audio players like Sony Walkman and SanDisk Clip, media players such as VLC, Winamp and Windows Media Player, various in-dash car audio systems, and is used on Spotify, Google Nest, Amazon Alexa. Apple Music, YouTube and also YouTube Music streaming services. AAC has been further extended into HE-AAC (High Efficiency, or AAC+), which improves efficiency over AAC-LC. Another variant is AAC-LD (Low Delay).

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