Audio coding in the context of Digital audio


Audio coding in the context of Digital audio

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⭐ Core Definition: Audio coding

An audio coding format (or sometimes 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 in the context of Digital signal processing

Digital signal processing (DSP) is the use of digital processing, such as by computers or more specialized digital signal processors, to perform a wide variety of signal processing operations. The digital signals processed in this manner are a sequence of numbers that represent samples of a continuous variable in a domain such as time, space, or frequency. In digital electronics, a digital signal is represented as a pulse train, which is typically generated by the switching of a transistor.

Digital signal processing and analog signal processing are subfields of signal processing. DSP applications include audio and speech processing, sonar, radar and other sensor array processing, spectral density estimation, statistical signal processing, digital image processing, data compression, video coding, audio coding, image compression, signal processing for telecommunications, control systems, biomedical engineering, and seismology, among others.

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Audio coding in the context of Karlheinz Brandenburg

Karlheinz Brandenburg (born 20 June 1954) is a German electrical engineer and mathematician. Together with Ernst Eberlein, Heinz Gerhäuser (former Institutes Director of Fraunhofer IIS), Bernhard Grill, Jürgen Herre and Harald Popp (all Fraunhofer IIS), he developed the widespread MP3 method for audio data compression. He is also known for his elementary work in the field of audio coding, perception measurement, wave field synthesis and psychoacoustics. Brandenburg has received numerous national and international research awards, prizes and honors for his work. Since 2000 he has been a professor of electronic media technology at the Technical University Ilmenau. Brandenburg was significantly involved in the founding of the Fraunhofer Institute for Digital Media Technology (IDMT) and currently serves as its director.

Brandenburg has been called the "father of the MP3" format.

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Audio coding in the context of AAC-LD

The MPEG-4 Low Delay Audio Coder (a.k.a. AAC Low Delay, or AAC-LD) is audio compression standard designed to combine the advantages of perceptual audio coding with the low delay necessary for two-way communication. It is closely derived from the MPEG-2 Advanced Audio Coding (AAC) standard. It was published in MPEG-4 Audio Version 2 (ISO/IEC 14496-3:1999/Amd 1:2000) and in its later revisions.

AAC-LD uses a version of the modified discrete cosine transform (MDCT) audio coding technique called the LD-MDCT. AAC-LD is widely used by Apple as the voice-over-IP (VoIP) speech codec in FaceTime.

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