An audio coding format(or sometimes audio compression format) is a content representation format for storage or transmission 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.
Audio content encoded in a particular audio coding format is normally encapsulated within a container format. As such, the user normally doesn't have a raw AAC file, but instead has a .m4a audio file, which is a MPEG-4 Part 14 container containing AAC-encoded audio. The container also contains metadata such as title and other tags, and perhaps an index for fast seeking.A notable exception is MP3 files, which are raw audio coding without a container format. De facto standards for adding metadata tags such as title and artist to MP3s, such as ID3, are hacks which work by appending the tags to the MP3, and then relying on the MP3 player to recognize the chunk as malformed audio coding and therefore skip it. In video files with audio, the encoded audio content is bundled with video (in a video coding format) inside a multimedia container format.
An audio coding format does not dictate all algorithms used by a codec implementing the format. An important part of how lossy audio compression works is by removing data in ways humans can't hear, according to a psychoacoustic model; the implementer of an encoder has some freedom of choice in which data to remove (according to their psychoacoustic model).
A lossless audio coding format reduces the total data needed to represent a sound but can be de-coded to its original, uncompressed form. A lossy audio coding format additionally reduces the bit resolution of the sound on top of compression, which results in far less data at the cost of irretrievably lost information.
Consumer audio is most often compressed using lossy audio codecs as the smaller size is far more convenient for distribution. The most widely used audio coding formats are MP3 and Advanced Audio Coding (AAC), both of which are lossy formats based on modified discrete cosine transform (MDCT) and perceptual coding algorithms.
Lossless audio coding formats such as FLAC and Apple Lossless are sometimes available, though at the cost of larger files.
Uncompressed audio formats, such as pulse-code modulation (PCM, or .wav), are also sometimes used. PCM was the standard format for Compact Disc Digital Audio (CDDA), before lossy compression eventually became the standard after the introduction of MP3.
In 1950, Bell Labs filed the patent on differential pulse-code modulation (DPCM).Adaptive DPCM (ADPCM) was introduced by P. Cummiskey, Nikil S. Jayant and James L. Flanagan at Bell Labs in 1973.
Perceptual coding was first used for speech coding compression, with linear predictive coding (LPC).Initial concepts for LPC date back to the work of Fumitada Itakura (Nagoya University) and Shuzo Saito (Nippon Telegraph and Telephone) in 1966. During the 1970s, Bishnu S. Atal and Manfred R. Schroeder at Bell Labs developed a form of LPC called adaptive predictive coding (APC), a perceptual coding algorithm that exploited the masking properties of the human ear, followed in the early 1980s with the code-excited linear prediction (CELP) algorithm which achieved a significant compression ratio for its time. Perceptual coding is used by modern audio compression formats such as MP3 and AAC.
Discrete cosine transform (DCT), developed by Nasir Ahmed, T. Natarajan and K. R. Rao in 1974,provided the basis for the modified discrete cosine transform (MDCT) used by modern audio compression formats such as MP3 and AAC. MDCT was proposed by J. P. Princen, A. W. Johnson and A. B. Bradley in 1987, following earlier work by Princen and Bradley in 1986. The MDCT is used by modern audio compression formats such as Dolby Digital, MP3, and Advanced Audio Coding (AAC).
|Basic compression algorithm||Audio coding standard||Abbreviation||Introduction||Market share (2019)||Ref|
|Modified discrete cosine transform (MDCT)||Dolby Digital (AC-3)||AC3||1991||58%|
|Adaptive Transform Acoustic Coding||ATRAC||1992||Unknown|
|MPEG Layer III||MP3||1993||49%|
|Advanced Audio Coding (MPEG-2 / MPEG-4)||AAC||1997||88%|
|Windows Media Audio||WMA||1999||Unknown|
|Constrained Energy Lapped Transform||CELT||2011||N/A|
|Adaptive differential pulse-code modulation (ADPCM)||aptX / aptX-HD||aptX||1989||Unknown|
|Digital Theater Systems||DTS||1990||14%|
|Master Quality Authenticated||MQA||2014||Unknown|
|Sub-band coding (SBC)||MPEG-1 Audio Layer II||MP2||1993||Unknown|
An audio file format is a file format for storing digital audio data on a computer system. The bit layout of the audio data is called the audio coding format and can be uncompressed, or compressed to reduce the file size, often using lossy compression. The data can be a raw bitstream in an audio coding format, but it is usually embedded in a container format or an audio data format with defined storage layer.
A codec is a device or computer program which encodes or decodes a digital data stream or signal. Codec is a portmanteau of coder-decoder.
In signal processing, data compression, source coding, or bit-rate reduction is the process of encoding information using fewer bits than the original representation. Any particular compression is either lossy or lossless. Lossless compression reduces bits by identifying and eliminating statistical redundancy. No information is lost in lossless compression. Lossy compression reduces bits by removing unnecessary or less important information. Typically, a device that performs data compression is referred to as an encoder, and one that performs the reversal of the process (decompression) as a decoder.
In information technology, lossy compression or irreversible compression is the class of data encoding methods that uses inexact approximations and partial data discarding to represent the content. These techniques are used to reduce data size for storing, handling, and transmitting content. The different versions of the photo of the cat to the right show how higher degrees of approximation create coarser images as more details are removed. This is opposed to lossless data compression which does not degrade the data. The amount of data reduction possible using lossy compression is much higher than through lossless techniques.
MP3 is a coding format for digital audio. Originally defined as the third audio format of the MPEG-1 standard, it was retained and further extended—defining additional bit-rates and support for more audio channels—as the third audio format of the subsequent MPEG-2 standard. A third version, known as MPEG 2.5—extended to better support lower bit rates—is commonly implemented, but is not a recognized standard.
Speech coding is an application of data compression of digital audio signals containing speech. Speech coding uses speech-specific parameter estimation using audio signal processing techniques to model the speech signal, combined with generic data compression algorithms to represent the resulting modeled parameters in a compact bitstream.
Windows Media Audio (WMA) is a series of audio codecs and their corresponding audio coding formats developed by Microsoft. It is a proprietary technology that forms part of the Windows Media framework. WMA consists of four distinct codecs. The original WMA codec, known simply as WMA, was conceived as a competitor to the popular MP3 and RealAudio codecs. WMA Pro, a newer and more advanced codec, supports multichannel and high resolution audio. A lossless codec, WMA Lossless, compresses audio data without loss of audio fidelity. WMA Voice, targeted at voice content, applies compression using a range of low bit rates. Microsoft has also developed a digital container format called Advanced Systems Format to store audio encoded by WMA.
Transform coding is a type of data compression for "natural" data like audio signals or photographic images. The transformation is typically lossless on its own but is used to enable better quantization, which then results in a lower quality copy of the original input.
Digital audio is a representation of sound recorded in, or converted into, digital form. In digital audio, the sound wave of the audio signal is encoded as numerical samples in a continuous sequence. For example, in CD audio, samples are taken 44100 times per second each with 16 bit sample depth. Digital audio is also the name for the entire technology of sound recording and reproduction using audio signals that have been encoded in digital form. Following significant advances in digital audio technology during the 1970s, it gradually replaced analog audio technology in many areas of audio engineering and telecommunications in the 1990s and 2000s.
An audio codec is a codec that encodes or decodes audio. In software, an audio codec is a computer program implementing an algorithm that compresses and decompresses digital audio data according to a given audio file or streaming media audio coding format. The objective of the algorithm is to represent the high-fidelity audio signal with minimum number of bits while retaining quality. This can effectively reduce the storage space and the bandwidth required for transmission of the stored audio file. Most software codecs are implemented as libraries which interface to one or more multimedia players. Most modern audio compression algorithms are based on modified discrete cosine transform (MDCT) coding and linear predictive coding (LPC).
Dolby Digital, also known as Dolby AC-3, is the name for audio compression technologies developed by Dolby Laboratories. Originally named Dolby Stereo Digital until 1995, except for Dolby TrueHD, the audio compression is lossy, based on the modified discrete cosine transform (MDCT) algorithm. The first use of Dolby Digital was to provide digital sound in cinemas from 35mm film prints; today, it is now also used for other applications such as TV broadcast, radio broadcast via satellite, digital video streaming, DVDs, Blu-ray discs and game consoles.
Advanced Audio Coding (AAC) is an audio coding standard for lossy digital audio compression. Designed to be the successor of the MP3 format, AAC generally achieves higher sound quality than MP3 at the same bit rate.
The modified discrete cosine transform (MDCT) is a lapped transform based on the type-IV discrete cosine transform (DCT-IV), with the additional property of being lapped: it is designed to be performed on consecutive blocks of a larger dataset, where subsequent blocks are overlapped so that the last half of one block coincides with the first half of the next block. This overlapping, in addition to the energy-compaction qualities of the DCT, makes the MDCT especially attractive for signal compression applications, since it helps to avoid artifacts stemming from the block boundaries. As a result of these advantages, the MDCT is the most widely used lossy compression technique in audio data compression. It is employed in most modern audio coding standards, including MP3, Dolby Digital (AC-3), Vorbis (Ogg), Windows Media Audio (WMA), ATRAC, Cook, Advanced Audio Coding (AAC), High-Definition Coding (HDC), LDAC, Dolby AC-4, and MPEG-H 3D Audio, as well as speech coding standards such as AAC-LD (LD-MDCT), G.722.1, G.729.1, CELT, and Opus.
Apple Lossless, also known as Apple Lossless Audio Codec (ALAC), or Apple Lossless Encoder (ALE), is an audio coding format, and its reference audio codec implementation, developed by Apple Inc. for lossless data compression of digital music. After initially keeping it proprietary from its inception in 2004, in late 2011 Apple made the codec available open source and royalty-free. Traditionally, Apple has referred to the codec as Apple Lossless, though more recently it has begun to use the abbreviated term ALAC when referring to the codec.
Pre-echo, sometimes called a forward echo, is a digital audio compression artifact where a sound is heard before it occurs. It is most noticeable in impulsive sounds from percussion instruments such as castanets or cymbals.
Constrained Energy Lapped Transform (CELT) is an open, royalty-free lossy audio compression format and a free software codec with especially low algorithmic delay for use in low-latency audio communication. The algorithms are openly documented and may be used free of software patent restrictions. Development of the format was maintained by the Xiph.Org Foundation and later coordinated by the Opus working group of the Internet Engineering Task Force (IETF).
A video coding format is a content representation format for storage or transmission of digital video content. It typically uses a standardized video compression algorithm, most commonly based on discrete cosine transform (DCT) coding and motion compensation. Examples of video coding formats include H.262, MPEG-4 Part 2, H.264, HEVC (H.265), Theora, RealVideo RV40, VP9, and AV1. A specific software or hardware implementation capable of compression or decompression to/from a specific video coding format is called a video codec; an example of a video codec is Xvid, which is one of several different codecs which implements encoding and decoding videos in the MPEG-4 Part 2 video coding format in software.