The term audio codec has two meanings, both referring to something that encodes and decodes. The term codec is a combination of 'coder-decoder'.

In software, a codec is a computer program Computer programs are instructions for a computer. A computer requires programs to function, typically executing the program's instructions in a central processor. The program has an executable form that the computer can use directly to execute the instructions. The same program in its human-readable source code form, from which executable that compresses/decompresses In computer science and information theory, data compression or source coding is the process of encoding information using fewer bits than an unencoded representation would use through use of specific encoding schemes digital audio Digital audio uses digital signals for sound reproduction. This includes analog-to-digital conversion, digital-to-analog conversion, storage, and transmission. In effect, the system commonly referred to as digital is in fact a discrete-time, discrete-level analog of a previous electrical analog. While modern systems can be quite subtle in their data according to a given audio file format An audio file format is a file format for storing audio data on a computer system. It can be a raw bitstream, but it is usually a container format or a data format with defined storage layer or streaming audio format However, computer networks were still limited, and media was usually delivered over non-streaming channels, such as by downloading a digital file from a remote web server and then saving it to a local drive on the end user's computer or storing it as a digital file and playing it back from CD-ROMs. The object of a codec algorithm In mathematics, computing, linguistics, and related subjects, an algorithm is a finite sequence of instructions, logic, an explicit, step-by-step procedure for solving a problem, often used for calculation and data processing and many other fields. It is formally a type of effective method in which a list of well-defined instructions for is to represent the high-fidelity audio signal with minimum number of bits while retaining the quality. This can effectively reduce the storage space and the bandwidth required for transmission of the stored audio file. Most codecs are implemented as libraries which interface to one or more multimedia players, such as QuickTime Player QuickTime is a multimedia framework developed by Apple Inc., capable of handling various formats of digital video, media clips, sound, text, animation, music, and interactive panoramic images. It is available for Mac OS , Mac OS X and Microsoft Windows operating systems. The latest version is 7.6, XMMS The X Multimedia System is a free software audio player very similar to Winamp, that runs on many Unix-like operating systems, Winamp Winamp is a proprietary media player written by Nullsoft, now a subsidiary of AOL. It is skinnable, multi-format freeware/shareware, VLC media player VLC media player is an open source, free software media player written by the VideoLAN project, MPlayer or Windows Media Player Windows Media Player is a digital media player and media library application developed by Microsoft that is used for playing audio, video and viewing images on personal computers running the Microsoft Windows operating system, as well as on Pocket PC and Windows Mobile-based devices. Editions of Windows Media Player were also released for Mac OS,.

In hardware Hardware is a general term that refers to the physical artifacts of a technology. It may also mean the physical components of a computer system, in the form of computer hardware, the term "audio codec" refers to a single device that encodes analog audio as digital signals and vice versa. This is used in sound cards A sound card is a computer expansion card that facilitates the input and output of audio signals to and from a computer under control of computer programs. Typical uses of sound cards include providing the audio component for multimedia applications such as music composition, editing video or audio, presentation, education, and entertainment ( that support both audio in and out, for instance.

See also

Data compression In computer science and information theory, data compression or source coding is the process of encoding information using fewer bits than an unencoded representation would use through use of specific encoding schemes methods
Lossless Lossless data compression is a class of data compression algorithms that allows the exact original data to be reconstructed from the compressed data. The term lossless is in contrast to lossy data compression, which only allows an approximation of the original data to be reconstructed, in exchange for better compression rates
Theory Information theory is a branch of applied mathematics and electrical engineering involving the quantification of information. Historically, information theory was developed by Claude E. Shannon to find fundamental limits on compressing and reliably storing and communicating data. Since its inception it has broadened to find applications in many Entropy In information theory, entropy is a measure of the uncertainty associated with a random variable. The term by itself in this context usually refers to the Shannon entropy, which quantifies, in the sense of an expected value, the information contained in a message, usually in units such as bits. Equivalently, the Shannon entropy is a measure of the · Complexity In algorithmic information theory , the Kolmogorov complexity (also known as descriptive complexity, Kolmogorov-Chaitin complexity, stochastic complexity, algorithmic entropy, or program-size complexity) of an object such as a piece of text is a measure of the computational resources needed to specify the object. For example, consider the · Redundancy Redundancy in information theory is the number of bits used to transmit a message minus the number of bits of actual information in the message. Informally, it is the amount of wasted "space" used to transmit certain data. Data compression is a way to reduce or eliminate unwanted redundancy, while checksums are a way of adding desired
Entropy encoding In information theory an entropy encoding is a lossless data compression scheme that is independent of the specific characteristics of the medium Shannon-Fano · Huffman In computer science and information theory, Huffman coding is an entropy encoding algorithm used for lossless data compression. The term refers to the use of a variable-length code table for encoding a source symbol where the variable-length code table has been derived in a particular way based on the estimated probability of occurrence for each · Adaptive Huffman · Arithmetic Arithmetic coding is a method for lossless data compression. Normally, a string of characters such as the words "hello there" is represented using a fixed number of bits per character, as in the ASCII code. Like Huffman coding, arithmetic coding is a form of variable-length entropy encoding that converts a string into another form that · Range · Golomb Golomb coding is a lossless data compression method using a family of data compression codes invented by Solomon W. Golomb in the 1960s. Alphabets following a geometric distribution will have a Golomb code as an optimal prefix code, making Golomb coding highly suitable for situations in which the occurrence of small values in the input stream is · Exp-Golomb · Universal (Elias Elias gamma code is a universal code encoding positive integers developed by Peter Elias. It is used most commonly when coding integers whose upper-bound cannot be determined beforehand · Fibonacci In mathematics, Fibonacci coding is a universal code which encodes positive integers into binary code words. All tokens end with "11" and have no "11" before the end)
Dictionary A dictionary coder, also sometimes known as a substitution coder, is a class of lossless data compression algorithms which operate by searching for matches between the text to be compressed and a set of strings contained in a data structure maintained by the encoder. When the encoder finds such a match, it substitutes a reference to the string's RLE Run-length encoding is a very simple form of data compression in which runs of data (that is, sequences in which the same data value occurs in many consecutive data elements) are stored as a single data value and count, rather than as the original run. This is most useful on data that contains many such runs: for example, relatively simple graphic · Byte pair encoding · DEFLATE Deflate is a lossless data compression algorithm that uses a combination of the LZ77 algorithm and Huffman coding. It was originally defined by Phil Katz for version 2 of his PKZIP archiving tool, and was later specified in RFC 1951 · LZ family (LZ77/78 · LZSS Lempel-Ziv-Storer-Szymanski is a lossless data compression algorithm, a derivative of LZ77, that was created in 1982 by James Storer and Thomas Szymanski. LZSS was described in article "Data compression via textual substitution" published in Journal of the ACM (pp. 928-951) · LZW · LZWL · LZO · LZMA The Lempel-Ziv-Markov chain-Algorithm is an algorithm used to perform data compression. It has been under development since 1998 and is used in the 7z format of the 7-Zip archiver. This algorithm uses a dictionary compression scheme somewhat similar to LZ77 and features a high compression ratio (generally higher than bzip2 ) and a variable · LZX · LZRW · LZJB LZJB is the name for the lossless data compression algorithm invented by Jeff Bonwick to compress crash dumps and data in ZFS. It includes a number of improvements to the LZRW1 algorithm, a member of the Lempel-Ziv family of compression algorithms · LZT)
Others CTW · BWT · PPM · DMC Dynamic Markov compression is a lossless data compression algorithm developed by Gordon Cormack and Nigel Horspool . It uses predictive arithmetic coding similar to prediction by partial matching (PPM), except that the input is predicted one bit at a time (rather than one byte at a time). DMC has a good compression ratio and moderate speed,
Audio Audio compression is a form of data compression designed to reduce the size of audio files. Audio compression algorithms are implemented in computer software as audio codecs. Generic data compression algorithms perform poorly with audio data, seldom reducing file sizes much below 87% of the original, and are not designed for use in real time
Theory Acoustics is the interdisciplinary science that deals with the study of sound, ultrasound and infrasound . A scientist who works in the field of acoustics is an acoustician. The application of acoustics in technology is called acoustical engineering. There is often much overlap and interaction between the interests of acousticians and acoustical Companding · Convolution · Dynamic range Dynamic range is a term used frequently in numerous fields to describe the ratio between the smallest and largest possible values of a changeable quantity, such as in sound and light · Latency · Sampling In signal processing, sampling is the reduction of a continuous signal to a discrete signal. A common example is the conversion of a sound wave to a sequence of samples (a discrete-time signal) · Nyquist–Shannon theorem · Sound quality Sound quality can be defined as the degree of accuracy with which a device records or emits the original sound waves. For digital recording/digital playback, this accuracy depends on the range of sound which is sampled, the rate at which it is sampled, and the various conversions that occur in any sound reproduction system. With lossy codecs such
Audio codec parts LPC Linear predictive coding is a tool used mostly in audio signal processing and speech processing for representing the spectral envelope of a digital signal of speech in compressed form, using the information of a linear predictive model. It is one of the most powerful speech analysis techniques, and one of the most useful methods for encoding good (LAR · LSP) · WLPC Warped Linear Predictive Coding is a variant of Linear predictive coding in which the spectral representation of the system is modified, for example by replacing the unit delays used in an LPC implementation with first-order allpass filters. This can have advantages in reducing the bitrate required for a given level of perceived audio quality/ · CELP · ACELP The ACELP method is widely employed in current speech coding standards such as AMR, EFR, AMR-WB and ITU-T G-series standards G.729, G.729.1 and G.723.1 · A-law An A-law algorithm is a standard companding algorithm, used in European digital communications systems to optimize, i.e., modify, the dynamic range of an analog signal for digitizing · μ-law The µ-law algorithm is a companding algorithm, primarily used in the digital telecommunication systems of North America and Japan. Companding algorithms reduce the dynamic range of an audio signal. In analog systems, this can increase the signal-to-noise ratio (SNR) achieved during transmission, and in the digital domain, it can reduce the · ADPCM Adaptive DPCM is a variant of DPCM (differential pulse-code modulation) that varies the size of the quantization step, to allow further reduction of the required bandwidth for a given signal-to-noise ratio · DPCM DPCM or differential pulse-code modulation is a signal encoder that uses the baseline of PCM but adds some functionalities based on the prediction of the samples of the signal. The input can be an analog signal or a digital signal · MDCT · Fourier transform In mathematics, the Fourier transform is an operation that transforms one complex-valued function of a real variable into another. In such applications as signal processing, the domain of the original function is typically time and is accordingly called the time domain. That of the new function is frequency, and so the Fourier transform is often · Psychoacoustic model
Others Bit rate In telecommunications and computing, bitrate is the number of bits that are conveyed or processed per unit of time (CBR · ABR Average bitrate refers to the average amount of data transferred per unit of time, usually measured per second. This is commonly referred to for digital music or video. An MP3 file, for example, that has an average bit rate of 128 kbit/s transfers, on average, 128,000 bits every second. It can have higher bitrate and lower bitrate parts, and the · VBR) · Speech compression · Sub-band coding
Image
Terms Color space · Pixel · Chroma subsampling · Compression artifact · Image resolution
Methods RLE · Fractal · Wavelet · EZW · SPIHT · LP · DCT · Chain code · KLT
Others Test images · PSNR quality measure · Quantization
Video
Terms Video Characteristics · Frame · Frame rate · Interlace · Frame types · Video quality · Video resolution
Video codec parts Motion compensation · DCT · Quantization
Others Video codecs · Rate distortion theory · Bit rate (CBR · ABR · VBR)
Timeline of information theory, data compression, and error-correcting codes
See for formats and for codecs
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'Audio Codec' Answers
how do i know which audio codec i need to install for a file that will not play?
Q. i have downloaded a video file and my windows media player as well as real player both say that the correct audio codec isnt installed on my system. how do i know which which to install??
Asked by katiangel - Sun Mar 5 08:06:40 2006 - - 2 Answers - 0 Comments

A. You can install a standalone such as VLC or mplayer. Or, you can grab a codec pack such as
Answered by will.platnick - Sun Mar 5 08:44:15 2006

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