More commonly referred to as MP3, is a digital audio encoding format using a form of lossy data compression. It is a common audio format for consumer audio storage, as well as a de facto standard encoding for the transfer and playback of music on digital audio players. MP3 is an audio-specific format that was designed by the Moving Picture Experts Group. The group was formed by several teams of engineers at Fraunhofer IIS in Erlangen, Germany, AT&T-Bell Labs in Murray Hill, NJ, USA, Thomson-Brandt, and CCETT as well as others. It was approved as an ISO/IEC standard in 1991.
The use in MP3 of a lossy compression algorithm is designed to greatly reduce the amount of data required to represent the audio recording and still sound like a faithful reproduction of the original uncompressed audio for most listeners, but is not considered high fidelity audio by audiophiles. An MP3 file that is created using the mid-range bit rate setting of 128 kbit/s will result in a file that is typically about 1/10th the size of the CD file created from the original audio source. An MP3 file can also be constructed at higher or lower bit rates, with higher or lower resulting quality. The compression works by reducing accuracy of certain parts of sound that are deemed beyond the auditory resolution ability of most people. This method is commonly referred to as perceptual coding. It internally provides a representation of sound within a short term time/frequency analysis window, by using psychoacoustic models to discard or reduce precision of components less audible to human hearing, and recording the remaining information in an efficient manner. This is relatively similar to the principles used by JPEG, an image compression format.
MP3 “tag” in an mp3 audio file is a section of the file that contains metadata such as the title, artist, album, track number or other information about the file’s contents.

It is most commonly used to store digital audio streams, support AAC audio data, Audio-only MPEG-4 files generally have a .m4a extension. This is especially true of non-protected content.
MPEG-4 files with audio streams encrypted by FairPlay Digital Rights Management as sold through the iTunes Store use the .m4p extension.
Audio book and podcast files, which also contain metadata including chapter markers, images, and hyperlinks, can use the extension .m4a, but more commonly use the .m4b extension. An .m4a audio file cannot “bookmark” (remember the last listening spot), whereas .m4b extension files can.
The Apple iPhone uses MPEG-4 audio for its ringtones but uses the .m4r extension rather than the .m4a extension.

It is a standardized, lossy compression and encoding scheme for digital audio. Designed to be the successor of the MP3 format, AAC generally achieves better sound quality than MP3 at many bit rates.
AAC has been standardized by ISO and IEC, as part of the MPEG-2 & MPEG-4 specifications. The MPEG-2 standard contains several audio coding methods, including the MP3 coding scheme. AAC is able to include 48 full-bandwidth (up to 96 kHz) audio channels in one stream plus 16 low frequency enhancement (LFE, limited to 120 Hz) channels, up to 16 “coupling” or dialog channels, and up to 16 data streams. AAC is able to achieve good audio quality at data rates of 320 kbit/s for five channels. The quality for stereo is satisfactory to modest requirements at 96 kbit/s in joint stereo mode, however hi-fi transparency demands data rates of at least 128kbit/s (VBR), better than MP3. The MPEG-2 Audio tests showed that AAC meets the requirements referred to as “transparent” for the ITU at 128 kbit/s for stereo, and 320kbit/s for 5.1 audio.
AAC’s best known use is as the default audio format of Apple’s iPhone, iPod, iTunes, and the format used for all iTunes Store audio.
AAC is also the standard audio format for Sony’s PlayStation 3 and is supported by Sony’s Playstation Portable, latest generation of Sony Walkman, Walkman Phones from Sony Ericsson, Nseries Phones from Nokia, Nintendo’s Wii (with the Photo Channel 1.1 update installed for Wii consoles purchased before late 2007), the Nintendo DSi, and the MPEG-4 video standard.
High-Efficiency AAC is part of digital radio standards like DAB+ and Digital Radio Mondiale.
The current version of the AAC standard is ISO/IEC 14496-3:2005 (with 14496-3:2005/Amd.2. for HE-AAC v2)
AAC+ v2 is also standardized by ETSI (European Telecommunications Standards Institute) as TS 102005.
The MPEG-4 standard also contains other ways of compressing sound. These are low bit-rate and generally used for speech.

AAC was designed to fix many of the serious performance flaws in the MP3 format (which was specified in MPEG-1 and MPEG-2) by the ISO/IEC in 11172-3 and 13818-3. Advanced Audio Coding is designed to be the successor of the MP3 format and demonstrates greater sound quality and transparency than MP3 files coded at the same bit rate.
Improvements include:
More sample frequencies (from 8 kHz to 96 kHz) than MP3 (16 kHz to 48 kHz)
Up to 48 channels (MP3 supports up to two channels in MPEG-1 mode and up to 5.1 channels in MPEG-2 mode)
Arbitrary bit-rates and variable frame length. Standardized constant bit rate with bit reservoir.
Higher efficiency and simpler filterbank (rather than MP3’s hybrid coding, AAC uses a pure MDCT)
Higher coding efficiency for stationary signals (AAC uses a blocksize of 1024 samples, allowing more efficient coding than MP3’s 576 sample blocks)
Higher coding accuracy for transient signals (AAC uses a blocksize of 128 samples, allowing more accurate coding than MP3’s 192 sample blocks)
Can use Kaiser-Bessel derived window function to eliminate spectral leakage at the expense of widening the main lobe
Much better handling of audio frequencies above 16 kHz
More flexible joint stereo (different methods can be used in different frequency ranges)

Dolby Digital, or AC-3, is the common version containing up to six discrete channels of sound. The most elaborate mode in common usage involves five channels for normal-range speakers (20 Hz – 20,000 Hz) (right front, center, left front, right rear and left rear) and one channel (20 Hz – 120 Hz allotted audio) for the subwoofer driven low-frequency effects. Mono and stereo modes are also supported. AC-3 supports audio sample-rates up to 48 kHz. Batman Returns was the first film to use Dolby Digital technology when it premiered in theaters in Summer 1992. The Laserdisc version of Clear and Present Danger featured the first Home theater Dolby Digital mix in 1995.
AC3 is a 5.1 format, which means that it provides five full-bandwidth channels, front left, front right, center, surround left, and surround right. A low-frequency effect (LFE) channel is included for the sound needed for special effects and action sequences in movies. AC3 also has a downmixing feature that ensures compatibility with devices that do not support the 5.1 format.
AC3 file, a Dolby Digital audio file, can be found as the standard audio track on Digital Versatile Discs (DVD) and High Definition Television (HDTV). This coder has been designed to take maximum advantage of human auditory masking in that it divides the audio spectrum of each channel into narrow frequency bands of different sizes optimized with respect to the frequency selectivity of human hearing. This makes it possible to sharply filter coding noise so that it is forced to stay very close in frequency to the frequency components of the audio signal being coded. By reducing or eliminating coding noise wherever there are no audio signals to mask it, the sound quality of the original signal can be subjectively preserved.

It is a patented audio data compression scheme optimized for speech coding. AMR was adopted as the standard speech codec by 3GPP in October 1998 and is now widely used in GSM and UMTS. It uses link adaptation to select from one of eight different bit rates based on link conditions.
AMR is also a file format for storing spoken audio using the AMR codec. Many modern mobile telephone handsets will allow you to store short recordings in the AMR format, both Open Source and commercial programs exist to convert between this and other formats such as MP3, although it should be remembered that AMR is a speech format and is unlikely to give ideal results for other audio. The common filename extension is .amr.

It is an audio data compression technology developed by Microsoft. The name can be used to refer to its audio file format or its audio codecs. 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. And WMA Voice, targeted at voice content, applies compression using a range of low bit rates.
Windows Media Audio (WMA) is the most common codec of the four WMA codecs. Colloquial usage of the term WMA, especially in marketing materials and device specifications, usually refers to this codec only. The first version of the codec released in 1999 is regarded as WMA 1. In the same year, the bit stream syntax, or compression algorithm, was altered in minor ways and became WMA 2. Since then, newer versions of the codec were released, but the decoding process remained the same, ensuring compatibility between codec versions. WMA is a lossy audio codec based on the study of psychoacoustics. Audio signals that are deemed to be imperceptible to the human ear are encoded with reduced resolution during the compression process. WMA can encode audio signals sampled at up to 48000 times per second (48 kHz) with up to two discrete channels (stereo). WMA 9 introduced variable bit rate (VBR) and average bit rate (ABR) coding techniques into the MS encoder although both were technically supported by the original format. WMA 9.1 also added support for low-delay audio, which reduces latency for encoding and decoding.

WAV (or WAVE) is a Microsoft and IBM audio file format standard for storing an audio bitstream on PCs. It is an application of the RIFF bitstream format method for storing data in “chunks”, and thus also close to the IFF and the AIFF format used on Amiga and Macintosh computers, respectively. It is the main format used on Windows systems for raw and typically uncompressed audio. The usual bitstream encoding is the Pulse Code Modulation (PCM) format.
Though a WAV file can hold compressed audio, the most common WAV format contains uncompressed audio in the linear pulse code modulation (LPCM) format. The standard audio file format for CDs, for example, is LPCM-encoded, containing two channels of 44,100 samples per second, 16 bits per sample. Since LPCM uses an uncompressed, lossless storage method, which keeps all the samples of an audio track, professional users or audio experts may use the WAV format for maximum audio quality. WAV audio can also be edited and manipulated with relative ease using software. The WAV format supports compressed audio, using, on Windows, the Audio Compression Manager. Any ACM codec can be used to compress a WAV file. The UI for Audio Compression Manager is accessible by default through Sound Recorder.
Beginning with Windows 2000, a WAVE_FORMAT_EXTENSIBLE header was defined which specifies multiple audio channel data (surround sound) along with speaker positions, eliminates ambiguity regarding sample types and container sizes in the standard WAV format and supports defining custom extensions to the format chunk.
Uncompressed WAV files are quite large in size, so, as file sharing over the Internet has become popular, the WAV format has declined in popularity. However, it is still a commonly used, relatively “pure”, i.e. lossless, file type, suitable for retaining “first generation” archived files of high quality, or use on a system where high fidelity sound is required and disk space is not restricted.
More frequently, the smaller file sizes of compressed but lossy formats such as MP3, AAC, Ogg(Vorbis) and WMA are used to store and transfer audio. Their small file sizes allow faster Internet transmission, as well as lower consumption of space on memory media. However, lossy formats trade off smaller file size against loss of audio quality, as all such compression algorithms compromise available signal detail. There are also more efficient lossless codecs available, such as FLAC, Shorten, Monkey’s Audio, ATRAC Advanced Lossless, Apple Lossless, WMA Lossless, TTA, and WavPack, but none of these is yet a ubiquitous standard for both professional and home audio.
Audio CDs do not use WAV as their sound format, using instead Red Book audio. The commonality is that both audio CDs and WAV files have the audio data encoded in PCM. WAV is a data file format for a computer to use that can’t be understood by CD players directly. To record WAV files to an Audio CD the file headers must be stripped and the remaining PCM data written directly to the disc as individual tracks with zero padding added to match the CD’s sector size.
The WAV format is limited to files that are less than 4 GB in size, due to its use of a 32 bit unsigned integer to record the file size header (some programs limit the file size to 2-4 GB). Although this is equivalent to about 6.6 hours of CD-quality audio (44.1 kHz, 16-bit stereo), it is sometimes necessary to go over this limit, especially when higher sampling rates or bit resolutions are required.

The term ‘Ogg’ is commonly used to refer to audio file format Ogg Vorbis, that is, Vorbis-encoded audio in the Ogg container.
Vorbis is a free and open source, lossy audio codec project headed by the Xiph.Org Foundation and intended to serve as a replacement for MP3. It is most commonly used in conjunction with the Ogg container and is therefore called Ogg Vorbis.
The Vorbis format has proven popular among supporters of free software.[5] They argue that its higher fidelity and completely free nature, unencumbered by patents, make it a well-suited replacement for patented and restricted formats like MP3. However, MP3 has been widely used since the late-1990s and as of 2008, continues to remain popular in the consumer electronics industry.
For many applications, Vorbis has clear advantages over other lossy audio codecs in that it is patent-free and has free and open-source implementations and therefore is free to use, implement, or modify as one sees fit, yet produces smaller files than most other codecs at equivalent or higher quality.

Listening tests have attempted to find the best quality lossy audio codecs at certain bitrates. Some conclusions made by recent listening tests:
Low bitrate (less than 64 kbit/s): the most recent public multiformat test at 48 kbit/s shows that aoTuV Vorbis has a better quality than WMA and LC-AAC, the same quality as WMA Professional, and a lower quality than HE-AAC.
Mid to low bitrates (less than 128 kbit/s down to 64 kbit/s): private tests at 80 kbit/s and 96 kbit/s shows that aoTuV Vorbis has a better quality than other lossy audio codecs (LC-AAC, HE-AAC, MP3, MPC, WMA).
Mid bitrate (128kbit/s): most recent public multiformat test at 128 kbit/s shows a four-way tie between aoTuV Vorbis, LAME-encoded MP3, WMA Pro, and QuickTime AAC, with each codec essentially transparent (sounds identical to the original music file).
High bitrates (more than 128 kbit/s): most people do not hear significant differences. However, trained listeners can often hear significant differences between codecs at identical bitrates, and aoTuV Vorbis performs better than LC-AAC, MP3, and MPC.