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MP3
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Not to be confused with MPEG-3.
For other uses, see MP3 (disambiguation).
MPEG-1 or MPEG-2 Audio Layer III Filename extension     .mp3[1]
Internet media type     audio/mpeg,[2] audio/MPA,[3] audio/mpa-robust[4]
Initial release     1995[5]
Type of format     Audio compression format, audio file format
Standard(s)     ISO/IEC 11172-3,[5]
ISO/IEC 13818-3[6]

MPEG-1 or MPEG-2 Audio Layer III,[4] more commonly referred to as MP3, is a patented encoding format for digital audio which uses a form of lossy data compression. It is a common audio format for consumer audio storage, as well as a de facto standard of digital audio compression for the transfer and playback of music on most digital audio players.

MP3 is an audio-specific format that was designed by the Moving Picture Experts Group (MPEG) as part of its MPEG-1 standard and later extended in MPEG-2 standard. The first MPEG subgroup – Audio group was formed by several teams of engineers at Fraunhofer IIS, University of Hannover, AT&T-Bell Labs, Thomson-Brandt, CCETT, and others.[7] MPEG-1 Audio (MPEG-1 Part 3), which included MPEG-1 Audio Layer I, II and III was approved as a committee draft of ISO/IEC standard in 1991,[8][9] finalised in 1992[10] and published in 1993 (ISO/IEC 11172-3:1993[5]). Backwards compatible MPEG-2 Audio (MPEG-2 Part 3) with additional bit rates and sample rates was published in 1995 (ISO/IEC 13818-3:1995).[6][11]

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. An MP3 file that is created using the setting of 128 kbit/s will result in a file that is about 1/11 the size[note 1] 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 considered to be beyond the auditory resolution ability of most people. This method is commonly referred to as perceptual coding.[13] It uses psychoacoustic models to discard or reduce precision of components less audible to human hearing, and then records the remaining information in an efficient manner.
Contents

Development

The MP3 lossy audio data compression algorithm takes advantage of a perceptual limitation of human hearing called auditory masking. In 1894, the American physicist Alfred Marshall Mayer reported that a tone could be rendered inaudible by another tone of lower frequency.[14] In 1959, Richard Ehmer described a complete set of auditory curves regarding this phenomenon.[15] Ernst Terhardt et al. created an algorithm describing auditory masking with high accuracy.[16] This work added to a variety of reports from authors dating back to Fletcher, and to the work that initially determined critical ratios and critical bandwidths.

The psychoacoustic masking codec was first proposed in 1979, apparently independently, by Manfred R. Schroeder, et al.[17] from AT&T-Bell Labs in Murray Hill, NJ, and M. A. Krasner[18] both in the United States. Krasner was the first to publish and to produce hardware for speech (not usable as music bit compression), but the publication of his results as a relatively obscure Lincoln Laboratory Technical Report did not immediately influence the mainstream of psychoacoustic codec development. Manfred Schroeder was already a well-known and revered figure in the worldwide community of acoustical and electrical engineers, but his paper was not much noticed, since it described negative results due to the particular nature of speech and the linear predictive coding (LPC) gain present in speech. Both Krasner and Schroeder built upon the work performed by Eberhard F. Zwicker in the areas of tuning and masking of critical bands,[19][20] that in turn built on the fundamental research in the area from Bell Labs of Harvey Fletcher and his collaborators.[21] A wide variety of (mostly perceptual) audio compression algorithms were reported in IEEE's refereed Journal on Selected Areas in Communications.[22] That journal reported in February 1988 on a wide range of established, working audio bit compression technologies, some of them using auditory masking as part of their fundamental design, and several showing real-time hardware implementations.

The immediate predecessors of MP3 were "Optimum Coding in the Frequency Domain" (OCF),[23] and Perceptual Transform Coding (PXFM).[24] These two codecs, along with block-switching contributions from Thomson-Brandt, were merged into a codec called ASPEC, which was submitted to MPEG, and which won the quality competition, but that was mistakenly rejected as too complex to implement. The first practical implementation of an audio perceptual coder (OCF) in hardware (Krasner's hardware was too cumbersome and slow for practical use), was an implementation of a psychoacoustic transform coder based on Motorola 56000 DSP chips.

As a doctoral student at Germany's University of Erlangen-Nuremberg, Karlheinz Brandenburg began working on digital music compression in the early 1980s, focusing on how people perceive music. He completed his doctoral work in 1989.[25] MP3 is directly descended from OCF and PXFM, representing the outcome of the collaboration of Brandenburg - working as a postdoc at AT&T-Bell Labs with James D. (JJ) Johnston of AT&T-Bell Labs - with the Fraunhofer Institut for Integrated Circuits, Erlangen, with relatively minor contributions from the MP2 branch of psychoacoustic sub-band coders. In 1990, Brandenburg became an assistant professor at Erlangen-Nuremberg. While there, he continued to work on music compression with scientists at the Fraunhofer Society (in 1993 he joined the staff of the Fraunhofer Institute).[25]

The song "Tom's Diner" by Suzanne Vega was the first song used by Karlheinz Brandenburg to develop the MP3. Brandenburg adopted the song for testing purposes, listening to it again and again each time refining the scheme, making sure it did not adversely affect the subtlety of Vega's voice.
Standardization

In 1991, there were only two proposals available that could be completely assessed for an MPEG audio standard: Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (the Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio.[26] The Musicam format, based on sub-band coding, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame).

Much of its technology and ideas were incorporated into the definition of ISO MPEG Audio Layer I and Layer II and the filter bank alone into Layer III (MP3) format as part of the computationally inefficient hybrid filter bank. Under the chairmanship of Professor Musmann (University of Hannover) the editing of the standard was made under the responsibilities of Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II).

ASPEC was the joint proposal of AT&T Bell Laboratories, Thomson Consumer Electronics, Fraunhofer Society and CNET.[27] It provided the highest coding efficiency.

A working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.

All algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991[8][9] and finalized in 1992[10] as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3 (a.k.a. MPEG-1 Audio or MPEG-1 Part 3), published in 1993.[5]

Further work on MPEG audio[28] was finalized in 1994 as part of the second suite of MPEG standards, MPEG-2, more formally known as international standard ISO/IEC 13818-3 (a.k.a. MPEG-2 Part 3 or backwards compatible MPEG-2 Audio or MPEG-2 Audio BC[11]), originally published in 1995.[6][29] MPEG-2 Part 3 (ISO/IEC 13818-3) defined additional bit rates and sample rates for MPEG-1 Audio Layer I, II and III. The new sampling rates are exactly half that of those originally defined in MPEG-1 Audio. This reduction in sampling rate serves to cut the available frequency fidelity in half while likewise cutting the bitrate by 50%. MPEG-2 Part 3 also enhanced MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel.[28]

An additional extension to MPEG-2 is named MPEG-2.5 audio, as MPEG-3 already had a different meaning. This extension was developed at Fraunhofer IIS, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling rates exactly half of that previously possible with MPEG-2. It thus widens the scope of MP3 to include human speech and other applications requiring only 25% of the frequency reproduction possible with MPEG-1. While not an ISO recognized standard, MPEG-2.5 is widely supported by both inexpensive and brand name digital audio players as well as computer software based MP3 encoders and decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further down. [30][31] MPEG-2.5 was not developed by MPEG and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format.

Mp3 Songs