MP3 (formally MPEG-1 Audio Layer III or MPEG-2 Audio Layer III) is an audio coding format developed largely by the Fraunhofer Society in Germany under the lead of Karlheinz Brandenburg. It was designed to greatly reduce the amount of data required to represent audio, yet still sound like a faithful reproduction of the original uncompressed audio to most listeners; for example, compared to CD-quality digital audio, MP3 compression can commonly achieve a 75–95% reduction in size, depending on the bit rate. In popular usage, MP3 often refers to files of sound or music recordings stored in the MP3 file format (.mp3) on consumer electronic devices.
MPEG-1 Audio Layer III was originally defined in 1991 as one of the three possible audio codecs of the MPEG-1 standard (along with MPEG-1 Audio Layer I and MPEG-1 Audio Layer II). All three options were retained and further extended—defining additional bit rates and support for more audio channels (supporting surround sound—in the subsequent MPEG-2 standard).
MP3 as a file format commonly designates files containing an elementary stream of MPEG-1 Audio or MPEG-2 Audio encoded data. Concerning audio compression, which is its most apparent element to end-users, MP3 uses lossy compression to reduce precision of encoded data and to partially discard data, allowing for a large reduction in file sizes when compared to uncompressed audio.
The combination of small size and acceptable fidelity led to a boom in the distribution of music over the Internet in the late 1990s, with MP3 serving as an enabling technology at a time when bandwidth and storage were still at a premium. The MP3 format soon became associated with controversies surrounding copyright infringement, music piracy, and the file-ripping and sharing services MP3.com and Napster, among others. With the advent of portable media players (including MP3 players), a product category also including smartphones, MP3 support became near-universal and it remains a de facto standard for digital audio despite the creation of newer coding formats such as AAC and FLAC.
History The Moving Picture Experts Group (MPEG) designed MP3 as part of its MPEG-1, and later MPEG-2, standards. MPEG-1 Audio (MPEG-1 Part 3), which included MPEG-1 Audio Layer I, II, and III, was approved as a committee draft for an ISO/IEC standard in 1991, finalized in 1992, and published in 1993 as ISO/IEC 11172-3:1993. An MPEG-2 Audio (MPEG-2 Part 3) extension with lower sample and bit rates was published in 1995 as ISO/IEC 13818-3:1995. It requires only minimal modifications to existing MPEG-1 decoders (recognition of the MPEG-2 bit in the header and addition of the new lower sample and bit rates).
Background
The MP3 lossy compression algorithm takes advantage of a perceptual limitation of human hearing called auditory masking. In 1894, the American physicist Alfred M. Mayer reported that a tone could be rendered inaudible by another tone of lower frequency. In 1959, Richard Ehmer described a complete set of auditory curves regarding this phenomenon. Between 1967 and 1974, Eberhard Zwicker did work in the areas of tuning and masking of critical frequency-bands, which in turn built on the fundamental research in the area from Harvey Fletcher and his collaborators at Bell Labs.
Perceptual coding was first used for speech coding compression with linear predictive coding (LPC), which has origins in the work of Fumitada Itakura (Nagoya University) and Shuzo Saito (Nippon Telegraph and Telephone) in 1966. In 1978, Bishnu S. Atal and Manfred R. Schroeder at Bell Labs proposed an LPC speech codec, called adaptive predictive coding, that used a psychoacoustic coding-algorithm exploiting the masking properties of the human ear. Further optimization by Schroeder and Atal with J.L. Hall was later reported in a 1979 paper. That same year, a psychoacoustic masking codec was also proposed by M. A. Krasner, who published and produced hardware for speech (not usable as music bit-compression), but the publication of his results in a relatively obscure Lincoln Laboratory Technical Report did not immediately influence the mainstream of psychoacoustic codec-development.
The discrete cosine transform (DCT), a type of transform coding for lossy compression, proposed by Nasir Ahmed in 1972, was developed by Ahmed with T. Natarajan and K. R. Rao in 1973; they published their results in 1974. This led to the development of the modified discrete cosine transform (MDCT), proposed by J. P. Princen, A. W. Johnson and A. B. Bradley in 1987,J. P. Princen, A. W. Johnson und A. B. Bradley: Subband/transform coding using filter bank designs based on time domain aliasing cancellation, IEEE Proc. Intl. Conference on Acoustics, Speech, and Signal Processing (ICASSP), 2161–2164, 1987 following earlier work by Princen and Bradley in 1986.John P. Princen, Alan B. Bradley: Analysis/synthesis filter bank design based on time domain aliasing cancellation, IEEE Trans. Acoust. Speech Signal Processing, ASSP-34 (5), 1153–1161, 1986 The MDCT later became a core part of the MP3 algorithm.
Ernst Terhardt and other collaborators constructed an algorithm describing auditory masking with high accuracy in 1982. 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.
In 1985, Atal and Schroeder presented code-excited linear prediction (CELP), an LPC-based perceptual speech-coding algorithm with auditory masking that achieved a significant data compression ratio for its time. IEEE's refereed Journal on Selected Areas in Communications reported on a wide variety of (mostly perceptual) audio compression algorithms in 1988. The "Voice Coding for Communications" edition published in February 1988 reported 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.
Development The genesis of the MP3 technology is fully described in a paper from Professor Hans Musmann,Genesis of the MP3 Audio Coding Standard in IEEE Transactions on Consumer Electronics, IEEE, Vol. 52, Nr. 3, pp. 1043–1049, August 2006 who chaired the ISO MPEG Audio group for several years. In December 1988, MPEG called for an audio coding standard. In June 1989, 14 audio coding algorithms were submitted. Because of certain similarities between these coding proposals, they were clustered into four development groups. The first group was ASPEC, by Fraunhofer Gesellschaft, AT&T, CNET(France Telecom) and Thomson. The second group was MUSICAM, by Matsushita, CCETT, ITT and Philips. The third group was ATAC (ATRAC Coding), by Fujitsu, JVC, NEC and Sony. And the fourth group was SB-ADPCM, by NTT and BTRL.
The immediate predecessors of MP3 were Optimum Coding in the Frequency Domain (OCF), and Perceptual Transform Coding (PXFM). These two codecs, along with block-switching contributions from Thomson-Brandt, were merged into a codec named ASPEC, which was submitted to MPEG and won the quality competition, but which was 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 digital signal processor (DSP) chips.
Another predecessor of the MP3 format and technology was the perceptual codec MUSICAM based on an integer arithmetic 32-sub-band filter bank, driven by a psychoacoustic model. It was primarily designed for Digital Audio Broadcasting (digital radio) and digital TV, and its basic principles were disclosed to the scientific community by CCETT (France) and IRT (Germany) in Atlanta during an IEEE-ICASSP conference in 1991,Y.F. Dehery, et al. (1991) A MUSICAM source codec for Digital Audio Broadcasting and storage Proceedings IEEE-ICASSP 91 pages 3605–3608 May 1991 after having worked on MUSICAM with Matsushita and Philips since 1989.
This codec, incorporated into a broadcasting system using COFDM modulation, was demonstrated on air and in the field with Radio Canada and CRC Canada during the NAB show (Las Vegas) in 1991. The implementation of the audio part of this broadcasting system was based on a two-chip encoder (one for the subband transform, one for the psychoacoustic model designed by the team of G. Stoll (IRT Germany), later known as psychoacoustic model I) and a real-time decoder using one Motorola 56001 DSP chip running an integer arithmetics software designed by Y.F. Dehery's team (CCETT, France). The simplicity of the corresponding decoder together with the high audio quality of this codec using for the first time a 48 kHz sampling rate, a 20 bits/sample input format (the highest available sampling standard in 1991, compatible with the AES/EBU professional digital input studio standard) were the main reasons to later adopt the characteristics of MUSICAM as the basic features for an advanced digital music compression codec.
During the development of the MUSICAM encoding software, Stoll and Dehery's team made thorough use of a set of high-quality audio assessment material selected by a group of audio professionals from the European Broadcasting Union, and later used as a reference for the assessment of music compression codecs. The subband coding technique was found to be efficient, not only for the perceptual coding of high-quality sound materials but especially for the encoding of critical percussive sound materials (drums, triangle,...), due to the specific temporal masking effect of the MUSICAM sub-band filterbank (this advantage being a specific feature of short transform coding techniques).
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. MP3 is directly descended from OCF and PXFM, representing the outcome of the collaboration of Brandenburg — working as a postdoctoral researcher at AT&T-Bell Labs with James D. Johnston ("JJ") of AT&T-Bell Labs — with the Fraunhofer Institute for Integrated Circuits, Erlangen (where he worked with Bernhard Grill and four other researchers – "The Original Six"), 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's Heinrich Herz Institute. In 1993, he joined the staff of Fraunhofer IIS in Erlangen. An acapella version of the song "Tom's Diner" by Suzanne Vega was the first song used by Brandenburg to develop the MP3 format. It was used as a benchmark to see how well MP3's compression algorithm handled the human voice. Brandenburg adopted the song for testing purposes, listening to it again and again each time he refined the compression algorithm, making sure it did not adversely affect the reproduction of Vega's voice. Accordingly, he dubbed Vega the "Mother of MP3".
Standardization