MP3 (formally MPEG-1 Audio Layer III or MPEG-2 Audio Layer III)[4] is a coding format for digital audio developed largely by the Fraunhofer Society in Germany under the lead of Karlheinz Brandenburg.[11][12] 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.[13] In popular usage, MP3 often refers to files of sound or music recordings stored in the MP3 file format (.mp3) on consumer electronic devices.
Originally defined in 1991 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. MP3 as a file format commonly designates files containing an elementary stream of MPEG-1 Audio or MPEG-2 Audio encoded data, without other complexities of the MP3 standard. Concerning audio compression, which is its most apparent element to end-users, MP3 uses lossy compression to encode data using inexact approximations and the partial discarding of 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 remains near-universal and a de facto standard for digital audio.
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,[14][15] finalized in 1992,[16] and published in 1993 as ISO/IEC 11172-3:1993.[7] 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.[8][17] 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
Further information: Linear predictive coding and Modified discrete cosine transform
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.[18] In 1959, Richard Ehmer described a complete set of auditory curves regarding this phenomenon.[19] Between 1967 and 1974, Eberhard Zwicker did work in the areas of tuning and masking of critical frequency-bands,[20][21] which in turn built on the fundamental research in the area from Harvey Fletcher and his collaborators at Bell Labs.[22]
Perceptual coding was first used for speech coding compression with linear predictive coding (LPC),[23] which has origins in the work of Fumitada Itakura (Nagoya University) and Shuzo Saito (Nippon Telegraph and Telephone) in 1966.[24] 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.[23][25] Further optimization by Schroeder and Atal with J.L. Hall was later reported in a 1979 paper.[26] That same year, a psychoacoustic masking codec was also proposed by M. A. Krasner,[27] 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[28] 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.[29][30][31] 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,[32] following earlier work by Princen and Bradley in 1986.[33] The MDCT later became a core part of the MP3 algorithm.[34]
Ernst Terhardt and other collaborators constructed an algorithm describing auditory masking with high accuracy in 1982.[35] 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.[23] IEEE‘s refereed Journal on Selected Areas in Communications reported on a wide variety of (mostly perceptual) audio compression algorithms in 1988.[36] The “Voice Coding for Communications” edition published in February 1988 reported on a wide range of established, working audio bit compression technologies,[36] 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,[37] 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, France Telecom, Deutsche and Thomson-Brandt. 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.[37]
The immediate predecessors of MP3 were “Optimum Coding in the Frequency Domain” (OCF),[38] and Perceptual Transform Coding (PXFM).[39] 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.
Another predecessor of the MP3 format and technology is to be found in the perceptual codec MUSICAM based on an integer arithmetics 32 sub-bands 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,[40] after having worked on MUSICAM with Matsushita and Philips since 1989.[37]
This codec incorporated into a broadcasting system using COFDM modulation was demonstrated on air and in the field[41] 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[42] 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.[43] 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”[44]), 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 HHI.[43] 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.[45] Accordingly, he dubbed Vega the “Mother of MP3”.[46] Instrumental music had been easier to compress, but Vega’s voice sounded unnatural in early versions of the format. Brandenburg eventually met Vega and heard Tom’s Diner performed live.
Standardization
In 1991, two available proposals were 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, proposed by Philips (Netherlands), CCETT (France), the Institute for Broadcast Technology (Germany), and Matsushita (Japan),[47] was chosen due to its simplicity and error robustness, as well as for its high level of computational efficiency.[48] The MUSICAM format, based on sub-band coding, became the basis for the MPEG Audio compression format, incorporating, for example, its frame structure, header format, sample rates, etc.
While much of MUSICAM technology and ideas were incorporated into the definition of MPEG Audio Layer I and Layer II, the filter bank alone and the data structure based on 1152 samples framing (file format and byte-oriented stream) of MUSICAM remained in the Layer III (MP3) format, as part of the computationally inefficient hybrid filter bank. Under the chairmanship of Professor Musmann of the Leibniz University Hannover, the editing of the standard was delegated to Leon van de Kerkhof (Netherlands), Gerhard Stoll (Germany), and Yves-François Dehery (France), who worked on Layer I and Layer II. ASPEC was the joint proposal of AT&T Bell Laboratories, Thomson Consumer Electronics, Fraunhofer Society, and CNET.[49] It provided the highest coding efficiency.
A working group consisting of van de Kerkhof, Stoll, Leonardo Chiariglione (CSELT VP for Media), Yves-François Dehery, Karlheinz Brandenburg (Germany) and James D. Johnston (United States) took ideas from ASPEC, integrated the filter bank from Layer II, added some of their ideas such as the joint stereo coding of MUSICAM and created the MP3 format, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.
The algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991[14][15] and finalized in 1992[16] 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.[7] Files or data streams conforming to this standard must handle sample rates of 48k, 44100, and 32k and continue to be supported by current MP3 players and decoders. Thus the first generation of MP3 defined 14 × 3 = 42 interpretations of MP3 frame data structures and size layouts.
The compression efficiency of encoders is typically defined by the bit rate because the compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. They may use the compact disc (CD) parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2×16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with the use of the term compression ratio for lossy encoders.
Karlheinz Brandenburg used a CD recording of Suzanne Vega‘s song “Tom’s Diner” to assess and refine the MP3 compression algorithm.[50] This song was chosen because of its nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts (glockenspiel, triangle, accordion, etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats.[citation needed]
Going public
A reference simulation software implementation, written in the C language and later known as ISO 11172-5, was developed (in 1991–1996) by the members of the ISO MPEG Audio committee to produce bit-compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). It was approved as a committee draft of the ISO/IEC technical report in March 1994 and printed as document CD 11172-5 in April 1994.[51] It was approved as a draft technical report (DTR/DIS) in November 1994,[52] finalized in 1996 and published as international standard ISO/IEC TR 11172-5:1998 in 1998.[53] The reference software in C language was later published as a freely available ISO standard.[54] Working in non-real time on several operating systems, it was able to demonstrate the first real-time hardware decoding (DSP based) of compressed audio. Some other real-time implementations of MPEG Audio encoders and decoders[55] were available for digital broadcasting (radio DAB, television DVB) towards consumer receivers and set-top boxes.
On 7 July 1994, the Fraunhofer Society released the first software MP3 encoder, called l3enc.[56] The filename extension .mp3 was chosen by the Fraunhofer team on 14 July 1995 (previously, the files had been named .bit).[1] With the first real-time software MP3 player WinPlay3 (released 9 September 1995) many people were able to encode and play back MP3 files on their PCs. Because of the relatively small hard drives of the era (≈500–1000 MB) lossy compression was essential to store multiple albums’ worth of music on a home computer as full recordings (as opposed to MIDI notation, or tracker files which combined notation with short recordings of instruments playing single notes).
Fraunhofer example implementation
A hacker named SoloH discovered the source code of the “dist10” MPEG reference implementation shortly after the release on the servers of the University of Erlangen. He developed a higher-quality version and spread it on the internet. This code started the widespread CD ripping and digital music distribution as MP3 over the internet.[57][58][59][60]
Further versions
Further work on MPEG audio[61] 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 backward compatible MPEG-2 Audio or MPEG-2 Audio BC[17]), originally published in 1995.[8][62] MPEG-2 Part 3 (ISO/IEC 13818-3) defined 42 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 rates serves to cut the available frequency fidelity in half while likewise cutting the bit rate 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.[61] An MP3 coded with MPEG-2 results in half of the bandwidth reproduction of MPEG-1 appropriate for piano and singing.
A third generation of “MP3” style data streams (files) extended the MPEG-2 ideas and implementation but was named MPEG-2.5 audio since MPEG-3 already had a different meaning. This extension was developed at Fraunhofer IIS, the registered patent holder of MP3, by reducing the frame sync field in the MP3 header from 12 to 11 bits. As in the transition from MPEG-1 to MPEG-2, MPEG-2.5 adds additional sampling rates exactly half of those available using MPEG-2. It thus widens the scope of MP3 to include human speech and other applications yet requires only 25% of the bandwidth (frequency reproduction) possible using MPEG-1 sampling rates. While not an ISO-recognized standard, MPEG-2.5 is widely supported by both inexpensive Chinese and brand-name digital audio players as well as computer software-based MP3 encoders (LAME), decoders (FFmpeg) and players (MPC) adding 3 × 8 = 24 additional MP3 frame types. Each generation of MP3 thus supports 3 sampling rates exactly half that of the previous generation for a total of 9 varieties of MP3 format files. The sample rate comparison table between MPEG-1, 2, and 2.5 is given later in the article.[63][64] MPEG-2.5 is supported by LAME (since 2000), Media Player Classic (MPC), iTunes, and FFmpeg.
MPEG-2.5 was not developed by MPEG (see above) and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to the MP3 format. It is nonetheless ubiquitous and especially advantageous for low-bit-rate human speech applications.
| Version | International Standard[*] | First edition public release date | Latest edition public release date |
|---|---|---|---|
| MPEG-1 Audio Layer III | ISO/IEC 11172-3 Archived 28 May 2012 at the Wayback Machine (MPEG-1 Part 3)[7][15] | 1993 | |
| MPEG-2 Audio Layer III | ISO/IEC 13818-3 Archived 11 May 2011 at the Wayback Machine (MPEG-2 Part 3)[8][65] | 1995 | 1998 |
| MPEG-2.5 Audio Layer III | nonstandard, Fraunhofer proprietary[63][64] | 2000 | 2008 |
* The ISO standard ISO/IEC 11172-3 (a.k.a. MPEG-1 Audio) defined three formats: the MPEG-1 Audio Layer I, Layer II and Layer III. The ISO standard ISO/IEC 13818-3 (a.k.a. MPEG-2 Audio) defined an extended version of MPEG-1 Audio: MPEG-2 Audio Layer I, Layer II, and Layer III. MPEG-2 Audio (MPEG-2 Part 3) should not be confused with MPEG-2 AAC (MPEG-2 Part 7 – ISO/IEC 13818-7).[17]
LAME is the most advanced MP3 encoder.[citation needed] LAME includes a variable bit rate (VBR) encoding which uses a quality parameter rather than a bit rate goal. Later versions (2008+) support an n.nnn quality goal which automatically selects MPEG-2 or MPEG-2.5 sampling rates as appropriate for human speech recordings that need only 5512 Hz bandwidth resolution.
Internet distribution
In the second half of the 1990s, MP3 files began to spread on the Internet, often via underground pirated song networks. The first known experiment in Internet distribution was organized in the early 1990s by the Internet Underground Music Archive, better known by the acronym IUMA. After some experiments[66] using uncompressed audio files, this archive started to deliver on the native worldwide low-speed Internet some compressed MPEG Audio files using the MP2 (Layer II) format and later on used MP3 files when the standard was fully completed. The popularity of MP3s began to rise rapidly with the advent of Nullsoft‘s audio player Winamp, released in 1997, which still had in 2023 a community of 80 million active users.[67] In 1998, the first portable solid-state digital audio player MPMan, developed by SaeHan Information Systems, which is headquartered in Seoul, South Korea, was released and the Rio PMP300 was sold afterward in 1998, despite legal suppression efforts by the RIAA.[68]
In November 1997, the website mp3.com was offering thousands of MP3s created by independent artists for free.[68] The small size of MP3 files enabled widespread peer-to-peer file sharing of music ripped from CDs, which would have previously been nearly impossible. The first large peer-to-peer filesharing network, Napster, was launched in 1999. The ease of creating and sharing MP3s resulted in widespread copyright infringement. Major record companies argued that this free sharing of music reduced sales, and called it “music piracy“. They reacted by pursuing lawsuits against Napster, which was eventually shut down and later sold, and against individual users who engaged in file sharing.[69]
Unauthorized MP3 file sharing continues on next-generation peer-to-peer networks. Some authorized services, such as Beatport, Bleep, Juno Records, eMusic, Zune Marketplace, Walmart.com, Rhapsody, the recording industry approved re-incarnation of Napster, and Amazon.com sell unrestricted music in the MP3 format.
