Editing MP3 (section)

== 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,<ref name="cd-1991" /><ref name="neuron2-cd-1991" /> finalized in 1992,<ref name="dis-1992" /> and published in 1993 as ISO/IEC 11172-3:1993.<ref name="11172-3" /> 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.<ref name="13818-3" /><ref name="mpeg-audio-faq-bc" /> 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|Linear predictive coding|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.<ref name="Mayer1894" /> In 1959, Richard Ehmer described a complete set of auditory curves regarding this phenomenon.<ref name="Ehmer1959" /> Between 1967 and 1974, [[Eberhard Zwicker]] did work in the areas of tuning and masking of critical frequency-bands,<ref name="Zwicker" /><ref name="Eberhard" /> which in turn built on the fundamental research in the area from [[Harvey Fletcher]] and his collaborators at [[Bell Labs]].<ref name="Fletcher" />

Perceptual coding was first used for [[speech coding]] compression with [[linear predictive coding]] (LPC),<ref name="Schroeder2014">{{cite book |last1= Schroeder |first1= Manfred R. |title= Acoustics, Information, and Communication: Memorial Volume in Honor of Manfred R. Schroeder |date= 2014 |publisher= Springer |isbn= 978-3-319-05660-9 |chapter= Bell Laboratories |page= 388 |chapter-url= https://books.google.com/books?id=d9IkBAAAQBAJ&pg=PA388}}</ref> which has origins in the work of [[Fumitada Itakura]] ([[Nagoya University]]) and Shuzo Saito ([[Nippon Telegraph and Telephone]]) in 1966.<ref>{{cite journal |last1= Gray |first1= Robert M. |title= A History of Realtime Digital Speech on Packet Networks: Part II of Linear Predictive Coding and the Internet Protocol |journal= Found. Trends Signal Process. |date= 2010 |volume= 3 |issue= 4 |pages= 203–303 |doi= 10.1561/2000000036 |url= https://ee.stanford.edu/~gray/lpcip.pdf |issn= 1932-8346 |doi-access= free |access-date= 14 July 2019 |archive-date= 9 October 2022 |archive-url= https://ghostarchive.org/archive/20221009/https://ee.stanford.edu/~gray/lpcip.pdf |url-status= live }}</ref> 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.<ref name="Schroeder2014"/><ref>{{cite book |last1= Atal |first1= B. |last2= Schroeder |first2= M. |title= ICASSP '78. IEEE International Conference on Acoustics, Speech, and Signal Processing |chapter= Predictive coding of speech signals and subjective error criteria |date= 1978 |volume= 3 |pages= 573–576 |doi= 10.1109/ICASSP.1978.1170564}}</ref> Further optimization by Schroeder and Atal with J.L. Hall was later reported in a 1979 paper.<ref name="Schroeder1979"/> That same year, a psychoacoustic masking codec was also proposed by M. A. Krasner,<ref name="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<ref>{{cite web|last1= Krasner|first1= M. A.|title= Digital Encoding of Speech Based on the Perceptual Requirement of the Auditory System (Technical Report 535)|url= https://apps.dtic.mil/dtic/tr/fulltext/u2/a077355.pdf|ref= Lincoln Laboratory, MIT|date= 18 June 1979|url-status= live|archive-url= https://web.archive.org/web/20170903070321/https://www.dtic.mil/dtic/tr/fulltext/u2/a077355.pdf|archive-date= 3 September 2017}}</ref> 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 [[N. Ahmed|Nasir Ahmed]] in 1972, was developed by Ahmed with T. Natarajan and [[K. R. Rao]] in 1973; they published their results in 1974.<ref>{{cite journal |last= Ahmed |first= Nasir |author-link= N. Ahmed |title= How I Came Up With the Discrete Cosine Transform |journal= [[Digital Signal Processing (journal)|Digital Signal Processing]] |date= January 1991 |volume= 1 |issue= 1 |pages= 4–5 |doi= 10.1016/1051-2004(91)90086-Z |bibcode= 1991DSP.....1....4A |url= https://www.scribd.com/doc/52879771/DCT-History-How-I-Came-Up-with-the-Discrete-Cosine-Transform |access-date= 19 November 2019 |archive-date= 10 June 2016 |archive-url= https://web.archive.org/web/20160610013109/https://www.scribd.com/doc/52879771/DCT-History-How-I-Came-Up-with-the-Discrete-Cosine-Transform |url-status= live }}</ref><ref>{{Citation |first1= Nasir |last1= Ahmed |author1-link= N. Ahmed |first2= T. |last2= Natarajan |first3= K. R. |last3= Rao |title= Discrete Cosine Transform |journal= IEEE Transactions on Computers |date= January 1974 |volume= C-23 |issue= 1 |pages= 90–93 |doi= 10.1109/T-C.1974.223784|s2cid= 149806273 }}</ref><ref>{{Citation |last1= Rao |first1= K. R. |author-link1= K. R. Rao |last2= Yip |first2= P. |title= Discrete Cosine Transform: Algorithms, Advantages, Applications |publisher= Academic Press |location= Boston |year= 1990 |isbn= 978-0-12-580203-1}}</ref> 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,<ref>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</ref> following earlier work by Princen and Bradley in 1986.<ref>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</ref> The MDCT later became a core part of the MP3 algorithm.<ref name="Guckert">{{cite web |last1= Guckert |first1= John |title= The Use of FFT and MDCT in MP3 Audio Compression |url= http://www.math.utah.edu/~gustafso/s2012/2270/web-projects/Guckert-audio-compression-svd-mdct-MP3.pdf |website= [[University of Utah]] |date= Spring 2012 |access-date= 14 July 2019 |archive-date= 12 February 2021 |archive-url= https://web.archive.org/web/20210212022237/http://www.math.utah.edu/~gustafso/s2012/2270/web-projects/Guckert-audio-compression-svd-mdct-MP3.pdf |url-status= live }}</ref>

Ernst Terhardt and other collaborators constructed an algorithm describing auditory masking with high accuracy in 1982.<ref name="Terhardt1982" /> 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.<ref name="Schroeder2014"/> [[IEEE]]'s refereed ''Journal on Selected Areas in Communications'' reported on a wide variety of (mostly perceptual) audio compression algorithms in 1988.<ref name="Voice Coding for Communications" /> The "Voice Coding for Communications" edition published in February 1988 reported on a wide range of established, working audio bit compression technologies,<ref name="Voice Coding for Communications" /> 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,<ref name="musmann">Genesis of the MP3 Audio Coding Standard in IEEE Transactions on Consumer Electronics, IEEE, Vol. 52, Nr. 3, pp. 1043–1049, August 2006</ref> 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-CSF|Thomson-Brandt]]. The second group was [[MUSICAM]], by [[Panasonic|Matsushita]], [[Centre commun d'études de télévision et télécommunications|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 [[Nippon Telegraph and Telephone|NTT]] and BTRL.<ref name="musmann"/>

The immediate predecessors of MP3 were "Optimum Coding in the Frequency Domain" (OCF),<ref name="Brandenburg" /> and Perceptual Transform Coding (PXFM).<ref name="Johnston1988" /> 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]] [[Digital Signal Processor|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,<ref>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</ref> after having worked on MUSICAM with Matsushita and Philips since 1989.<ref name="musmann"/>

This codec incorporated into a broadcasting system using COFDM modulation was demonstrated on air and in the field<ref>{{cite web |title=A DAB commentary from Alan Box, EZ communication and chairman NAB DAB task force |url=https://www.americanradiohistory.com/Archive-BC/BC-1991/BC-1991-04-15.pdf}}</ref> 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&nbsp;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<ref>{{cite book | url = https://tech.ebu.ch/publications/sqamcd | title = EBU SQAM CD Sound Quality Assessment Material recordings for subjective tests | date = 2008-10-07 | access-date = 8 February 2017 | archive-date = 11 February 2017 | archive-url = https://web.archive.org/web/20170211162447/https://tech.ebu.ch/publications/sqamcd | url-status = live }}</ref> 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 (musical instrument)|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.<ref name="BusinessWeek_2007" /> 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 Society|Fraunhofer Institute for Integrated Circuits]], Erlangen (where he worked with [[Bernhard Grill]] and four other researchers – "The Original Six"<ref>{{Cite book|title=How Music Got Free: The End of an Industry, the Turn of the Century, and the Patient Zero of Piracy|last=Witt|first=Stephen|publisher=Penguin Books|year=2016|isbn=978-0-14-310934-1|location=United States of America|page=13|quote=Brandenburg and Grill were joined by four other Fraunhofer researchers. Heinz Gerhauser oversaw the institute´s audio research group; [[Harald Popp]] was a hardware specialist; Ernst Eberlein was a signal processing expert; Jurgen Herre was another graduate student whose mathematical prowess rivaled Brandenburg´s own. In later years this group would refer to themselves as "the original six".}}</ref>), 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 [[Fraunhofer Institute for Telecommunications|Heinrich Herz Institute]]. In 1993, he joined the staff of Fraunhofer HHI.<ref name="BusinessWeek_2007" /> 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.<ref name="Sterne2012_Vega" /> Accordingly, he dubbed Vega the "Mother of MP3".<ref name="motherofmp3" /> 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]] (<u>M</u>asking pattern adapted <u>U</u>niversal <u>S</u>ubband <u>I</u>ntegrated <u>C</u>oding <u>A</u>nd <u>M</u>ultiplexing) and ASPEC (<u>A</u>daptive <u>S</u>pectral <u>P</u>erceptual <u>E</u>ntropy <u>C</u>oding). The MUSICAM technique, proposed by [[Philips]] (Netherlands), [[Centre commun d'études de télévision et télécommunications|CCETT]] (France), the [[Institut für Rundfunktechnik|Institute for Broadcast Technology]] (Germany), and Matsushita (Japan),<ref>Digital Video and Audio Broadcasting Technology: A Practical Engineering Guide (Signals and Communication Technology) {{ISBN|3-540-76357-0}} p. 144: "In the year 1988, the MASCAM method was developed at the Institut für Rundfunktechnik (IRT) in Munich in preparation for the digital audio broadcasting (DAB) system. From MASCAM, the MUSICAM (masking pattern universal subband integrated coding and multiplexing) method was developed in 1989 in cooperation with CCETT, Philips and Matsushita."</ref> was chosen due to its simplicity and error robustness, as well as for its high level of computational efficiency.<ref name="santa-clara-1990" /> 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 (software)|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 [[Centre national d'études des télécommunications|CNET]].<ref name="Aspec" /> 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 {{nowrap|128 kbit/s}} as [[MPEG-1 Audio Layer II|MP2]] at {{nowrap|192 kbit/s}}.

The algorithms for MPEG-1 Audio Layer I, II and III were approved in 1991<ref name="cd-1991" /><ref name="neuron2-cd-1991" /> and finalized in 1992<ref name="dis-1992" /> 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.<ref name = "11172-3" /> 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 player]]s and decoders. Thus the first generation of MP3 defined {{math|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 [[audio bit depth|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&nbsp;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]].<ref>{{cite web |title=The MP3: A History Of Innovation And Betrayal |url=https://www.npr.org/sections/therecord/2011/03/23/134622940/the-mp3-a-history-of-innovation-and-betrayal |website=NPR |access-date=3 August 2023 |date=2011-03-23 |archive-date=3 August 2023 |archive-url=https://web.archive.org/web/20230803092021/https://www.npr.org/sections/therecord/2011/03/23/134622940/the-mp3-a-history-of-innovation-and-betrayal |url-status=live }}</ref> 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|date=August 2023}}

=== 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.<ref name="paris_press" /> It was approved as a draft technical report (DTR/DIS) in November 1994,<ref name="singapore_press" /> finalized in 1996 and published as international standard ISO/IEC TR 11172-5:1998 in 1998.<ref name="ISO/IEC TR 11172-5:1998" /> The [[Reference implementation (computing)|reference software]] in C language was later published as a freely available ISO standard.<ref name="Software_Simulation.zip" /> 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<ref>{{Cite book|title=A high-quality sound coding standard for broadcasting, telecommunications and multimedia systems.|last=Dehery |first=Yves-Francois|publisher=Elsevier Science BV |year=1994|isbn= 978-0-444-81580-4 |location=The Netherlands |pages=53–64|quote= This article refers to a Musicam (MPEG Audio Layer II) compressed digital audio workstation implemented on a microcomputer used not only as a professional editing station but also as a server on Ethernet for a compressed digital audio library, therefore anticipating the future MP3 on Internet }}</ref> were available for digital broadcasting (radio [[Digital audio broadcasting|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]].<ref name="MP3_Todays_Technology" /> The [[filename extension]] ''.mp3'' was chosen by the Fraunhofer team on 14 July 1995 (previously, the files had been named ''.bit'').<ref name="mp3-name" /> 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 drive]]s of the era (≈500–1000 [[megabyte|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 (music software)|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 ripper|CD ripping]] and digital music distribution as MP3 over the internet.<ref>{{cite web |url-status=live |url=https://www.theatlantic.com/magazine/archive/2000/09/the-heavenly-jukebox/305141/ |archive-url=https://web.archive.org/web/20130430043648/https://www.theatlantic.com/magazine/archive/2000/09/the-heavenly-jukebox/305141/ |archive-date=30 April 2013 |website= [[The Atlantic]] |quote=To show industries how to use the codec, MPEG cobbled together a free sample program that converted music into MP3 files. The demonstration software created poor-quality sound, and Fraunhofer did not intend that it be used. The software's "source code"—its underlying instructions—was stored on an easily accessible computer at the University of Erlangen, from which it was downloaded by one SoloH, a hacker in the Netherlands (and, one assumes, a Star Wars fan). SoloH revamped the source code to produce software that converted compact-disc tracks into music files of acceptable quality. |url-access=subscription |title=The Heavenly Jukebox |first1=Charles C. |last1=Mann  |date=September 2000 }}</ref><ref>''[https://books.google.com/books?id=3M2hAgAAQBAJ&dq=SoloH+mp3+source+code&pg=PT75 Pop Idols and Pirates: Mechanisms of Consumption and the Global Circulation of Popular Music]'' by Charles Fairchild. {{Webarchive|url=https://web.archive.org/web/20231015095150/https://books.google.com/books?id=3M2hAgAAQBAJ&dq=SoloH+mp3+source+code&pg=PT75 |date=15 October 2023 }}.</ref><ref>[http://ijoc.org/index.php/ijoc/article/viewFile/1765/989 Technologies of Piracy? - Exploring the Interplay  Between Commercialism and Idealism in the Development of MP3 and DivX] {{Webarchive|url=https://web.archive.org/web/20200919101723/https://ijoc.org/index.php/ijoc/article/viewFile/1765/989 |date=19 September 2020 }} by HENDRIK STORSTEIN SPILKER, SVEIN HÖIER, page 2072</ref><ref>[https://web.archive.org/web/20170103170919/http://www.euronet.nl/~soloh/mpegEnc/ www.euronet.nl/~soloh/mpegEnc/] ([[Archive.org]])</ref>

=== Further versions ===

Further work on MPEG audio<ref name="sydney1993" /> 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''<ref name="mpeg-audio-faq-bc" />), originally published in 1995.<ref name="13818-3" /><ref name="Brandenburg1997" /> 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.<ref name="sydney1993" /> 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 {{math|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.<ref name="MPEG-2.5" /><ref name="MPEG-2.5-2" /> 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.

{| class="wikitable sortable"
|+MPEG Audio Layer III versions
|-
! Version
! International Standard{{ref label|mp3standard|*|}}
! First edition public release date
! Latest edition public release date
|-
| MPEG-1 Audio Layer III
| [http://www.iso.org/iso/iso_catalogue/catalogue_tc/catalogue_detail.htm?csnumber=22412 ISO/IEC 11172-3] {{Webarchive|url=https://web.archive.org/web/20120528230220/http://www.iso.org/iso/iso_catalogue/catalogue_tc/catalogue_detail.htm?csnumber=22412 |date=28 May 2012 }} (MPEG-1 Part 3)<ref name="11172-3" /><ref name="neuron2-cd-1991" />
| 1993
|
|-
| MPEG-2 Audio Layer III
| [http://www.iso.org/iso/iso_catalogue/catalogue_ics/catalogue_detail_ics.htm?csnumber=26797 ISO/IEC 13818-3] {{Webarchive|url=https://web.archive.org/web/20110511043216/http://www.iso.org/iso/iso_catalogue/catalogue_ics/catalogue_detail_ics.htm?csnumber=26797 |date=11 May 2011 }} (MPEG-2 Part 3)<ref name="13818-3" /><ref name="mp3tech-iso13818-3" />
| 1995
| 1998
|-
| MPEG-2.5 Audio Layer III
| nonstandard, Fraunhofer proprietary<ref name="MPEG-2.5" /><ref name="MPEG-2.5-2" />
|2000
|2008
|}
{{refbegin}}
{{note label|mp3standard|*|}}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&nbsp;– ISO/IEC 13818-7).<ref name="mpeg-audio-faq-bc" />
{{refend}}

LAME is the most advanced MP3 encoder.{{Citation needed|date=August 2021|reason=Bold claims require verifiable citations}} 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&nbsp;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<ref>{{cite web | url = https://archive.org/details/iuma-archive&tab=about | title = About Internet Underground Music Archive }}</ref> 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.<ref>{{Cite web |last=Vainilavičius |first=Justinas |date=15 November 2023 |title=Winamp is back after revamp; nostalgia-inducing looks intact |url=https://cybernews.com/news/winamp-is-back-after-revamp-nostalgia-inducing-looks-intact/ |access-date=8 December 2023 |website=cybernews |archive-date=4 December 2023 |archive-url=https://web.archive.org/web/20231204111949/https://cybernews.com/news/winamp-is-back-after-revamp-nostalgia-inducing-looks-intact/ |url-status=live }}</ref> In 1998, [[Windows Media Player]] 5.2 and later added support for MP3 format. 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]].<ref name="seattlepi" />

In November 1997, the website [[mp3.com]] was offering thousands of MP3s created by independent artists for free.<ref name="seattlepi" /> The small size of MP3 files enabled widespread peer-to-peer [[file sharing]] of music [[Ripping|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.<ref name="Giesler" />

Unauthorized MP3 file sharing continues on next-generation [[peer-to-peer file sharing|peer-to-peer networks]]. Some authorized services, such as [[Beatport]], [[Bleep.com|Bleep]], [[Juno Records]], [[eMusic]], [[Zune Marketplace]], [[Walmart.com]], [[Rhapsody (online music service)|Rhapsody]], the recording industry approved re-incarnation of [[Napster (pay service)|Napster]], and [[Amazon.com]] sell unrestricted music in the MP3 format.