Editing MP3 (section)

=== 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.