MP3

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, 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. 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, following earlier work by Princen and Bradley in 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. 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. after having worked on MUSICAM with Matsushita and Philips since 1989. 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.), 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. was chosen due to its simplicity and error robustness, as well as for its high level of computational efficiency. 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. 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. 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. Further versions } 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). 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. In 1998, Windows Media Player 5.2 and later added support for MP3 format, and 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. In November 1997, the website mp3.com was offering thousands of MP3s created by independent artists for free. 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. Napster later returned as a legitimate music streaming service. Unauthorized MP3 file sharing continues on next-generation peer-to-peer networks. Authorized services, such as Amazon.com, Beatport, Bleep, eMusic, Juno Records, and the reincarnated Napster, sell unrestricted music in the MP3 format. == Design ==