Editing MP3 (section)

=== 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}}