วันอาทิตย์ที่ 6 กรกฎาคม พ.ศ. 2551



The MP3 audio data compression algorithm takes advantage of a perceptual limitation of human hearing called auditory masking. In 1894, Mayer reported that a tone could be rendered inaudible by another tone of lower frequency.[2] In 1959, Richard Ehmer described a complete set of auditory curves regarding this phenomena.[3] Ernst Terhardt et al. created an algorithm describing auditory masking with high accuracy.[4] In 1983, at the University of Buenos Aires, Oscar Bonello started developing a PC audio card based on bit compression technology. In 1989 he introduced the first working device using auditory masking: Audicom.[5] In 1992 and 1994, MPEG-1 and MPEG-2 audio standards were completed.[6]
The psychoacoustic masking codec was first proposed in 1979, apparently independently, by Manfred Schroeder, et al..[7] Received 8 June 1979; accepted for publication 13 August 1979 from AT&T-Bell Labs in Murray Hill, NJ, and M. A.Krasner[8] both in the United States. Krasner was the first to publish and to produce hardware, but the publication of his results as a relatively obscure Lincoln Laboratory Technical Report did not immediately influence the mainstream of psychoacoustic codec development. Manfred Schroeder was already a well-known and revered figure in the worldwide community of acoustical and electrical engineers, and his paper had influence in acoustic and source-coding (audio data compression) research. Both Krasner and Schroeder built upon the work performed by Eberhard F. Zwicker in the areas of tuning and masking of critical bands,[9][10] that in turn built on the fundamental research in the area from Bell Labs of Harvey Fletcher and his collaborators.[11] A wide variety of (mostly perceptual) audio compression algorithms were reported in IEEE's refereed Journal on Selected Areas in Communications.[12][page # needed] That journal reported in February 1988 on a wide range of established, working audio bit compression technologies, most of them using auditory masking as part of their fundamental design, and several showing real-time hardware implementations.
The immediate predecessors of MP3 were "Optimum Coding in the Frequency Domain" (OCF),
[13] and Perceptual Transform Coding (PXFM).[14] 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. MP3 is directly descended from OCF and PXFM. MP3 represents the outcome of the collaboration of Dr. Karlheinz Brandenburg, working as a postdoc at AT&T-Bell Labs with Mr. James D. Johnston of AT&T-Bell Labs, collaborating with the Fraunhofer Society for Integrated Circuits, Erlangen, with relatively minor contributions from the MP2 branch of psychoacoustic sub-band coders.
MPEG-1 Audio Layer 2 encoding began as the Digital Audio Broadcast (DAB) project managed by Egon Meier-Engelen of the Deutsche Forschungs- und Versuchsanstalt für Luft- und Raumfahrt (later on called Deutsches Zentrum für Luft- und Raumfahrt, German Aerospace Center) in Germany. The European Community financed this project, commonly known as EU-147, from 1987 to 1994 as a part of the EUREKA research program.
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 and became an assistant professor at Erlangen-Nuremberg. While there, he continued to work on music compression with scientists at the Fraunhofer Society (in 1993 he joined the staff of the Fraunhofer Institute).[15]
In 1991 there were two proposals available: Musicam and ASPEC - (Short excerpt on German Wikipedia) (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (The Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio.[16] The Musicam format, based on sub-band coding, was the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame). Much of its technology and ideas were incorporated into the definition of ISO MPEG Audio Layer I and Layer II and the filter bank alone into Layer III (MP3) format as part of the computationally inefficient hybrid filter bank. Under the chairmanship of Professor Musmann (University of Hannover) the editing of the standard was made under the responsibilities of Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II).
A
working group consisting of Leon van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) and James D. Johnston (USA) took ideas from ASPEC, integrated the filter bank from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.
All algorithms were approved in 1991 and finalized in 1992 as part of
MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3, published in 1993. Further work on MPEG audio 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, originally published in 1995.[6]
Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. They may use the 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 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.[citation needed] 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. Some jokingly refer to Suzanne Vega as "The mother of MP3".[who?] 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. It is important to understand that Suzanne Vega is recorded in an interesting fashion that results in substantial difficulties that arise due to Binaural Masking Level Depression (BMLD) as discussed in Brian C. J. Moore's book on the Psychology of Human Hearing, for instance.[