MP3
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Not to be confused with
MPEG-3.
MPEG-1 or MPEG-2 Audio Layer III,
[4] more commonly referred to as
MP3, is a
patented digital audio encoding format using a form of
lossy data compression. It is a common audio format for consumer audio storage, as well as a
de facto standard of digital audio compression for the transfer and playback of music on
digital audio players.
MP3 is an audio-specific format that was designed by the
Moving Picture Experts Group (MPEG) as part of its
MPEG-1 standard and later extended in
MPEG-2 standard. The first MPEG subgroup –
Audio group was formed by several teams of engineers at
Fraunhofer IIS,
University of Hannover,
AT&T-Bell Labs,
Thomson-Brandt,
CCETT, and others.
[7] MPEG-1 Audio (MPEG-1 Part 3), which included MPEG-1 Audio Layer I, II and III was approved as a committee draft of
ISO/
IEC standard in 1991,
[8][9] finalised in 1992
[10] and published in 1993 (ISO/IEC 11172-3:1993
[5]).
Backwards compatible MPEG-2 Audio (MPEG-2 Part 3) with additional bit
rates and sample rates was published in 1995 (ISO/IEC 13818-3:1995).
[6][11]
The use in MP3 of a
lossy compression algorithm
is designed to greatly reduce the amount of data required to represent
the audio recording and still sound like a faithful reproduction of the
original uncompressed audio for most listeners. An MP3 file that is
created using the setting of 128
kbit/s will result in a file that is about 1/11 the size
[note 1] of the
CD
file created from the original audio source. An MP3 file can also be
constructed at higher or lower bit rates, with higher or lower resulting
quality.
The compression works by reducing accuracy of certain parts of sound that are considered to be beyond the
auditory resolution ability of most people. This method is commonly referred to as
perceptual coding.
[13] It uses
psychoacoustic
models to discard or reduce precision of components less audible to
human hearing, and then records the remaining information in an
efficient manner.
[edit] History
[edit] Development
The MP3 lossy
audio data compression algorithm takes advantage of a perceptual limitation of human hearing called
auditory masking. In 1894,
Alfred Marshall Mayer reported that a tone could be rendered inaudible by another tone of lower frequency.
[14] In 1959, Richard Ehmer described a complete set of auditory curves regarding this phenomenon.
[15] Ernst Terhardt
et al. created an algorithm describing auditory masking with high accuracy.
[16]
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.
The psychoacoustic
masking codec was first proposed in 1979, apparently independently, by
Manfred R. Schroeder, et al.
[17] from AT&T-Bell Labs in
Murray Hill, NJ, and M. A. Krasner
[18]
both in the United States. Krasner was the first to publish and to
produce hardware for speech (not usable as music bit compression), 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, but his paper was not much noticed, since it
described negative results due to the particular nature of speech and
the
linear predictive coding
(LPC) gain present in speech. Both Krasner and Schroeder built upon the
work performed by Eberhard F. Zwicker in the areas of tuning and
masking of critical bands,
[19][20] that in turn built on the fundamental research in the area from
Bell Labs of Harvey Fletcher and his collaborators.
[21] A wide variety of (mostly perceptual) audio compression algorithms were reported in
IEEE's refereed Journal on Selected Areas in Communications.
[22]
That journal reported in February 1988 on a wide range of established,
working audio bit compression technologies, some 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),
[23] and Perceptual Transform Coding (PXFM).
[24]
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.
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.
[25]
MP3 is directly descended from OCF and PXFM, representing the outcome
of the collaboration of Brandenburg - working as a postdoc at
AT&T-Bell Labs with James D. (JJ) Johnston of AT&T-Bell Labs -
with the Fraunhofer Institut for Integrated Circuits, Erlangen, 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 (in 1993 he joined the staff of the Fraunhofer Institute).
[25]
The song
Tom's Diner by
Suzanne Vega was the first song used by
Karlheinz Brandenburg
to develop the MP3. Brandenburg adopted the song for testing purposes,
listening to it again and again each time refining the scheme, making
sure it did not adversely affect the subtlety of Vega's voice.
[edit] Standardization
In 1991, there were only two. proposals available that could be completely assessed for an MPEG audio standard:
Musicam (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and
ASPEC (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.
[26] 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).
ASPEC was the joint proposal of AT&T Bell Laboratories, Thomson Consumer Electronics, Fraunhofer Society and
CNET.
[27] It provided the highest coding efficiency.
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 for MPEG-1 Audio Layer I, II and III were approved in 1991
[8][9] and finalized in 1992
[10] 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.
[5] Further work on MPEG audio
[28] 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 backwards compatible
MPEG-2 Audio or
MPEG-2 Audio BC[11]), originally published in 1995.
[6][29]
MPEG-2 Part 3 (ISO/IEC 13818-3) defined 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 for MPEG-1 Audio. 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.
[28] There is also
MPEG-2.5
audio, a proprietary unofficial extension developed by Fraunhofer IIS.
It enables MP3 to work satisfactorily at very low bitrates and added
lower sampling frequencies.
[30][31] MPEG-2.5 wa