MPEG-1 Audio Layer III or
MPEG-2 Audio Layer III) is
an audio coding format for digital audio. Originally defined as the
third audio format of the
MPEG-1 standard, it was retained and further
extended—defining additional bit rates and support for more audio
channels—as the third audio format of the subsequent MPEG-2
standard. A third version, known as
MPEG 2.5—extended to better
support lower bit rates—is commonly implemented, but is not a
MP3 (or mp3) as a file format commonly designates files containing an
elementary stream of
MPEG-1 audio and video encoded data, without
other complexities of the
In the aspects of
MP3 pertaining to audio compression—the aspect of
the standard most apparent to end users (and for which is it best
MP3 uses lossy data compression to encode data using inexact
approximations and the partial discarding of data. This allows a large
reduction in file size when compared to uncompressed audio. The
combination of small size and acceptable fidelity led to a boom in the
distribution of music over the
Internet in the mid to late 1990s, as
an enabling technology when bandwidth and storage were still at a
MP3 format soon became associated with controversies
surrounding copyright infringement, music piracy, the file
MP3.com and Napster, among others. With the
advent of portable media players, a product category also including
MP3 support remains near-universal.
MP3 compression works by reducing (or approximating) the accuracy of
certain components of sound that are considered to be beyond the
hearing capabilities of most humans. This method is commonly referred
to as perceptual coding, or psychoacoustic modeling. The remaining
audio information is then recorded in a space-efficient manner.
Compared to CD-quality digital audio,
MP3 compression can commonly
achieve a 75 to 95% reduction in size. For example, an
MP3 encoded at
a constant bitrate of 128 kbit/s would result in a file
approximately 9% the size of the original CD audio.
Also designed as a streamable format, segments of a transmission can
be lost without affecting the ability to decode later segments.
MP3 was designed by the
Moving Picture Experts Group
Moving Picture Experts Group (MPEG) as part of
its MPEG-1, and later MPEG-2, standards. The first subgroup for audio
was formed by several teams of engineers at CCETT, Matsushita,
Philips, Sony, AT&T-Bell Labs, Thomson-Brandt, and others.
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, finalised in 1992, and published in 1993 as ISO/IEC
11172-3:1993. A backwards-compatible
MPEG-2 Audio (
MPEG-2 Part 3)
extension with lower sample and bit rates was published in 1995 as
1.3 Going public
2.2 Encoding and decoding
2.4 Bit rate
2.5 Ancillary data
3 Licensing, ownership and legislation
4 Alternative technologies
5 See also
7 Further reading
8 External links
MP3 lossy audio data compression algorithm takes advantage of a
perceptual limitation of human hearing called auditory masking. In
1894, the American physicist
Alfred M. Mayer
Alfred M. Mayer reported that a tone
could be rendered inaudible by another tone of lower frequency. In
1959, Richard Ehmer described a complete set of auditory curves
regarding this phenomenon. Ernst Terhardt et al. created an
algorithm describing auditory masking with high accuracy. 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. from
Bell Telephone Laboratories, Inc. in Murray Hill, New Jersey, and M.
A. Krasner 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
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 frequency
bands, that in turn built on the fundamental research in the
Bell Labs of Harvey Fletcher and his collaborators. A
wide variety of (mostly perceptual) audio compression algorithms were
reported in IEEE's refereed Journal on Selected Areas in
Communications. 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.
Moving Picture Experts Group
Moving Picture Experts Group (MPEG) was established in 1988 by the
Hiroshi Yasuda (Nippon Telegraph and Telephone) and
Leonardo Chiariglione. Yasuda was leading an initiative in Japan,
called the Digital Audio and Picture Architecture (DAPA), while
Chiariglione was leading an initiative in Europe, called the Coding of
Moving Images for Storage (COMIS). Both eventually met in May 1988 to
work on a global standard.
The genesis of the
MP3 technology is fully described in a paper from
Professor Hans Musmann, 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
MUSICAM, by Matsushita, CCETT, ITT and Philips. The second group was
ASPEC, by AT&T, France Telecom, Fraunhofer Gesellschaft, Deutsche
and Thomson-Brandt. The third group was ATAC, by Fujitsu, JVC,
Sony. And the fourth group was SB-ADPCM, by NTT and BTRL.
The immediate predecessors of
MP3 were "Optimum Coding in the
Frequency Domain" (OCF), and Perceptual Transform Coding
(PXFM). 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.
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 filterbank, driven by a psychoacoustic model. It was
primarily designed for Digital Audio Broadcasting (digital radio) and
digital TV, and its basic principles disclosed to the scientific
community by CCETT (France) and IRT (Germany) in Atlanta during an
IEEE-ICASSP conference in 1991, after having worked on MUSICAM
with Matsushita and
Philips since 1989.
This codec incorporated into a broadcasting system using COFDM
modulation was demonstrated on air and on the field together 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 chips 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 frequency, 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 a thorough use of a set of high quality audio
assessment material selected by a group of audio professionals
European Broadcasting Union
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 the 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
filterbank (this advantage being a specific feature of short transform
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.
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.
Johnston ("JJ") of AT&T-Bell Labs—with the Fraunhofer Institute
for Integrated Circuits, Erlangen (where he worked with Bernhard Grill
and four other researchers – "The Original Six"), 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). The song "Tom's
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
In 1991, there were two available proposals that were assessed for an
MPEG audio standard:
MUSICAM (Masking pattern adapted Universal
Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive
Spectral Perceptual Entropy Coding). As proposed by the Dutch
corporation Philips, the French research institute CCETT, and the
German standards organization Institute for Broadcast Technology, the
MUSICAM technique was chosen due to its simplicity and error
robustness, as well as for its high level of computational
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
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 bank.
Under the chairmanship of Professor Musmann of the University of
Hanover, the editing of the standard was delegated to Dutchman Leon
van de Kerkhof, to German Gerhard Stoll, to Frenchman Yves-François
Dehery, who worked on Layer I and Layer II. ASPEC was the joint
proposal of AT&T Bell Laboratories, Thomson Consumer Electronics,
Fraunhofer Society and CNET. It provided the highest coding
A working group consisting of van de Kerkhof, Stoll, Italian Leonardo
CSELT VP for Media), Frenchman Yves-François Dehery,
German Karlheinz Brandenburg, and American James D. Johnston (United
States) took ideas from ASPEC, integrated the filter bank from Layer
II, added some of their own ideas such as the joint stereo coding of
MUSICAM and created the
MP3 format, which was designed to achieve the
same quality at 128 kbit/s as MP2 at 192 kbit/s.
The algorithms for
MPEG-1 Audio Layer I, II and III 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 (a.k.a.
MPEG-1 Audio or
MPEG-1 Part 3), published in
1993. Files or data streams conforming to this standard must handle
sample rates of 48k, 44100 and 32k and continue to be supported by
MP3 players and decoders. Thus the first generation of MP3
defined 14*3=42 interpretations of
MP3 frame data structures and size
Further work on
MPEG audio was finalized in 1994 as part of the
second suite of
MPEG standards, MPEG-2, more formally known as
ISO/IEC 13818-3 (a.k.a.
MPEG-2 Part 3 or
MPEG-2 Audio or
MPEG-2 Audio BC), originally
published in 1995.
MPEG-2 Part 3 (ISO/IEC 13818-3) defined 42
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
MPEG-1 Audio. This reduction in sampling rate serves to cut
the available frequency fidelity in half while likewise cutting the
bitrate by 50%.
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
MP3 coded with
MPEG-2 results in half of the
bandwidth reproduction of
MPEG-1 appropriate for piano and singing.
A third generation of "MP3" style data streams (files) extended the
MPEG-2 ideas and implementation but was named MPEG-2.5 audio, since
MPEG-3 already had a different meaning. This extension was developed
at Fraunhofer IIS, the registered patent holders of
MP3 by reducing
the frame sync field in the
MP3 header from 12 to 11 bits. As in the
MPEG-1 to MPEG-2, MPEG-2.5 adds additional sampling
rates exactly half of those available using MPEG-2. It thus widens the
MP3 to include human speech and other applications yet
requires only 25% of the bandwidth (frequency reproduction) possible
MPEG-1 sampling rates. While not an ISO recognized standard,
MPEG-2.5 is widely supported by both inexpensive Chinese and brand
name digital audio players as well as computer software based MP3
encoders (LAME), decoders (FFmpeg) and players (MPC) adding 3*8=24
MP3 frame types. Each generation of
MP3 thus supports 3
sampling rates exactly half that of the previous generation for a
total of 9 varieties of
MP3 format files. The sample rate comparison
table between MPEG-1, 2 and 2.5 is given later in the article.
MPEG-2.5 is supported by
LAME (since 2000), Media Player Classic
(MPC), iTunes, and FFmpeg.
MPEG-2.5 was not developed by
MPEG (see above) and was never approved
as an international standard. MPEG-2.5 is thus an unofficial or
proprietary extension to the
MP3 format. It is nonetheless ubiquitous
and especially advantageous for low-bit rate human speech
MPEG Audio Layer III versions
First edition public release date
Latest edition public release date
MPEG-1 Audio Layer III
ISO/IEC 11172-3 (
MPEG-1 Part 3)
MPEG-2 Audio Layer III
ISO/IEC 13818-3 (
MPEG-2 Part 3)
MPEG-2.5 Audio Layer III
The ISO standard
ISO/IEC 11172-3 (a.k.a.
MPEG-1 Audio) defined three
MPEG-1 Audio Layer I, Layer II and Layer III. The ISO
ISO/IEC 13818-3 (a.k.a.
MPEG-2 Audio) defined extended
version of the
MPEG-2 Audio Layer I, Layer II and Layer
MPEG-2 Audio (
MPEG-2 Part 3) should not be confused with MPEG-2
MPEG-2 Part 7 – ISO/IEC 13818-7).
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
Compact Disc (CD) parameters as references
(44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit), or
Digital Audio Tape
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. 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 refer to
Suzanne Vega as
"The mother of MP3". 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.
LAME is the most advanced
MP3 encoder. LAME
includes a VBR variable bit rate encoding which uses a quality
parameter rather than a bit rate goal. Later versions 2008+) support
an n.nnn quality goal which automatically selects
MPEG-2 or MPEG-2.5
sampling rates as appropriate for human speech recordings which need
only 5512 Hz bandwidth resolution.
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 in order
to produce bit compliant
MPEG Audio files (Layer 1, Layer 2, Layer 3).
It was approved as a committee draft of ISO/IEC technical report in
March 1994 and printed as document CD 11172-5 in April 1994. It
was approved as a draft technical report (DTR/DIS) in November
1994, finalized in 1996 and published as international standard
ISO/IEC TR 11172-5:1998 in 1998. The reference software in C
language was later published as a freely available ISO standard.
Working in non-real time on a number of operating systems, it was able
to demonstrate the first real time hardware decoding (DSP based) of
compressed audio. Some other real time implementation of
encoders and decoders were available for the purpose of 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. The filename extension .mp3 was chosen by
the Fraunhofer team on 14 July 1995 (previously, the files had been
named .bit). With the first real-time software
MP3 player WinPlay3
(released 9 September 1995) many people were able to encode and play
MP3 files on their PCs. Because of the relatively small hard
drives back in that time (~ 500–1000 MB) lossy compression was
essential to store non-instrument based (see tracker and MIDI) music
for playback on computer. As sound scholar Jonathan Sterne notes, "An
Australian hacker acquired l3enc using a stolen credit card. The
hacker then reverse-engineered the software, wrote a new user
interface, and redistributed it for free, naming it "thank you
In the second half of the 1990s,
MP3 files began to spread on the
Internet, often via underground pirated song networks. The first known
Internet distribution was organized in the early 1990s
Internet Underground Music Archive, better known by the acronym
IUMA. After some experiments using uncompressed audio files, this
archive started to deliver on the native worldwide low speed Internet
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. In 1998, 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 afterwards 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
MP3 file sharing continues on next-generation
peer-to-peer networks. Some authorized services, such as Beatport,
Bleep, Juno Records, eMusic, Zune Marketplace, Walmart.com, Rhapsody,
the recording industry approved re-incarnation of Napster, and
Amazon.com sell unrestricted music in the
Diagram of the structure of an
MP3 file (
MPEG version 2.5 not
supported, hence 12 instead of 11 bits for
MP3 Sync Word).
MP3 file is made up of
MP3 frames, which consist of a header and a
data block. This sequence of frames is called an elementary stream.
Due to the "byte reservoir", frames are not independent items and
cannot usually be extracted on arbitrary frame boundaries. The MP3
Data blocks contain the (compressed) audio information in terms of
frequencies and amplitudes. The diagram shows that the
consists of a sync word, which is used to identify the beginning of a
valid frame. This is followed by a bit indicating that this is the
MPEG standard and two bits that indicate that layer 3 is used; hence
MPEG-1 Audio Layer 3 or MP3. After this, the values will differ,
depending on the
ISO/IEC 11172-3 defines the range of values
for each section of the header along with the specification of the
MP3 files today contain
ID3 metadata, which precedes or
MP3 frames, as noted in the diagram. The data stream can
contain an optional checksum.
Joint stereo is done only on a frame-to-frame basis.
Encoding and decoding
MPEG-1 standard does not include a precise specification for an
MP3 encoder, but does provide example psychoacoustic models, rate
loop, and the like in the non-normative part of the original
MPEG-2 doubles the number of sampling rates which are
supported and MPEG-2.5 adds 3 more. When this was written, the
suggested implementations were quite dated. Implementers of the
standard were supposed to devise their own algorithms suitable for
removing parts of the information from the audio input. As a result,
MP3 encoders became available, each producing files of
differing quality. Comparisons were widely available, so it was easy
for a prospective user of an encoder to research the best choice. Some
encoders that were proficient at encoding at higher bit rates (such as
LAME) were not necessarily as good at lower bit rates. Over time, LAME
evolved on the SourceForge website until it became the de facto CBR
MP3 encoder. Later an ABR mode was added. Work progressed on true
variable bit rate using a quality goal between 0 and 10. Eventually
numbers (such as -V 9.600) could generate excellent quality low bit
rate voice encoding at only 41 kbit/s using the MPEG-2.5
During encoding, 576 time-domain samples are taken and are transformed
to 576 frequency-domain samples.[clarification needed] If there is a
transient, 192 samples are taken instead of 576. This is done to limit
the temporal spread of quantization noise accompanying the transient.
(See psychoacoustics.) Frequency resolution is limited by the small
long block window size, which decreases coding efficiency. Time
resolution can be too low for highly transient signals and may cause
smearing of percussive sounds.
Due to the tree structure of the filter bank, pre-echo problems are
made worse, as the combined impulse response of the two filter banks
does not, and cannot, provide an optimum solution in time/frequency
resolution. Additionally, the combining of the two filter banks'
outputs creates aliasing problems that must be handled partially by
the "aliasing compensation" stage; however, that creates excess energy
to be coded in the frequency domain, thereby decreasing coding
Decoding, on the other hand, is carefully defined in the standard.
Most decoders are "bitstream compliant", which means that the
decompressed output that they produce from a given
MP3 file will be
the same, within a specified degree of rounding tolerance, as the
output specified mathematically in the ISO/IEC high standard document
(ISO/IEC 11172-3). Therefore, comparison of decoders is usually based
on how computationally efficient they are (i.e., how much memory or
CPU time they use in the decoding process). Over time this concern has
become less of an issue as
CPU speeds transitioned from MHz to GHz.
Encoder/decoder overall delay is not defined, which means there is no
official provision for gapless playback. However, some encoders such
LAME can attach additional metadata that will allow players that
can handle it to deliver seamless playback.
When performing lossy audio encoding, such as creating an
stream, there is a trade-off between the amount of data generated and
the sound quality of the results. The person generating an
a bit rate, which specifies how many kilobits per second of audio are
desired. The higher the bit rate, the larger the
MP3 data stream will
be, and, generally, the closer it will sound to the original
recording. With too low a bit rate, compression artifacts (i.e.,
sounds that were not present in the original recording) may be audible
in the reproduction. Some audio is hard to compress because of its
randomness and sharp attacks. When this type of audio is compressed,
artifacts such as ringing or pre-echo are usually heard. A sample of
applause or a triangle instrument with a relatively low bit rate
provide good examples of compression artifacts. Most subjective
testings of perceptual codecs tend to avoid using these types of sound
materials, however, the artifacts generated by percussive sounds are
barely perceptible due to the specific temporal masking feature of the
32 sub-band filterbank of Layer II on which the format is based.
Besides the bit rate of an encoded piece of audio, the quality of MP3
encoded sound also depends on the quality of the encoder algorithm as
well as the complexity of the signal being encoded. As the MP3
standard allows quite a bit of freedom with encoding algorithms,
different encoders do feature quite different quality, even with
identical bit rates. As an example, in a public listening test
featuring two early
MP3 encoders set at about 128 kbit/s, one
scored 3.66 on a 1–5 scale, while the other scored only 2.22.
Quality is dependent on the choice of encoder and encoding
This observation caused a revolution in audio encoding. Early on
bitrate was the prime and only consideration. At the time
were of the very simplest type: they used the same bit rate for the
entire file: this process is known as
Constant Bit Rate (CBR)
encoding. Using a constant bit rate makes encoding simpler and less
CPU intensive. However, it is also possible to create files where the
bit rate changes throughout the file. These are known as Variable Bit
Rate The bit reservoir and VBR encoding were actually part of the
MPEG-1 standard. The concept behind them is that, in any
piece of audio, some sections are easier to compress, such as silence
or music containing only a few tones, while others will be more
difficult to compress. So, the overall quality of the file may be
increased by using a lower bit rate for the less complex passages and
a higher one for the more complex parts. With some advanced MP3
encoders, it is possible to specify a given quality, and the encoder
will adjust the bit rate accordingly. Users that desire a particular
"quality setting" that is transparent to their ears can use this value
when encoding all of their music, and generally speaking not need to
worry about performing personal listening tests on each piece of music
to determine the correct bit rate.
Perceived quality can be influenced by listening environment (ambient
noise), listener attention, and listener training and in most cases by
listener audio equipment (such as sound cards, speakers and
headphones). Furthermore, sufficient quality may be achieved by a
lesser quality setting for lectures and human speech applications and
reduces encoding time and complexity. A test given to new students by
Stanford University Music Professor Jonathan Berger showed that
student preference for MP3-quality music has risen each year. Berger
said the students seem to prefer the 'sizzle' sounds that MP3s bring
An in-depth study of
MP3 audio quality, sound artist and composer Ryan
Maguire's project "The Ghost in the MP3" isolates the sounds lost
MP3 compression. In 2015, he released the track "moDernisT" (an
anagram of "Tom's Diner"), composed exclusively from the sounds
MP3 compression of the song "Tom's Diner",
the track originally used in the formulation of the
MP3 standard. A
detailed account of the techniques used to isolate the sounds deleted
MP3 compression, along with the conceptual motivation for the
project, was published in the 2014 Proceedings of the International
Computer Music Conference.
MPEG Audio Layer III
available bit rates (kbit/s)
Audio Layer III
Audio Layer III
Audio Layer III
Supported sampling rates
MPEG Audio Format
Audio Layer III
Audio Layer III
Audio Layer III
Bitrate is the product of the sample rate and number of bits per
sample used to encode the music. CD audio is 44100 samples per second.
The number of bits per sample also depends on the number of audio
channels. CD is stereo and 16 bits per channel. So, multiplying 44100
by 32 gives 1411200—the bitrate of uncompressed CD digital audio.
MP3 was designed to encode this 1411 kbit/s data at
320 kbit/s or less. As less complex passages are detected by MP3
algorithms then lower bitrates may be employed. When using MPEG-2
instead of MPEG-1,
MP3 supports only lower sampling rates (16000,
22050 or 24000 samples per second) and offers choices of bitrate as
low as 8 kbit/s but no higher than 160 kbit/s. By lowering
the sampling rate,
MPEG-2 layer III removes all frequencies above half
the new sampling rate that may have been present in the source audio.
As shown in these two tables, 14 selected bit rates are allowed in
MPEG-1 Audio Layer III standard: 32, 40, 48, 56, 64, 80, 96, 112, 128,
160, 192, 224, 256 and 320 kbit/s, along with the 3 highest
available sampling frequencies of 32, 44.1 and 48 kHz. MPEG-2
Audio Layer III also allows 14 somewhat different (and mostly lower)
bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144,
160 kbit/s with sampling frequencies of 16, 22.05 and 24 kHz
which are exactly half that of MPEG-1 MPEG-2.5 Audio Layer III
frames are limited to only 8 bit rates of 8, 16, 24, 32, 40, 48, 56
and 64 kbit/s with 3 even lower sampling frequencies of 8,
11.025, and 12 kHz.
MPEG-1 frames contain the most detail in 320 kbit/s mode with
silence and simple tones still requiring 32 kbit/s.
can capture up to 12 kHz sound reproductions needed up to
MP3 files made with
MPEG-2 don't have 20 kHz
bandwidth because of the Nyquist–Shannon sampling theorem. Frequency
reproduction is always strictly less than half of the sampling
frequency, and imperfect filters require a larger margin for error
(noise level versus sharpness of filter), so an 8 kHz sampling
rate limits the maximum frequency to 4 kHz, while a 48 kHz
sampling rate limits an
MP3 to a maximum 24 kHz sound
MPEG-2 uses half and MPEG-2.5 only a quarter of MPEG-1
For the general field of human speech reproduction, a bandwidth of
5512 Hz is sufficient to produce excellent results (for voice)
using the sampling rate of 11025 and VBR encoding from 44100
(standard) wave files.. This is easily accomplished using
3.99.5 and the command line "lame -V 9.6 lecture.WAV" English speakers
average 41–42 kbit/s with -V 9.6 setting but this may vary with
amount of silence recorded or the rate of delivery (wpm). Resampling
to 12000 (6K bandwidth) is selected by the
LAME parameter -V 9.4
Likewise -V 9.2 selects 16000 sample rate and a resultant 8K lowpass
filtering. For more info see Nyquist – Shannon. Older versions of
LAME and FFmpeg only support integer arguments for variable bit rate
quality selection parameter. The n.nnn quality parameter (-V) is
documented at lame.sourceforge.net but is only supported in
the new style VBR variable bit rate quality selector—not average bit
A sample rate of 44.1 kHz is commonly used for music
reproduction, because this is also used for CD audio, the main source
used for creating
MP3 files. A great variety of bit rates are used on
the Internet. A bit rate of 128 kbit/s is commonly used, at a
compression ratio of 11:1, offering adequate audio quality in a
relatively small space. As
Internet bandwidth availability and hard
drive sizes have increased, higher bit rates up to 320 kbit/s are
widespread. Uncompressed audio as stored on an audio-CD has a bit rate
of 1,411.2 kbit/s, (16 bit/sample × 44100 samples/second × 2
channels / 1000 bits/kilobit), so the bitrates 128, 160 and
192 kbit/s represent compression ratios of approximately 11:1,
9:1 and 7:1 respectively.
Non-standard bit rates up to 640 kbit/s can be achieved with the
LAME encoder and the freeformat option, although few
MP3 players can
play those files. According to the ISO standard, decoders are only
required to be able to decode streams up to 320 kbit/s. Early
MPEG Layer III encoders used what is now called Constant Bit Rate
(CBR). The software was only able to use a uniform bitrate on all
frames in an
MP3 file. Later more sophisticated
MP3 encoders were able
to use the bit reservoir to target an average bit rate selecting the
encoding rate for each frame based on the complexity of the sound in
that portion of the recording.
A more sophisticated
MP3 encoder can produce variable bitrate audio.
MPEG audio may use bitrate switching on a per-frame basis, but only
layer III decoders must support it. VBR is used when
the goal is to achieve a fixed level of quality. The final file size
of a VBR encoding is less predictable than with constant bitrate.
Average bitrate is a type of VBR implemented as a compromise between
the two: the bitrate is allowed to vary for more consistent quality,
but is controlled to remain near an average value chosen by the user,
for predictable file sizes. Although an
MP3 decoder must support VBR
to be standards compliant, historically some decoders have bugs with
VBR decoding, particularly before VBR encoders became widespread. The
MP3 encoder supports the generation of VBR, ABR, and
even the ancient CBR
Layer III audio can also use a "bit reservoir", a partially full
frame's ability to hold part of the next frame's audio data, allowing
temporary changes in effective bitrate, even in a constant bitrate
stream. Internal handling of the bit reservoir increases
encoding delay. There is no scale factor band 21
(sfb21) for frequencies above approx 16 kHz, forcing the encoder
to choose between less accurate representation in band 21 or less
efficient storage in all bands below band 21, the latter resulting in
wasted bitrate in VBR encoding.
The ancillary data field can be used to store user defined data. The
ancillary data is optional and the number of bits available is not
explicitly given. The ancillary data is located after the Huffman code
bits and ranges to where the next frame's main_data_begin points to.
mp3PRO uses ancillary data to encode their bits to improve audio
ID3 and APEv2 tag
A "tag" in an audio file is a section of the file that contains
metadata such as the title, artist, album, track number or other
information about the file's contents. The
MP3 standards do not define
tag formats for
MP3 files, nor is there a standard container format
that would support metadata and obviate the need for tags. However,
several de facto standards for tag formats exist. As of 2010, the most
widespread are ID3v1 and ID3v2, and the more recently introduced
APEv2. These tags are normally embedded at the beginning or end of MP3
files, separate from the actual
MP3 frame data.
MP3 decoders either
extract information from the tags, or just treat them as ignorable,
MP3 junk data.
Playing & editing software often contains tag editing
functionality, but there are also tag editor applications dedicated to
the purpose. Aside from metadata pertaining to the audio content, tags
may also be used for DRM.
ReplayGain is a standard for measuring
and storing the loudness of an
MP3 file (audio normalization) in its
metadata tag, enabling a ReplayGain-compliant player to automatically
adjust the overall playback volume for each file.
MP3Gain may be used
to reversibly modify files based on
ReplayGain measurements so that
adjusted playback can be achieved on players without ReplayGain
Licensing, ownership and legislation
MP3 decoding and encoding technology is patent-free in the
European Union, all patents having expired there by 2012 at the
latest. In the United States, the technology became substantially
patent-free on 16 April 2017 (see below). The majority of
expired in the US between 2007 and 2015. In the past, many
organizations have claimed ownership of patents related to MP3
decoding or encoding. These claims led to a number of legal threats
and actions from a variety of sources. As a result, uncertainty about
which patents must be licensed in order to create
MP3 products without
committing patent infringement in countries that allow software
patents was a common feature of the early stages of adoption of the
The initial near-complete
MPEG-1 standard (parts 1, 2 and 3) was
publicly available on 6 December 1991 as ISO CD 11172. In most
countries, patents cannot be filed after prior art has been made
public, and patents expire 20 years after the initial filing date,
which can be up to 12 months later for filings in other countries. As
a result, patents required to implement
MP3 expired in most countries
by December 2012, 21 years after the publication of ISO CD 11172.
An exception is the United States, where patents in force but filed
prior to 8 June 1995 expire after the later of 17 years from the issue
date or 20 years from the priority date. A lengthy patent prosecution
process may result in a patent issuing much later than normally
expected (see submarine patents). The various MP3-related patents
expired on dates ranging from 2007 to 2017 in the United States.
Patents for anything disclosed in ISO CD 11172 filed a year or more
after its publication are questionable. If only the known
filed by December 1992 are considered, then
MP3 decoding has been
patent-free in the US since 22 September 2015, when U.S. Patent
5,812,672, which had a PCT filing in October 1992,
expired. If the longest-running patent mentioned in the
aforementioned references is taken as a measure, then the MP3
technology became patent-free in the United States on 16 April 2017,
Patent 6,009,399, held and administered by
Technicolor, expired. As a result, many free and open-source
software projects, such as the Fedora operating system, have decided
to start shipping
MP3 support by default, and users will no longer
have to resort to installing unofficial packages maintained by third
party software repositories for
MP3 playback or encoding.
Technicolor (formerly called Thomson Consumer Electronics) claimed to
MP3 licensing of the Layer 3 patents in many countries,
including the United States, Japan, Canada and EU countries.
Technicolor had been actively enforcing these patents.
revenues from Technicolor's administration generated about €100
million for the
Fraunhofer Society in 2005. In September 1998, the
Fraunhofer Institute sent a letter to several developers of MP3
software stating that a license was required to "distribute and/or
sell decoders and/or encoders". The letter claimed that unlicensed
products "infringe the patent rights of Fraunhofer and Thomson. To
make, sell or distribute products using the [
MPEG Layer-3] standard
and thus our patents, you need to obtain a license under these patents
from us." This led to the situation where the
project could not offer its users official binaries that could run on
their computer. The project's position was that as source code, LAME
was simply a description of how an
MP3 encoder could be implemented.
Unofficially, compiled binaries were available from other sources.
Sisvel S.p.A. and its United States subsidiary Audio MPEG, Inc.
previously sued Thomson for patent infringement on
but those disputes were resolved in November 2005 with Sisvel granting
Thomson a license to their patents. Motorola followed soon after, and
signed with Sisvel to license MP3-related patents in December
2005. Except for three patents, the US patents administered by
Sisvel had all expired in 2015. The three exceptions are: U.S.
Patent 5,878,080, expired February 2017; U.S.
expired February 2017; and U.S.
Patent 5,960,037, expired 9 April
In September 2006, German officials seized
MP3 players from SanDisk's
booth at the
IFA show in Berlin after an Italian patents firm won an
injunction on behalf of Sisvel against
SanDisk in a dispute over
licensing rights. The injunction was later reversed by a Berlin
judge, but that reversal was in turn blocked the same day by
another judge from the same court, "bringing the
Patent Wild West to
Germany" in the words of one commentator. In February 2007, Texas
MP3 Technologies sued Apple, Samsung Electronics and Sandisk in
eastern Texas federal court, claiming infringement of a portable MP3
player patent that Texas
MP3 said it had been assigned. Apple,
Samsung, and Sandisk all settled the claims against them in January
Alcatel-Lucent has asserted several
MP3 coding and compression
patents, allegedly inherited from AT&T-Bell Labs, in litigation of
its own. In November 2006, before the companies' merger, Alcatel sued
Microsoft for allegedly infringing seven patents. On 23 February 2007,
a San Diego jury awarded
Alcatel-Lucent US $1.52 billion in damages
for infringement of two of them. The court subsequently revoked
the award, however, finding that one patent had not been infringed and
that the other was not owned by Alcatel-Lucent; it was co-owned by
AT&T and Fraunhofer, who had licensed it to Microsoft, the judge
ruled. That defense judgment was upheld on appeal in 2008. See
Microsoft for more information.
MP3 and Vorbis
The first is uncompressed
WAV file. The second is a
encoded at 48kbit/s, and third is an
MP3 encoded at 48kbit/s using
Problems playing this file? See media help.
Main article: List of codecs
Other lossy formats exist. Among these, mp3PRO, AAC, and MP2 are all
members of the same technological family as
MP3 and depend on roughly
similar psychoacoustic models. The
Fraunhofer Society owns many of the
basic patents underlying these formats as well, with others held by
Alcatel-Lucent, and Thomson Consumer Electronics. There are also
open compression formats like Opus and
Vorbis that are available free
of charge and without any known patent restrictions. Some of the newer
audio compression formats, such as AAC, WMA Pro and Vorbis, are free
of some limitations inherent to the
MP3 format that cannot be overcome
Besides lossy compression methods, lossless formats are a significant
MP3 because they provide unaltered audio content,
though with an increased file size compared to lossy compression.
Lossless formats include
FLAC (Free Lossless Audio Codec), Apple
Lossless and many others.
Information technology portal
Comparison of audio coding formats
MPEG-4 Part 14
Portable media player
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Wikimedia Commons has media related to MP3.
MP3 at Curlie (based on DMOZ)
MP3-history.com, The Story of MP3: How
MP3 was invented, by Fraunhofer
MP3 News Archive, Over 1000 articles from 1999-2011 focused on
MPEG Official Web site
HydrogenAudio Wiki, MP3
RFC 3119, A More Loss-Tolerant RTP Payload Format for
RFC 3003, The audio/mpeg Media Type
Multimedia compression and container formats
Motion JPEG 2000
Alliance for Open Media
Microsoft Video 1
Sorenson Video, Spark
MPEG-1 Layer III (MP3)
MPEG-1 Layer II
MPEG-1 Layer I
MPEG-H 3D Audio
G.711 (A-law, µ-law)
ITU-T, W3C, IETF
CCITT Group 4
ISO base media file format
MPEG-4 Part 14
MPEG-4 Part 14 (MP4)
Motion JPEG 2000
MPEG-21 Part 9
MPEG media transport
3GP and 3G2
DivX Media Format
MOD and TOD
VOB, IFO and BUP
See Compression methods for methods and Compression software for
MPEG (Moving Picture Experts Group)
Part 1: Systems
Part 2: Video
based on H.261
Part 3: Audio
Part 1: Systems (H.222.0)
Part 2: Video (H.262)
Part 3: Audio
Part 6: DSM CC
Part 7: Advanced Audio Coding
Part 2: Video
based on H.263
Part 3: Audio
Part 6: DMIF
Part 10: Advanced Video Coding (H.264)
Part 11: Scene description
Part 12: ISO base media file format
Part 14: MP4 file format
Part 17: Streaming text format
Part 20: LASeR
Part 22: Open Font Format
Part 2: Description definition language
Parts 2, 3 and 9: Digital Item
Part 5: Rights Expression Language
Part 3: Unified Speech and Audio Coding
MPEG media transport
Part 2: High Efficiency Video Coding
MPEG-H 3D Audio
Part 12: High Efficiency Image
Electronic and digital
Audio console (mixing board)
Digital audio workstation (DAW)
Comparison of analog and digital recording
Experimental musical instrument
Reel-to-reel audio tape recording
Sound reinforcement system
Digital signal processing
Sound reinforcement system
Electronic musical instrument
Digital audio editor
Digital audio workstation
Software effect processor
Sound recording engineer
People and organizations
Audio Engineering Society
Institute of Broadcast Sound
Musical Electronics Library
Professional Lighting and Sound Association
Professional audio store
New Interfaces for Musical Expression
New Interfaces for Musical Expression (NIME)