MPEG-1 or MPEG-2 Audio
Layer III,[4] more commonly referred to
as MP3, is a patented encoding format for digital audio which uses a form oflossy data
compression. It is a common audio format for consumer audio streaming or storage, as well as a de facto standard of digital audio
compression for the transfer and playback of music on most 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 asperceptual 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.
Development:
The MP3 lossy audio data
compression algorithm takes advantage of a perceptual
limitation of human hearing called auditory masking. In 1894, the American
physicist Alfred
Marshall Mayerreported 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.
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 in 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 multichannel.[28]
An additional extension
to MPEG-2 is named MPEG-2.5 audio, as MPEG-3
already had a different meaning. This extension was developed at Fraunhofer
IIS, the registered MP3 patent holders. Like MPEG-2, MPEG-2.5 adds new sampling
rates exactly half of that previously possible with MPEG-2. It thus widens the
scope of MP3 to include human speech and other applications requiring only 25%
of the frequency reproduction possible with MPEG-1. While not an ISO recognized
standard, MPEG-2.5 is widely supported by both inexpensive and brand name
digital audio players as well as computer software based MP3 encoders and
decoders. A sample rate comparison between MPEG-1, 2 and 2.5 is given further
down. [30][31] MPEG-2.5 was not developed by
MPEG and was never approved as an international standard. MPEG-2.5 is thus an
unofficial or proprietary extension to the MP3 format.
|
Version
|
First
edition
public release date |
Latest
edition
public release date |
|
|
MPEG-1 Audio Layer III
|
ISO/IEC 11172-3 (MPEG-1 Part 3)
|
1993
|
|
|
MPEG-2 Audio Layer III
|
ISO/IEC 13818-3 (MPEG-2 Part 3)
|
1995
|
1998
|
|
MPEG-2.5 Audio Layer III
|
nonstandard, proprietary
|
||
·
Note: 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 extended version of
the 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).[11]
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 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.
This song was chosen because of its nearly monophonicnature 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".[33] 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. 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.
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.[34] It
was approved as a draft technical report (DTR/DIS) in November 1994,[35] finalized
in 1996 and published as international standard ISO/IEC TR 11172-5:1998 in
1998.[36] Thereference
software in C language was later published as a freely
available ISO standard.[37] 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 MPEG Audio encoders
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.[38] The filename extension .mp3 was
chosen by the Fraunhofer team on 14 July 1995 (previously, the files had been
named .bit).[1] With the first real-time
software MP3 player WinPlay3 (released 9
September 1995) many people were able to encode and play back 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.
In the second half of
'90s, MP3 files began to spread on the Internet. 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.[39]
In November 1997, the
website mp3.com was offering thousands of MP3s
created by independent artists for free.[39] 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.[40]
Despite the popularity of
the MP3 format, online music retailers often use other proprietary formats that
are encrypted or obfuscated in order to make it difficult to use purchased
music files in ways not specifically authorized by the record companies. Attempting
to control the use of files in this way is known as Digital Rights
Management. Record companies argue that this is necessary to prevent
the files from being made available on peer-to-peer file sharing networks. This
has other side effects, though, such as preventing users from playing back
their purchased music on different types of devices. However, the audio content
of these files can usually be converted into an unencrypted format. For
instance, users are often allowed to burn files to audio CD,
which requires conversion to an unencrypted audio format.
Unauthorized 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
MP3 format.
The 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
standard.[41] At present, these suggested
implementations are 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, there are many different MP3 encoders available,
each producing files of differing quality. Comparisons are widely available, so
it is easy for a prospective user of an encoder to research the best choice. An
encoder that is proficient at encoding at higher bit rates (such as LAME)
is not necessarily as good at lower bit rates.
During encoding, 576
time-domain samples are taken and are transformed to 576 frequency-domain
samples. 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.)
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).
When performing lossy
audio encoding, such as creating an MP3 file, there is a trade-off between the
amount of space used and the sound quality of the result. Typically, the
creator is allowed to set a bit rate, which specifies
how many kilobits the file may use per second of
audio. The higher the bit rate, the larger the compressed file will be, and,
generally, the closer it will sound to the original file.
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 compressed with a relatively low bit rate provides a good example of
compression artifacts.
Besides the bit rate of an
encoded piece of audio, the quality of MP3 files also depends on the quality of
the encoder itself, and the difficulty of the signal being encoded. As the MP3
standard allows quite a bit of freedom with encoding algorithms, different
encoders may feature quite different quality, even with identical bit rates. As
an example, in a public listening test featuring two different MP3 encoders at
about 128 kbit/s,[42] 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 parameters.[43]
The simplest type of MP3
file uses one bit rate for the entire file – this is known as Constant Bit Rate (CBR) encoding. Using a
constant bit rate makes encoding simpler and faster. However, it is also
possible to create files where the bit rate changes throughout the file. These
are known as Variable Bit Rate (VBR)
files. The idea behind this is that, in any piece of audio, some parts will be
much easier to compress, such as silence or music containing only a few
instruments, 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
encoders, it is possible to specify a given quality, and the encoder will vary
the bit rate accordingly. Users who know 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).
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 to music.[44]
Several bit rates are specified in the MPEG-1
Audio Layer III standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224,
256 and 320 kbit/s, with available sampling frequencies of
32, 44.1 and 48 kHz. [31] MPEG-2 Audio Layer III
allows 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.[31] MPEG-2.5 Audio Layer III is
restricted to bit rates of 8,
16, 24, 32, 40, 48, 56 and 64 kbit/s with sampling frequencies of
8, 11.025, and 12 kHz. Because of the Nyquist/Shannon theorem, frequency
reproduction is always strictly less than half of the sampling frequency, and
imperfect filters requires a larger margin for error (noise level versus
sharpness of filter), so 8 kHz sampling rate limits the maximum frequency
to 4 kHz, while 48 kHz maximum sampling rate limits an MP3 to
24 kHz sound reproduction.
A sample rate of
44.1 kHz is almost always used, because this is also used for CD audio,
the main source used for creating MP3 files. A greater variety of bit rates are
used on the Internet. The rate of 128 kbit/s is commonly used,[45] 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,[note
2] so the bitrates 128, 160 and 192 kbit/s
represent compression ratios of
approximately 11:1, 9:1 and 7:1 respectively.
Bit Rates:
Several bit rates are specified in the MPEG-1
Audio Layer III standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224,
256 and 320 kbit/s, with available sampling frequencies of
32, 44.1 and 48 kHz. [31] MPEG-2 Audio Layer III
allows 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.[31] MPEG-2.5 Audio Layer III is
restricted to bit rates of 8,
16, 24, 32, 40, 48, 56 and 64 kbit/s with sampling frequencies of
8, 11.025, and 12 kHz. Because of the Nyquist/Shannon theorem, frequency
reproduction is always strictly less than half of the sampling frequency, and
imperfect filters requires a larger margin for error (noise level versus
sharpness of filter), so 8 kHz sampling rate limits the maximum frequency
to 4 kHz, while 48 kHz maximum sampling rate limits an MP3 to
24 kHz sound reproduction.
A sample rate of
44.1 kHz is almost always used, because this is also used for CD audio,
the main source used for creating MP3 files. A greater variety of bit rates are
used on the Internet. The rate of 128 kbit/s is commonly used,[45] 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,[note
2] 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 Bit Rate audio. MPEG audio may
use bitrate switching on a per-frame basis, but only layer III decoders must
support it.[31][48][49][50]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 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.
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.[31][48]
An MP3 file is made up of
multiple 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 MP3 Header 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 MP3 file. ISO/IEC 11172-3 defines
the range of values for each section of the header along with the specification
of the header. Most MP3 files today contain ID3 metadata, which precedes or follows the MP3
frames, as noted in the diagram.
There are several
limitations inherent to the MP3 format that cannot be overcome by any MP3
encoder. Newer audio compression formats such as AAC, WMA Pro and Vorbis are generally free of a number of these
limitations.[51] In technical terms, some
limitations include:
·
Time resolution can be too low for highly transient signals and
may cause smearing of percussive sounds.[52]
·
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.[52]
·
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 efficiency.[citation needed]
·
Frequency resolution is limited by the small long block window
size, which decreases coding efficiency.[52]
·
There is no scale factor band for frequencies above
15.5/15.8 kHz.[original
research?]
·
Joint stereo is
done only on a frame-to-frame basis.[52]
·
Internal handling of the bit reservoir increases encoding delay.[citation needed]
·
Encoder/decoder overall delay is not defined,
which means there is no official provision for gapless playback. However, some encoders such
as LAME can attach additional metadata that
will allow players that can handle it to deliver seamless playback.
·
The data stream can contain an optional checksum, but the
checksum only protects the header data, not the audio data.
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, non-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.[citation needed]
Since volume levels of
different audio sources can vary greatly, due to the loudness war and other factors, it is
sometimes desirable to adjust the playback volume of audio files such that a
consistent average loudness is
perceived. This normalization,
while similar in purpose, is distinct from dynamic range
compression.
ReplayGain is one standard for measuring
and storing the loudness of an MP3 file 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 capability.
Many organizations have
claimed ownership of patents related
to MP3 decoding or encoding. These claims have led to a number of legal threats
and actions from a variety of sources, resulting in uncertainty about which patents
must be licensed in order to create MP3 products without committing patent
infringement in countries that allow software patents.
The initial near-complete
MPEG-1 standard (parts 1, 2 and 3) was publicly available on 6 December 1991 as
ISO CD 11172.[53][54] 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 filed prior to 8 June 1995 expire 17 years after the
publication date of the patent, and a loophole known as submarine patents that makes it possible
to extend the effective lifetime of a patent through application extensions.
The various MP3-related patents expire on dates ranging from 2007 to 2017 in
the U.S.[55] Patents filed for anything
disclosed in ISO CD 11172 a year or more after its publication are
questionable; if only the known MP3 patents filed by December 1992 are
considered, then MP3 decoding may be patent-free in the US by September 2015
when U.S. Patent 5,812,672 expires which had a
PCT filing in Oct 1992.[56][57][58]
Technicolor (formerly called Thomson
Consumer Electronics) claims to control MP3 licensing of the Layer 3 patents in
many countries, including the United States, Japan, Canada and EU countries.[59] Technicolor
has been actively enforcing these patents.[60]
MP3 license revenues
generated about €100 million for the Fraunhofer Society in 2005.[61]
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
and/or distribute products using the [MPEG Layer-3] standard and thus our
patents, you need to obtain a license under these patents from us."[62]
In spite of the patent
restrictions and the existence of free and more quality alternatives, the
perpetuation of the MP3 format continues. The reasons for this appear to be
the network effectscaused
by:
·
familiarity with the format; portable media player are
often sold as "MP3 players", even if they support other file formats;
·
the large quantity of music now available in the MP3 format;
·
the wide variety of existing hardware (and some software) that
supports the file format but not the alternatives (or not ones which are better
in any way);
·
the lack of DRM restrictions,
which makes all MP3 files easy to edit, copy and play in different portable
digital players (Apple, Creative, Samsung, etc.)[relevant? – discuss];
·
the majority of home users not knowing or not caring about the
patents' existence and often not considering such legal issues when choosing
their music format for personal use.
Additionally, patent
holders have declined to enforce license fees on free and open source decoders, which has allowed
many free MP3 decoders to be developed.[citation needed]
Sisvel S.p.A. and its U.S.
subsidiary Audio MPEG, Inc. previously sued Thomson for patent infringement on
MP3 technology,[63] 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.[64]
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,[65] 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.[66]
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 and Sandisk both settled the claims against them in January
2009.[67] Samsung settled as well.[68]
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.[69] The
court subsequently tossed the award, however, finding that one patent had not
been infringed and that the other was not even owned by Alcatel-Lucent; it was co-owned by AT&T and Fraunhofer, who had licensed
it to Microsoft, the judge ruled.[70] That
defense judgment was upheld on appeal in 2008.[71] See Alcatel-Lucent
v. Microsoft for more information.
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.
TheFraunhofer
Gesellschaft owns many of the basic patents underlying these formats as well, with others
held by Dolby Labs, Sony, Thomson
Consumer Electronics, and AT&T. In addition, there are also the open
source compression formats Opus and Vorbis that have been available free of charge and
without any known patent restrictions.
Besides lossy compression
methods, lossless codecs are
a significant alternative for MP3, as well as for lossy codecs in their
entirety, because they provide unaltered audio content, although with an
increased file size compared to lossy compression. Lossless codecs
include FLAC, a free lossless codec, Apple Lossless and others.
Source: Wikipedia
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