PoC at 21:39, 16 January 2006

2006-01-16T21:39:11Z

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'''MPEG Audio Layer-3''', or more commonly refered to as '''MP3''', is a popular [[digital audio]] encoding and [[lossy data compression|lossy]] [[audio compression|compression]] [[audio format|format]] invented and standardized in [[1991]] by a team of engineers working in the framework of the ISO/IEC MPEG audio committee under the chairmanship of Professor Hans Musmann (University of Hannover - [[Germany]]). It was designed to greatly reduce the amount of data required to represent audio, yet still sound like a faithful reproduction of the original uncompressed audio to most listeners. In popular usage, ''MP3'' also refers to [[files]] of sound or music recordings stored in the MP3 format on computers.

== Overview ==
MP3 is a [[data compression| compression]] format. It provides a representation of [[Pulse-code_modulation|pulse-code modulation-encoded]] (PCM) audio data in a much smaller size by discarding portions that are considered less important to human hearing (similar to [[JPEG]], a lossy compression for images).

A number of techniques are employed in MP3 to determine which portions of the audio can be discarded, including [[psychoacoustics]]. MP3 audio can be compressed with different [[bit rate]]s, providing a range of tradeoffs between data size and sound quality.

The MP3 format uses, at its heart, a hybrid transformation to transform a time domain signal into a frequency domain signal:

* 32-band [[polyphase quadrature filter]]
* 36 or 12 tap [[MDCT]]; size can be selected independent for sub-band 0...1 and 2...31
* [[Aliasing]] reduction postprocessing

[[MP3 Surround]], a version of the format supporting 5.1 channels for [[surround sound]], was introduced in December 2004. MP3 Surround is [[backward compatible]] with standard stereo MP3, and file sizes are similar.

In terms of the [[MPEG]] specifications, AAC ([[Advanced audio coding]]) from [[MPEG-4]] is to be the successor of the MP3 format, although there has been a significant movement to create and popularize other audio formats. Nevertheless, any succession is not likely to happen for a significant amount of time due to MP3's overwhelming popularity (MP3 enjoys extremely wide popularity and support, not just by end-users and software but by hardware such as DVD and CD players).

==History==
===Development===
MPEG-1 Audio Layer 2 encoding began as the [[Digital Audio Broadcast]] (DAB) project managed by Egon Meier-Engelen of the DFVLR (later on called DLR = Deutsche Luft und Raumfahrt = German Aerospace Agency) in [[Germany]]. This project was financed by the [[European Union]] as a part of the [[EUREKA]] research program where it was commonly known as EU-147. EU-147 ran from [[1987]] to [[1994]].

In [[1991]], there were two proposals available: [[Musicam]] (known as ''Layer 2''), and [[ASPEC]] (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (The Netherlands), CCETT (France), IRT (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated to the encoding of high quality compressed audio. The Musicam format based on subband coding was key to settle the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame). Its technologies and ideas were fully incorporated into the definition of ISO MPEG Audio Layer I and Layer II and further on of the Layer III (MP3) format. Under the chairmanship of Professor Mussmann (University of Hannover) the editing of the standard was made under the responsibilities of L. van de Kerkhof (Layer I) and G. Stoll (Layer II).

Further, on a [[working group]] consisting of [[J.D. Johnston]] (US), [[Gerhard Stoll]] (Germany), [[Yves-François Dehery]] (France), [[Karlheinz Brandenburg]] (Germany) took ideas from Musicam and ASPEC, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 [[kbit/s]] as [[MP2 (format)|MP2]] at 192 kbit/s.

All algorithms were finalized in [[1992]] as part of [[MPEG-1]], the first standard suite by [[MPEG]], which resulted in the international standard ''[[International Organization for Standardization|ISO]]/[[International Electrotechnical Commission|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 ''[[International Organization for Standardization|ISO]]/[[International Electrotechnical Commission|IEC]] 13818-3'', originally published in [[1995]].

Compression efficiency of encoders is typically defined by the bit rate because compression rate depends on the bit depth and [[sampling rate]] of the input signal. Nevertheless, there are often published compression rates that use the [[compact disc|CD]] parameters as references (44.1 [[kHz]], 2 channels at 16 bits per channel or 2x16 bit). Sometimes the [[Digital Audio Tape]] (DAT) SP parameters are used (48 kHz, 2x16 bit). Compression ratios with this reference are higher, which demonstrates the problem of the term ''compression ratio'' for lossy encoders.

Karlheinz Brandenburg used a CD recording of [[Suzanne Vega]]'s song ''[[Tom's Diner]]'' to assess the MP3 [[compression algorithm]]. This song was chosen because of its softness and simplicity, making it easier to hear imperfections in the compression format during playbacks. Some more serious and critical audio excerpts (glockenspiel, triangle, accordion, ...) 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.

=== MP3 goes public ===
A reference simulation software written in C language known as ISO 11172-5 was developed by the members of the ISO MPEG Audio committee in order to produce bit compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). 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.

Later on, on [[July 7]] [[1994]] the Fraunhofer Society released the first software MP3 encoder called [[l3enc]]. The [[filename extension]] ''.mp3'' was chosen by the Fraunhofer team on [[July 14]], [[1995]] (previously, the files had been named ''.bit''). With the first real-time software MP3 player [[Winplay3]] (released September 9th, 1995) many people were able to encode and playback MP3 files on their PCs. Because of the relatively small [[hard drive]]s back in that time (~500 [[MB]]) the technology was essential to store music for listening pleasure on a computer.

=== MP2 and MP3 and the Internet ===
In [[October]] [[1993]], [[MP2]] (''MPEG-1 Audio Layer 2'') files appeared on the [[Internet]] and were often played back using the ''[[Xing]] MPEG Audio Player'', and later in a program for [[Unix]] by [[Tobias Bading]] called [[MAPlay]], which was initially released on February 22nd, 1994 (MAPlay was also ported to the [[Microsoft Windows]]).

Initially the only encoder available for MP2 production was the Xing Encoder, accompanied by the program CDDA2WAV, a CD ripper that transformed CD audio tracks to computer data files.

The [[Internet Underground Music Archive]] (IUMA) is generally recognized as the start of the on-line music revolution. IUMA was the Internet's first high-fidelity music web site, hosting thousands of authorized MP2 recordings before MP3 or the web was popularized. IUMA was started by [[Rob Lord]] (who later headed pioneering [[Nullsoft]]) and [[Jeff Patterson (IUMA)|Jeff Patterson]], both from the University of California, Santa Cruz, in 1993. Other founding members include Jon Luini, Brandee Selck, and Ahin Savara.

In the first half of [[1995]] through the late [[1990s]], MP3 files began flourishing on the [[Internet]]. MP3 popularity was mostly due to, and interchangeable with, the successes of companies and software packages like Nullsoft's [[Winamp]] (released in [[1997]]), [[mpg123]], and [[Napster]] (released in [[1999]]). Those programs made it very easy for the average user to playback, create, share, and collect MP3s.

Controversies regarding [[peer-to-peer]] [[file sharing]] of MP3 files have flourished in recent years — largely because high compression enables sharing of files that would otherwise be too large and cumbersome to share. Due to the vastly increased spread of MP3s through the Internet some major record labels reacted by filing a [[Napster#Legal_challenges|lawsuit against Napster]] to protect their [[Copyright]]s (see also [[intellectual property]]).

Commercial online music distribution services (like the [[iTunes Music Store]]) usually prefer other/proprietary music file formats that support [[Digital_Rights_Management|Digital Rights Management]] (DRM) to control and restrict the use of digital music. The use of formats that supports DRM is in an attempt to prevent [[Pirate (disambiguation)|piracy]] of [[copyright]] protected materials, but any computer savvy person can easily rip the DRM from a song file turning it into a file that is not locked to any computer.

==Quality of MP3 audio==
Because MP3 is a [[lossy]] format, it is able to provide a number of different options for its "bit rate"—that is, the number of [[bit|bits]] of encoded data that are used to represent each second of audio. Typically rates chosen are between 128 and 256 [[kbit|kilobit]] per second. By contrast, uncompressed audio as stored on a [[compact disc]] has a bit rate of about 1400 kbit/s.

MP3 files encoded with a lower bit rate will generally play back at a lower quality. With too low a bit rate, "compression artifacts" (i.e., sounds that were not present in the original recording) may appear in the reproduction. A good demonstration of compression artifacts is provided by the sound of applause: it is hard to compress because it is random, therefore the failings of the encoder are more obvious, and are audible as ringing.

As well as the bit rate of the encoded file, the quality of MP3 files depend on the quality of the encoder and the difficulty of the signal being encoded. For average signals with good encoders, many listeners accept the MP3 bit rate of 128 kibit/s as near enough to compact disc quality for them, providing a [[compression ratio]] of approximately 11:1. However, listening tests show that with a bit of practice many listeners can reliably distinguish 128 kbit/s MP3s from CD originals; in many cases reaching the point where they consider the MP3 audio to be of unacceptably low quality. Yet other listeners, and the same listeners in other environments (such as in a noisy moving vehicle or at a party) will consider the quality acceptable. Obviously, imperfections in an MP3 encode will be much less apparent on low-end computer speakers than on a good stereo system connected to a computer or -- especially -- using high-quality headphones.

[[Fraunhofer Gesellschaft]] (FhG) publish on their official webpage the following compression ratios and data rates for MPEG-1 Layer 1, 2 and 3, intended for comparison:

* Layer 1: 384 kbit/s, compression 4:1
* Layer 2: 192...256 kbit/s, compression 8:1...6:1
* Layer 3: 112...128 kbit/s, compression 12:1...10:1

The differences between the layers are caused by the different psychoacoustic models used by them; the Layer 1 algorithm is typically substantially simpler, therefore a higher bit rate is needed for [[Transparency (data compression)|transparent]] encoding. However, as different encoders use different models, it is difficult to draw absolute comparisons of this kind.

Many people consider these quoted rates as being heavily skewed in favour of Layer 2 and Layer 3 recordings. They would contend that more realistic rates would be as follows:

* Layer 1: excellent at 384 kbit/s
* Layer 2: excellent at 256...384 kbit/s, very good at 224...256 Kbit/s, good at 192...224 Kbit/s
* Layer 3: excellent at 224...320 Kbit/s, very good at 192...224 Kbit/s, good at 128...192 Kbit/s

When comparing compression schemes, it is important to use encoders that are of equivalent quality. Tests may be biased against older formats in favour of new ones by using older encoders based on out-of-date technologies, or even buggy encoders for the old format. Due to the fact that their lossy encoding loses information, MP3 algorithms work hard to ensure that the parts lost cannot be detected by human listeners by modeling the general characteristics of human hearing (e.g., due to [[noise masking]]). Different encoders may achieve this with varying degrees of success.

A few possible encoders:
* [[LAME]] first created by Mike Cheng in early 1998. It is (in contrast to others) a fully [[LGPL]]'d MP3 encoder, with excellent speed and quality, rivaling even MP3's technological successors.
* [[Fraunhofer Gesellschaft]]: Some encoders are good, some have bugs.

Many early encoders that are no longer widely used:
* ISO dist10 reference code
* Xing
* BladeEnc
* ACM Producer Pro.

Good encoders produce acceptable quality at 128 to 160 Kibit/s and near-[[Transparency (data compression)|transparency]] at 160 to 192 kbit/s, while low quality encoders may never reach transparency, not even at 320 kbit/s. It is therefore misleading to speak of 128 kbit/s or 192 kbit/s quality, except in the context of a particular encoder or of the best available encoders. A 128 kbit/s MP3 produced by a good encoder might sound better than a 192 kbit/s MP3 file produced by a bad encoder.

It is important to note that quality of an audio signal is subjective. A given bit rate suffices for some listeners but not for others. Individual acoustic perception may vary, so it is not evident that a certain [[psychoacoustic]] model can give satisfactory results for everyone. Merely changing the conditions of listening, such as the audio playing system or environment, can expose unwanted distortions caused by lossy compression. The numbers given above are rough guidelines that work for many people, but in the field of lossy audio compression the only true measure of the quality of a compression process is to listen to the results.

If your aim is to archive sound files with no loss of quality (or work on the sound files in a studio for example), then you should use [[lossless data compression|Lossless compression]] algorithms, currently capable of compressing 16-bit PCM audio to 38% while leaving the audio identical to the original, such as Lossless Audio [http://www.lossless-audio.com LA], [[Apple Lossless]], [http://flac.sourceforge.net/ FLAC], Windows Media Audio 9 Lossless (wma) and [http://www.monkeysaudio.com/ Monkey's Audio] (among others). Lossless formats are strongly preferred for material that will be edited, mixed, or otherwise processed because the perceptual assumptions made by lossy encoders may not hold true after processing. The losses produced by multiple stages of coding may also compound each other, becoming more evident when the signal is reencoded after processing. Lossless formats produce the best possible result, at the expense of a lower compression ratio.

Some simple editing operations, such as cutting sections of audio, may be performed directly on the encoded MP3 data without necessitating reencoding. For these operations, the concerns mentioned above are not necessarily relevant, as long as appropriate software (such as ''mp3DirectCut'' and ''MP3Gain'') is used to prevent extra decoding-encoding steps.

==Bit rate==
The [[bit rate]] is variable for MP3 files. The general rule is that more information is included from the original sound file when a higher bit rate is used, and thus the higher the quality during play back. In the early days of MP3 encoding, a fixed bit rate was used for the entire file.

Bit rates available in MPEG-1 Layer 3 are 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, and the available sample frequencies are 32, 44.1 and 48 [[kHz]]. 44.1 kHz is almost always used (coincides with the sampling rate of [[compact disc]]s), and 128 kbit/s has become the [[de facto]] "good enough" standard, although 192 Kbit/s is becoming increasingly popular over [[peer-to-peer]] [[file sharing]] networks. MPEG-2 and [the non-official] MPEG-2.5 includes some additional bit rates: 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s

[[Variable bit rate]]s (VBR) are also possible. Audio in MP3 files are divided into frames (which have their own bit rate) so it is possible to change the bit rate dynamically as the file is encoded (although not originally implemented, VBR is in extensive use today). This technique makes it possible to use more bits for parts of the sound with higher dynamics (more ''sound movement'') and fewer bits for parts with lower dynamics, further increasing quality and decreasing storage space. This method compares to a sound activated tape recorder that reduces tape consumption by not recording silence. Some encoders utilize this technique to a great extent.

Non-standard bitrates up to 640 kbit/s can be achieved with the [[LAME]] encoder and the --freeformat option, however only few MP3 players can play those files.

==Design limitations of MP3==
There are several limitations inherent to the MP3 format that cannot be overcome by using a better encoder.

Newer audio compression formats such as [[Vorbis]] and [[advanced audio coding|AAC]] no longer have these limitations.

In technical terms, MP3 is limited in the following ways:

* Bitrate is limited to a maximum of 320 kbit/s
* Time resolution can be too low for highly transient signals
* No scale factor band for frequencies above 15.5/15.8 [[kHz]]
* [[Joint stereo]] is done on a frame-to-frame basis
* [[Encoder]]/[[decoder]] overall delay is not defined, which means lack of official provision for [[gapless playback]]; gaps may be introduced between tracks, although this can be avoided to a degree by using [[LAME]] to encode.

Nevertheless, a well-tuned MP3 encoder can perform competitively even with these restrictions.

==Encoding of MP3 audio==
The [[MPEG-1]] standard does not include a precise specification for an MP3 encoder.
The decoding algorithm and file format, as a contrast, are well defined.
Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information in the raw audio (or rather its [[Modified discrete cosine transform|MDCT]] representation in the frequency domain).During encoding 576 time domain samples are taken and is 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.

This is the domain of [[psychoacoustics]], which aims at understanding how human acoustical perception works (both in our [[ear]]s and in our [[brain]]).

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.
It must be kept in mind that an encoder that is proficient at encoding at higher bitrates (such as [[LAME]], which is in widespread use for encoding at higher bitrates) is not necessarily as good at other, lower bitrates.

==Decoding of MP3 audio==
Decoding, on the other hand, is carefully defined in the standard.
Most [[decoder]]s are "[[bitstream compliance|bitstream compliant]]", meaning that the uncompressed output 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 standard document.
The MP3 file has a standard format which is a frame consisting of 384, 576, or 1152 samples (depends on MPEG version and layer) and all the frames have associated header information(32 bits)
and side information(9, 17, or 32 bytes, depending on MPEG version and stereo/mono).The header and side information help the decoder to decode the associated huffman encoded data correctly.

Therefore, for the most part, comparison of decoders is almost exclusively based on how computationally efficient they are (i.e., how much [[computer memory|memory]] or [[CPU time]] they use in the decoding process).

==ID3 and other tags==
:''Main articles: [[ID3]] and [[APEv2 tag]]''

A "tag" is data stored in an MP3 (as well as other formats) that contains [[metadata]] such as the title, artist, album, track number or other information about the MP3 file to be added to the file itself.
The most widespread standard tag formats are currently the [[ID3]] ID3v1 and ID3v2 tags, and the more recent [[APEv2 tag]].

APEv2 was originally developed for the [[MPC (audio compression format)|MPC file format]] (see [http://www.personal.uni-jena.de/~pfk/mpp/sv8/apetag.html the APEv2 specification]).
APEv2 can coexist with ID3 tags in the same file, but it can also be used by itself.

==Volume normalization==
As [[compact disc]]s and other various sources are recorded and mastered at different volumes, it is useful to store volume information about a file in the tag so that at playback time, the volume can be dynamically adjusted.

A few standards for encoding the gain of an MP3 file have been proposed.
The idea is to normalize the volume (not the volume ''peaks'') of audio files, so that the volume does not change between consecutive tracks.

The most popular and widely used solution for storing replay gain is known simply as "[[Replay Gain]]".
Typically, the average volume and clipping information about an audio track is stored in the metadata tag.

==Alternative technologies==

Many other lossy audio [[codec]]s exist, including:
* [[MP2 (format)|MPEG-1/2 Audio Layer 2]] (MP2), MP3's predecessor;
* [[Ogg]] [[Vorbis]] from the [[Xiph.org Foundation]], a [[free software]] and patent free codec.
* [[MPC (audio compression format)|MPC]], also known as Musepack (formerly MP+), a derivative of MP2;
* [[mp3PRO]] from [[Thomson Multimedia]] combining MP3 with [[Spectral Band Replication|SBR]];
* [[AC-3]], used in [[Dolby Digital]] and [[DVD]];
* [[ATRAC]], used in [[Sony]]'s [[Minidisc]];
* [[MPEG-4]] [[Advanced audio coding|AAC]], used by [[Apple Computer|Apple]]'s [[iTunes Music Store]] and [[Apple iPod | iPod]]
* [[Windows Media Audio]] (WMA) from [[Microsoft]].
* [[QDesign]], used in [[QuickTime]] at low bitrates;
* [[AMR-WBplus|AMR-WB+]] Enhanced Adaptive Multi Rate WideBand codec, optimized for cellular and other limited bandwidth use;
* [[RealAudio]] from [[RealNetworks]], frequently in use for streaming on websites;
* [[Speex]], free software and patent free codec based on [[CELP]] specifically designed for speech and [[VoIP]].

mp3PRO, MP3, AAC, and MP2 are all members of the same technological family and depend on roughly similar [[psychoacoustic model]]s. The [[Fraunhofer Gesellschaft]] owns many of the basic [[patent]]s underlying these codecs, with [[Dolby Labs]], [[Sony]], [[Thomson Consumer Electronics]], and [[AT&T]] holding other key patents.

There are also some [[lossless data compression|lossless]] audio compression methods used on the Internet. While they are not similar to MP3, they are good examples of other compression schemes available. These include:
* [[FLAC]] stands for 'Free Lossless Audio Codec'
* [[Monkey's Audio]]
* [[SHN]], also known as Shorten
* [[TTA]]
* [[Wavpack]]
* [[Apple Lossless]]

Listening [http://www.rjamorim.com/test/ tests] have attempted to find the best-quality lossy audio codecs at certain bitrates. At 128kbit/s, Ogg Vorbis, AAC, MPC and WMA Pro tied for first place with LAME MP3 a little behind. At 64kbit/s, AAC-HE and mp3pro performed marginally better than other codecs. At high bitrates (128kbit/s+), most people do not hear significant differences. What is considered 'CD quality' is quite subjective; for some 128kbit/s MP3 is sufficient, while for others 200kbit/s or higher MP3 is necessary.

Though proponents of newer codecs such as WMA and RealAudio have asserted that their respective algorithms can achieve CD quality at 64 kbit/s, listening tests have shown otherwise; however, the quality of these codecs at 64 kbit/s is definitely superior to MP3 at the same bitrate. The developers of the patent-free [[Ogg]] [[Vorbis]] codec claim that their algorithm surpasses MP3, RealAudio and WMA sound quality, and the listening tests mentioned above support that claim. Thomson claims that its mp3PRO codec achieves CD quality at 64 kbit/s, but listeners have reported that a 64 kbit/s mp3PRO file compares in quality to a 112 kbit/s MP3 file and does not come reasonably close to CD quality until about 80 kbit/s.

MP3, which was designed and tuned for use alongside MPEG-1/2 Video, generally performs poorly on monaural data at less than 48 kbit/s or in stereo at less than 80 kbit/s.

==Licensing and patent issues==

[[Thomson SA|Thomson Consumer Electronics]] controls licensing of the [http://www.mp3licensing.com/patents/index.html MPEG-1/2 Layer 3 patents] in countries that recognize [[software patent]]s, including the [[United States]] and [[Japan]], but not EU countries. Thomson has been actively enforcing these patents. Thomson has been granted [[software patents]] in EU countries, but it is unclear whether or not they would be enforced by courts there. See [[Software patents under the European Patent Convention]].

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."

These patent issues significantly slowed the development of unlicensed MP3 software and led to increased focus on creating and popularizing alternatives such as [[WMA]] and [[Ogg Vorbis]]. [[Microsoft]], the makers of the Windows operating system, chose to move away from MP3 to their own proprietary [[Windows Media]] formats to avoid the licensing issues associated with the patents. Until the key patents expire, [[open source]] / [[free software]] encoders and players appear to be illegal for commercial use in countries that recognize software patents.

For information about licensing fees see [http://www.mp3licensing.com/help/developer.html here] and [http://www.mp3licensing.com/royalty/index.html here].

In spite of the patent restrictions, the perpetuation of the MP3 format continues; the reasons for this appear to be the [[network effect]]s caused by:
* familiarity with the format, not knowing alternatives exist,
* the fact that these alternatives do not universally provide a definite advantage over MP3,
* the large quantity of music now available in the MP3 format,
* the wide variety of existing software and hardware that takes advantage of the file format,
* the lack of [[DRM]]-protection technology, which makes MP3 files easy to edit, copy and distribute over networks,
* the majority of home users not knowing or not caring about the software patent controversy, which is in general irrelevant to their choice of the MP3 format for personal use.

Additionally, patent holders declined to enforce license fees on open source decoders, allowing many free MP3 decoders to develop. Furthermore, while attempts have been made to discourage distribution of encoder binaries, Thomson has stated that individuals using free MP3 encoders are not required to pay fees. Thus while patent fees have been an issue for companies attempting to use MP3, they have not meaningfully impacted users, allowing the format to grow in popularity.

Sisvel S.p.A. [http://www.sisvel.com] and Audio MPEG, Inc. [http://www.audiompeg.com], are suing Thomson for patent infringement on MP3 technology[http://www.zdnetindia.com/news/pressreleases/stories/128960.html]. Audio MPEG also starts licensing MP3 to vendors of MP3, so legal status of MP3 is unclear.

==See also==
*[[ID3]]
*[[MP3 Surround]]
*[[Joint stereo]]
*[[Digital audio player]]
*[[Media player]]
*[[Software patent]]
*[[Copyright infringement]]
*[[Comparison of audio codecs]]

==External links==
* [http://www.iis.fraunhofer.de/amm/techinf/layer3/ Fraunhofer IIS]
* [http://www.mp3licensing.com/patents/index.html List of relevant patents]
* [http://mpgedit.org/mpgedit/mpeg_format/MP3Format.html MP3 File Format Specification] (For MPEG 2 Layer III they say the frame contains 1152 frames and it actually contains 576 frames)

[[Category:Audio codecs]]
[[Category:Digital Revolution]]
[[Category:Digital audio]]

← Older revision		Revision as of 16:09, 11 March 2015
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−	'''MPEG Audio Layer-3''', or more commonly refered to as '''MP3''', is a popular [[digital audio]] encoding and [[lossy ~~data compression\|lossy]] [[~~audio compression~~\|compression]] [[audio format\|~~format]] invented and standardized in [[1991]] by a team of engineers working in the framework of the ISO/IEC MPEG audio committee under the chairmanship of Professor Hans Musmann (University of Hannover - [[Germany]]). It was designed to greatly reduce the amount of data required to represent audio, yet still sound like a faithful reproduction of the original uncompressed audio to most listeners. In popular usage, ''MP3'' also refers to [[files]] of sound or music recordings stored in the MP3 format on computers.	+	'''MPEG Audio Layer-3''', or more commonly refered to as '''MP3''', is a popular digital audio encoding and lossy audio compression format invented and standardized in 1991 by a team of engineers working in the framework of the ISO/IEC MPEG audio committee under the chairmanship of Professor Hans Musmann (University of Hannover - Germany). It was designed to greatly reduce the amount of data required to represent audio, yet still sound like a faithful reproduction of the original uncompressed audio to most listeners. In popular usage, ''MP3'' also refers to files of sound or music recordings stored in the MP3 format on computers.

	== Overview ==		== Overview ==

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MP3 - Revision history

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Squashua at 18:58, 3 February 2006

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 * [http://www.mp3licensing.com/patents/index.html List of relevant patents]
 * [http://mpgedit.org/mpgedit/mpeg_format/MP3Format.html MP3 File Format Specification] (For MPEG 2 Layer III they say the frame contains 1152 frames and it actually contains 576 frames)
 Wiki Source: [http://en.wikipedia.org/wiki/ Wikipedia]