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'''[[MPEG-1]] Audio Layer 3''', more commonly referred to as '''MP3''', is a [[digital audio]] encoding format using a form of [[lossy data compression]].
'''[[MPEG-1]] Audio Layer 3''', more commonly referred to as '''niggers''', is a [[digital audio]] encoding format using a form of [[intelligence lossy data compression]].

[[Image:Pyramidhead.png]]


This encoding format is used to create the MP3's small [[computer file|file]], as a way to store a single segment of [[audio]], commonly a [[song]], such that the file can be easily organized and transferred between computers or other devices such as [[digital audio player|MP3 players]].
This encoding format is used to create the MP3's small [[computer file|file]], as a way to store a single segment of [[audio]], commonly a [[song]], such that the file can be easily organized and transferred between computers or other devices such as [[digital audio player|MP3 players]].


MP3's use of a [[lossy data compression|lossy]] [[audio data compression|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 some listeners, but is not considered [[High Fidelity]] audio by the elite connoisseur. An MP3 file that is created using the mid-range [[bitrate]] setting of 128 kbit/s will result in a file that is typically about 1/10th the size of the [[Red Book (audio CD standard)|CD]] file created from the original audio source. An MP3 file can also be constructed at higher or lower bitrates, with higher or lower resulting quality.
MP3's use of a [[god damn bullshit]] [[audio data compression|compression]] [[algorithm]] is designed to greatly increase the amount of cum required to represent the audio recording and still sound like a faithful to marriage of sandwich nigger reproduction of the original uncompressed audio for some listeners, but is not considered [[High-fi Fidelity]] audio by the elite connoisseuwhateverthefuckthiswordis. An MP3 file is shittier than SID format. That is created using the mid-range [[bittorrentclients]] setting of 1280000000 gaybit/s will result in a file that is typically about 1/10th the size of the [[Red Book (audio CD standard)|CD]] file created from the original audio source. An MP3 file can also be constructed at higher or lower braces, with higher or lower resulting quality.

MP3 is an audio-specific format. It was invented by a team of international engineers at [[Philips]], [http://fr.wikipedia.org/wiki/CCETT CCETT] (Centre commun d'études de télévision et télécommunications), [[Institut für Rundfunktechnik|IRT]], [[Bell labs|AT&T-Bell Labs]] and [[Fraunhofer Society]], and it became an [[International Organization for Standardization|ISO]]/[[International Electrotechnical Commission|IEC]] standard in 1991. The compression works by reducing accuracy of certain parts of sound that are deemed beyond the [[auditory]] resolution ability of most people. This method is commonly referred to as Perceptual Coding.
<ref>{{cite paper
| author = Jayant, N. S., Johnston, J. D. and Safranek, R. J.
| title = Signal compression based on models of human perception
| publisher = Proc. IEEE, Oct. 1993, pp. 1385-1422
| date= 1993–10
| url = http://ieeexplore.ieee.org/xpl/freeabs_all.jsp?arnumber=241504 }}</ref>

It provides a representation of sound within a short term time/frequency analysis window, by using [[psychoacoustic]] models to discard or reduce precision of components less audible to human hearing, and recording the remaining information in an efficient manner. This is relatively similar to the principles used by, say, [[JPEG]], an image compression format.

==History==
===Development===

The psycho-acoustic masking [[codec]] was first proposed, apparently independently in 1979, by Manfred Schroeder, ''et al.''<ref>"Optimizing Digital Speech Coding by Exploiting Masking Properties of the Human Ear"; M. R. Schroeder, B. S. Atal and J. L. Hall; J. Acoust. Soc. Am.; received 8 June 1979; accepted for publication 13 August 1979; Dec. 1979</ref> from AT&T-Bell Labs in [[Murray Hill, NJ]], and M. A.Krasner<ref>"Digital Encoding of Speech and Audio Signals Based on the Perceptual Requirements of the Auditory System"; M. A. Krasner; Massachusetts Institute of Technology Lincoln Laboratory Technical Report 535; 18 June 1979</ref> both in the United States. Krasner was the first to publish and to produce hardware, but the publication of his results as a relatively obscure [[Lincoln Laboratory]] Technical Report did not immediately influence the mainstream of psychoacoustic codec development. Manfred Schroeder was already a well known and revered figure in the world wide community of acoustical and electrical engineers and his paper had influence in acoustic and source-coding (audio compression) research. Both Krasner and Schroeder built upon the work of E. F. Zwicker <ref>"On the Psychoacoustical Equivalent of Tuning Curves"; E. F. Zwicker; Proceedings of the Symposium on Psychophysical Models and Physiological Facts in Hearing; held at Tuzing, Oberbayern, April 22–26, 1974</ref>, that in turn built on the fundamental Labs]] of Harvey Fletcher and his collaborators. <ref>"The ASA Edition of Speech and Hearing in Communication", edited by J.B. Allen, Acoustical Society of America, reprinted in 1995</ref> A wide variety of audio compression algorithms, mostly (but not completely) perceptual were reported in a refereed journal, the Journal on Selected Areas in Communications, <ref> IEEE Jour. Selected Areas in Commun., vol. 6, no. 2, Feb 1988</ref>. That journal reported in Feb. 1988 on a wide range of established, working audio bit compression technologies, most of them using auditory masking as part of their fundamental design.

The immediate predecessors of MP3 were "Optimum Coding in the Frequency Domain" (OCF),<ref>{{cite paper
| author = K. Brandenburg, D. Seitzer; Universitaet Erlangen-Nuernberg, Erlangen
| title = OCF: Coding High Quality Audio with Data Rates of 64 KBit/sec
| publisher = Audio Engineering Society, 85th Convention
| date= November 3–6 1988
| url = http://www.aes.org/e-lib/browse.cfm?elib=4721 }}</ref> and Perceptual Transform Coding (PXFM).<ref>Johnston, J. D., “Transform coding of audio signals using perceptual noise criteria,” IEEE Jour. Selected Areas in Commun., vol. 6, no. 2, Feb 1988, pp. 314-323 </ref> 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 which was an implementation of a psychoacoustic transform coder based on Motorola 56000 [[Digital Signal Processor|DSP]] chips. MP3 is directly descended from OCF and PXFM. MP3 represents the outcome of the collaboration of Dr. Karlheinz Brandenburg, working as a PostDoc at AT&T-Bell Labs with Mr. James D. Johnston of AT&T-Bell Labs, collaborating with the Fraunhofer Society for Integrated Circuits, Erlangen, with relatively minor contributions from the Musicam (MP2) branch of psychoacoustic sub-band coders.

[[MPEG-1 Audio Layer 2]] encoding began as the [[Digital Audio Broadcast]] (DAB) project managed by [[Egon Meier-Engelen]] of the ''Deutsche Forschungs- und Versuchsanstalt für Luft- und Raumfahrt'' (later on called ''Deutsches Zentrum für Luft- und Raumfahrt'', [[German Aerospace Center]]) in [[Germany]]. This project was financed by the European Union as a part of the [[EUREKA]] research program where it was commonly known as EU-147, which ran from 1987 to 1994.

As a doctoral student at Germany's [[University of Erlangen-Nuremberg]], [[Karlheinz Brandenburg]] began working on digital music compression in the early 1980s, focusing on how people perceive music. He completed his doctoral work in 1989 and became an assistant professor at Erlangen-Nuremberg. While there, he continued to work on music compression with scientists at the [[Fraunhofer Society]] (in 1993 he joined the staff of the Fraunhofer Institute).<ref>{{cite web |author=Jack Ewing |title=How MP3 Was Born |publisher=BusinessWeek.com |date=March 5, 2007 |url=http://www.businessweek.com/print/globalbiz/content/mar2007/gb20070305_707122.htm |accessdate=2007-07-24 }}</ref>

In 1991, there were two proposals available: [[Musicam]] (known as ''Layer 2''), and [[Adaptive Spectral Perceptual Entropy Coding|ASPEC]] - [http://de.wikipedia.org/wiki/Adaptive_Spectral_Perceptual_Entropy_Coding (Short excerpt on German Wikipedia)] (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by [[Philips]] (The Netherlands), [[Centre commun d'études de télévision et télécommunications|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. The Musicam format, based on [[sub-band coding]], was a basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame). Its technology 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 Musmann ([[University of Hannover]]) the editing of the standard was made under the responsibilities of [[Leon van de Kerkhof]] (Layer I) and [[Gerhard Stoll]] (Layer II).

A [[working group]] consisting of [[Leon van de Kerkhof]] (The Netherlands), [[Gerhard Stoll]] (Germany), [[Leonardo Chiariglione]] (Italy), [[Yves-François Dehery]] (France), [[Karlheinz Brandenburg]] (Germany) and [[James D.Johnston]] (USA) took ideas from ASPEC, integrated the filterbank 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 (format)|MP2]] at 192 kbit/s.

All algorithms were approved in 1991 and finalized in 1992 as part of [[MPEG-1]], the first standard suite by [[MPEG]], which resulted in the international standard ''[[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 ''ISO/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 2×16 bit). Sometimes the [[Digital Audio Tape]] (DAT) SP parameters are used (48 kHz, 2×16 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 nearly monophonic nature and wide spectral content, making it easier to hear imperfections in the compression format during playbacks. Some jokingly refer to Suzanne Vega as "The mother of MP3". Some more critical audio excerpts ([[glockenspiel]], [[triangle (instrument)|triangle]], [[accordion]], etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats. It is important to understand that [[Suzanne Vega]] is recorded in an interesting fashion that results in substantial difficulties that arise due to [[Binaural Masking Level Depression]] (BMLD).{{Fact|date=March 2008}}

=== Going public ===
{{Unreferencedsection|date=December 2007}}
A reference simulation software implementation, written in the C language and 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 ([[Digital Signal Processor|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 [[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 9]], [[1995]]) many people were able to encode and play back MP3 files on their PCs. Because of the relatively small [[hard drive]]s back in that time (~ 500 [[megabyte|MB]]) the technology was essential to store non-instrument based (see [[tracker]] and [[Musical Instrument Digital Interface|MIDI]]) music for listening on a computer.

=== Internet ===
From the first half of 1995 through the late 1990s, MP3 files began to spread on the [[Internet]]. MP3's popularity began to rise rapidly with the advent of [[Nullsoft]]'s audio player [[Winamp]] (released in 1997), and the Unix audio player [[mpg123]]. The small size of MP3 files has enabled widespread [[peer-to-peer]] [[file sharing]] of music [[Ripping|ripped]] from [[compact disc]]s, which would previously have been nearly impossible. The first large peer-to-peer filesharing network, [[Napster]], was released in 1999.

The ease of creating and sharing MP3s resulted in widespread [[copyright]] infringment. Major record companies argue that this free sharing of music reduces sales, and call it "[[music piracy]]". They have reacted by pursuing lawsuits against Napster, which was eventually closed down, and eventually against individual users who engaged in file sharing.

Despite the popularity of MP3, online music retailers often use other proprietary formats that are encrypted (known as [[Digital rights management]]) to prevent users from using purchased music in ways not specifically authorized by the record companies. The record companies argue that this is necessary to prevent the files from being made available on peer-to-peer file sharing networks. However, this has other side effects such as preventing users from playing back their purchased music on different types of devices. The audio content of these files can be converted into an unencrypted format, however, because often the user permissions include "burn to [[Red Book (audio CD standard)|audio CD]]". And even when that option is not available, many [[sound card|soundcards]] allow the user to record anything they play.
Unauthorized MP3 filesharing continues on next-generation peer-to-peer networks, though some authorized services, such as [[eMusic]], [[Amazon.com#Amazon_MP3_Downloads|Amazon.com]], and [[Zune]] sell unrestricted music in the MP3 format.

==Encoding 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 are transformed to 576 frequency domain samples. If there is a [[Transient (acoustics)|transient]], 192 samples are taken instead of 576. This is done to limit the temporal spread of quantization noise accompanying the transient. (See [[psychoacoustics]].)

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 bit rates (such as [[LAME]]) is not necessarily as good at other, lower bit rates.

==Decoding audio==
Decoding, on the other hand, is carefully defined in the standard.
Most [[decoder]]s are "[[Elementary stream|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 [http://le-hacker.org/hacks/mpeg-drafts/11172-3.pdf ISO/IEC standard document].
The MP3 file has a standard format, which is a frame that consists 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 coding|Huffman]] encoded data correctly.

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

==Audio quality ==
When 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 of the MP3 file is allowed to set a [[bit rate]], which specifies how many [[kilobits]] the file may use per second of audio, for example, when [[ripping]] a [[Red Book (audio CD standard)|compact disc]] to this [[format]]. The lower the bit rate used, the lower the audio quality will be, but the smaller the file size. Likewise, the higher the bit rate used, the higher quality, and therefore, larger the file size the resulting MP3 will be.

As described, MP3 files encoded with a lower bit rate will generally play back at a lower quality. With too low a bit rate, "[[compression artifact]]s" (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 bitrate 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 when targeting similar bit rates. As an example, in a public listening test featuring two different MP3 encoders at about 128 kbit/s,<ref>{{Citation
| last = Amorim
| first = Roberto
| author-link = http://www.rjamorim.com/home-en.html
| title = Results of 128kbps Extension Public Listening Test
| date= 2003-08-03
| year = 2003
| url = http://www.rjamorim.com/test/128extension/results.html
| accessdate = 2007-03-17 }}</ref> one scored 3.66 on a 1–5 scale, while the other scored only 2.22.

Quality is heavily dependent on the choice of encoder and encoding parameters. While quality around 128&nbsp;kbit/s was somewhere between annoying and acceptable with older encoders, modern MP3 encoders can provide adequate quality at those bit rates<ref name="listening-test-128-2006">{{Citation
| last = Mares
| first = Sebastian
| author-link = http://www.maresweb.de/about/smares.php
| title = Results of Public, Multiformat Listening Test @ 128 kbps
| date= 2006–01
| year = 2006
| url = http://www.listening-tests.info/mf-128-1/results.htm
| accessdate = 2007-03-17 }}</ref> (January 2006). However, in 1998, MP3 at 128&nbsp;kbit/s was only providing quality equivalent to AAC-LC at 96&nbsp;kbit/s and MP2 at 192&nbsp;kbit/s.<ref>{{cite paper
| author = David Meares, Kaoru Watanabe & Eric Scheirer
| title = Report on the MPEG-2 AAC Stereo Verification Tests
| publisher = [[International Organisation for Standardisation]]
| date= 1998–02
| url = http://sound.media.mit.edu/mpeg4/audio/public/w2006.pdf
| format = PDF
| accessdate = 2007-03-17 }}</ref>

The [[Transparency (data compression)|transparency]] threshold of MP3 can be estimated to be at about 128&nbsp;kbit/s with good encoders on typical music as evidenced by its strong performance in the above test, however some particularly difficult material, or music encoded for the use of people with more sensitive hearing can require 192&nbsp;kbit/s or higher. As with all lossy formats, some samples cannot be encoded to be transparent for all users.

The simplest type of MP3 file uses one bit rate for the entire file — this is known as [[Constant bitrate|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 bitrate|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 not need to worry about performing personal listening tests on each piece of music to determine the correct settings.

In a listening test, MP3 encoders at low bit rates performed significantly worse than those using more modern compression methods (such as AAC). In a 2004 public listening test at 32 kbit/s,<ref name="listening-test-32-2004">{{Citation
| last = Amorim
| first = Roberto
| author-link = http://www.rjamorim.com/
| title = Results of Dial-up bit rate public Listening Test
| date= 2004-07-11
| year = 2004
| url = http://www.rjamorim.com/test/32kbps/results.html
| accessdate = 2007-03-17 }}</ref> the LAME MP3 encoder scored only 1.79/5 — behind all modern encoders — with [[Nero Digital|Nero Digital HE AAC]] scoring 3.30/5.

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).

==Bit rate==
Several bit rates are specified in the MPEG-1 Layer 3 standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 192, 224, 256 and 320 kbit/s, and the available [[sampling frequencies]] are 32, 44.1 and 48 [[kHz]]. A sample rate of 44.1 kHz is almost always used, because this is also used for [[Red Book (audio CD standard)|CD audio]], the main source used for creating MP3 files. A greater variety of bit rates are used on the Internet. 128 kbit/s is the most common, beause it typically offers adequate audio quality in a relatively small space. 192 kbit/s is often used by those who notice artifacts at lower bit rates. As the Internet [[bandwidth]] availability and hard drive sizes have increased, 128 kbit/s bitrate files are slowly being replaced with higher bitrates like 192 kbit/s, with some being encoded up to MP3's maximum of 320 kbit/s. It is unlikely that higher bit rates will be popular with any [[lossy]] audio codec as higher bit rates than 320 kbit/s encroach on the domain of [[lossless]] codecs such as [[FLAC]].

By contrast, uncompressed audio as stored on a [[compact disc]] has a bit rate of 1,411.2&nbsp;kbit/s (16 bits/sample &times; 44100 samples/second &times; 2 channels / 1000 bits/kilobit).

Some additional bit rates and sample rates were made available in the MPEG-2 and the (unofficial) MPEG-2.5 standards: bit rates of 8, 16, 24, and 144&nbsp;kbit/s and sample rates of 8, 11.025, 12, 16, 22.05 and 24 kHz.

Non-standard bit rates up to 640&nbsp;kbit/s can be achieved with the [[LAME]] encoder and the freeformat option, although few MP3 players can play those files. Gabriel Bouvigne, a principal developer of the LAME project, says that the freeformat option is compliant with the standard but, according to the standard, decoders are only required to be able to decode streams up to 320 kbit/s.<ref>{{Citation
| last = Bouvigne
| first = Gabriel
| author-link = http://gabriel.mp3-tech.org/
| title = freeformat at 640 kbps and foobar2000, possibilities?
| date= 2006-11-28
| year = 2006
| url = http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=38808&view=findpost&p=452751
| accessdate = 2007-03-17 }}</ref>

==File structure==

[[Image:Mp3filestructure.svg|950px]]

An MP3 file is made up of multiple MP3 frames, which consist of the MP3 header and the MP3 data. This sequence of frames is called an [[Elementary stream]]. Frames are not independent items ("byte reservoir") and therefore cannot be extracted on arbitrary frame boundaries. The MP3 data is the actual audio payload. 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. ''[[International Organization for Standardization|ISO]]/[[International Electrotechnical Commission|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; this is also shown in the diagram.

==Design limitations {{Anchor | design}} ==
There are several limitations inherent to the MP3 format that cannot be overcome by any MP3 encoder. Newer audio compression formats such as [[Vorbis]], [[WMA Pro]] and [[Advanced Audio Coding|AAC]] no longer have these limitations. In technical terms, MP3 is limited in the following ways:
* Time resolution can be too low for highly transient signals, may cause some smearing of percussive sounds although this effect is to a great extent limited by the psychoacoustical properties of the Musicam polyphase [[Filter_bank|filterbank]] (Layer II). Pre-echo is concealed due to the specific time-domain characteristics of the filter.
* Due to the tree structure of the filterbank, pre-echo issues are made worse, as the combined impulse response of the two filterbanks does not, and can not, provide an optimum solution in time/frequency resolution.
* The combination of the two filterbanks creates aliasing issues that must be handled partially by the "aliasing compensation" stage, but that create excess energy to be coded in the frequency domain, thereby decreasing coding efficiency
* Frequency resolution is limited by the small long block window size, decreasing coding efficiency
* No scale factor band for frequencies above 15.5/15.8 [[kHz]]
* [[Joint stereo]] is done only on a frame-to-frame basis
* Internal handling of the bit reservoir increases encoding delay
* [[Encoder]]/[[decoder]] overall delay is not defined, which means lack of official provision for [[gapless playback]]. However, some encoders such as [[LAME]] can attach additional metadata that will allow players that are aware of it to deliver seamless playback.

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

A "tag" in a compressed 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.

As of [[2006]], the most widespread standard tag formats are [[ID3|ID3v1 and ID3v2]], and the more recently introduced [[APEv2 tag|APEv2]].

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 or it can also be used by itself.

Tag editing functionality is often built-in to MP3 players and editors, but there also exist [[tag editor]]s dedicated to the purpose (see [http://filerename.co.uk/index.html filerename.co.uk] for a free open source example).

==Volume normalization==
As [[compact disc]]s and other various sources are recorded and mastered at different volumes, it may be 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 average volume (not the volume ''peaks'') of audio files, so that the volume does not change between consecutive tracks. This should not be confused with [[dynamic range compression]] (DRC), which is a form of normalization used in audio mastering.

Listeners who prefer to experience music as it was intended to be heard on the original compact disc may prefer to not use volume normalization, because the average volume of each track was set intentionally by a professional mastering engineer.

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 the audio track is stored in its metadata tag.

==Licensing and patent issues {{Anchor | patent}}==<!-- This section is linked from [[SUSE Linux]] -->

A large number of different organizations have claimed ownership of patents necessary to implement MP3 (decoding and/or encoding). These different claims have led to a number of legal actions, and legal threats, from a variety of sources, resulting in uncertainty about what is necessary to legally create MP3-supporting products with MP3 support in countries that permit software patents.

The various patents claimed to cover MP3 by different patent-holders have many different expiration dates, ranging from 2007 to 2017 in the U.S.<ref>{{cite web|title= Big List of MP3 Patents (and supposed expiration dates)|url=http://www.tunequest.org/a-big-list-of-mp3-patents/20070226/|author=tunequest|date=2007-02-26}}</ref>

[[Thomson SA|Thomson Consumer Electronics]] claims to control MP3 licensing of the [http://www.mp3licensing.com/patents/index.html MPEG-1/2 Layer 3 patents] in many countries, including the [[United States]], [[Japan]], [[Canada]] and EU countries.<ref>{{cite web |title=Acoustic Data Compression -- MP3 Base Patent |publisher=Foundation for a Free Information Infrastructure |date=January 15, 2005 |url=http://eupat.ffii.org/patents/samples/ep287578/index.en.html |accessdate=2007-07-24 }}</ref> Thomson has been actively enforcing these patents.

For current information about [[Fraunhofer Society|Fraunhofer IIS]] and Thomson's [[patent portfolio]] and licensing terms and fees see their website [http://www.mp3licensing.com/ mp3licensing.com]. MP3 license revenues generated ca. 100 million Euro revenue to the Fraunhofer Society in 2005.<ref>{{cite web |author=Muzinée Kistenfeger |title=The Fraunhofer Society (Fraunhofer-Gesellschaft, FhG) |publisher=British Consulate-General Munich |date=May, 2006 |url=http://www.britischebotschaft.de/en/embassy/r&t/notes/rt-fs005_Fraunhofer.html |accessdate=2007-07-24 }}</ref>

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."<ref>{{cite web |title=Early MP3 Patent Enforcement |publisher=Chilling Effects Clearinghouse |date=September 1, 1998 |url=http://www.chillingeffects.org/patent/notice.cgi?NoticeID=464 |accessdate=2007-07-24 }}</ref>

These patent issues significantly slowed the development of unlicensed MP3 software {{Fact|date=February 2007}} and led to increased focus on creating and popularizing alternatives such as [[Vorbis]], [[Advanced Audio Coding|AAC]], and [[Windows Media Audio|WMA]]. [[Microsoft]] chose to move away from MP3 to its own proprietary [[Windows Media]] format to avoid licensing issues associated with these patents.{{Fact|date=February 2007}} Until the key patents expires, unlicensed encoders and players could be [[patent infringement|infringing]] in countries where the patents are valid.

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,
* 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 [[Digital Rights Management|DRM]] restrictions, which makes MP3 files easy to edit, copy and play in different portable digital players ([[Samsung]], [[Apple Inc.|Apple]], Creative, etc.),
* the majority of home users not knowing or not caring about the patents controversy, who often do not consider such legal issues in choosing their music format for personal use.

Additionally, patent holders declined to enforce license fees on [[free software|free]] and [[open source]] decoders, which allows many free MP3 decoders to develop.<ref>{{cite web |author=Glyn Moody |title=Should We Fight for Ogg Vorbis? |publisher=Linux Journal |date=June 15, 2007 |url=http://www.linuxjournal.com/node/1000238 |accessdate=2007-07-24 }}</ref> Furthermore, while attempts have been made to discourage distribution of encoder binaries, Thomson has stated that individuals who use free MP3 encoders are [http://www.webcitation.org/5MeUrGbFN not required to pay fees]. Thus, while patent fees have been an issue for companies that attempt to use MP3, they have not meaningfully impacted users, which allows the format to grow in popularity.

[http://www.sisvel.com Sisvel S.p.A.] and its U.S. subsidiary [http://www.audiompeg.com Audio MPEG, Inc.] previously sued Thomson for patent infringement on MP3 technology,<ref>{{cite web |title=Audio MPEG and Sisvel: Thomson sued for patent infringement in Europe and the United States - MP3 players stopped by customs |publisher=ZDNet India |date=October 6, 2005 |url=http://www.zdnetindia.com/news/pressreleases/stories/128960.html |accessdate=2007-07-24 }}</ref> but those disputes were resolved in November 2005 with Sisvel granting Thomson a license to their patents. Motorola also recently signed with Audio MPEG to license MP3-related patents.

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 licencing rights. The injunction was later reversed by a Berlin judge;<ref>{{cite web |author=Erica Ogg |title=SanDisk MP3 seizure order overturned |publisher=CNET News.com |date=September 7, 2006 |url=http://news.com.com/2100-1047_3-6113326.html |accessdate=2007-07-24 }}</ref> 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.<ref>{{cite web |title=Sisvel brings Patent Wild West into Germany |publisher=IPEG blog |date=September 7, 2006 |url=http://ipgeek.blogspot.com/2006/09/sisvels-brings-patent-wild-west-into.html |accessdate=2007-07-24 }}</ref>

On [[February 16]] [[2007]], Texas MP3 Technologies sued Apple, Samsung Electronics and Sandisk with a patent-infringement lawsuit regarding portable MP3 players. The suit was filed in [[United States District Court for the Eastern District of Texas|Marshall, Texas]]; this is a common location for patent infringement suits due to speedy trials.
Texas MP3 Technologies claimed infringement with U.S. patent 7,065,417, awarded in June 2006 to multimedia chip-maker SigmaTel, covering "an MPEG portable sound reproducing system and a method for reproducing sound data compressed using the MPEG method."<ref>{{cite web|title=Texas MP3 Technologies claims the companies infringed its patent covering 'an MPEG portable sound reproducing system'|url=http://www.infoworld.com/article/07/02/26/HNmp3lawsuits_1.html|author=Martyn Williams|date=2007-02-26|publisher=IDG News Service}}</ref>

[[Alcatel-Lucent]] also claims ownership of several patents relating to MP3 encoding and compression, inherited from AT&T-Bell Labs. In November 2006, (prior to the companies' merger) Alcatel filed a lawsuit against [[Microsoft]] (see [[Alcatel-Lucent v. Microsoft]]), alleging infringement of seven of its patents. On [[February 23]] [[2007]] a San Diego court upheld the suit, and awarded [[Alcatel-Lucent]] a record-breaking US$1.52 billion in damages.<ref>{{cite web|title=BBC report of the Alcatel-Lucent lawsuit verdict: ''Microsoft faces $1.5bn MP3 payout''|url=http://news.bbc.co.uk/1/hi/business/6388273.stm|date=February 22, 2007|accessdate=2007-07-24}}</ref> [[Microsoft]] has said it will appeal the verdict, maintaining that the federal jury's decision is "unsupported by the law or facts", since [[Microsoft]] had already paid US$16 million to license the technology from [[Fraunhofer Society|Fraunhofer IIS]], which, it claims, is "the industry-recognized rightful licensor".<ref>{{cite web|title=Microsoft's Patent Disputes with Alcatel-Lucent, AT&T Make Waves|url=http://www.eweek.com/article2/0,1895,2098063,00.asp|author=Joe Wilcox|date=2007-02-23}}</ref>
A week later on [[March 2]], U.S. District Judge Rudi Brewster ruled from the bench in a related suit and dismissed all of Alcatel-Lucent's patents claims relating to speech recognition. Alcatel-Lucent plans to appeal the ruling.<ref>{{cite web|title=Microsoft wins in second Alcatel-Lucent patent suit|url=http://news.zdnet.com/2100-3513_22-6163828.html|author=Anne Broache|date=2007-03-02|publisher=CNET News.com, published on ZDNet news}}</ref>

In short, with Thomson, Fraunhofer IIS, Sisvel (and its U.S. subsidiary Audio MPEG), Texas MP3 Technologies, and Alcatel-Lucent all claiming legal control of relevant MP3 patents related to decoders, the legal status of MP3 remains unclear in countries where those patents are valid.

==Alternative technologies==
{{main|List of codecs}}
Many other lossy and lossless audio [[codec]]s exist. Among these, [[mp3PRO]], [[Advanced Audio Coding|AAC]], and [[MP2]] are all members of the same technological family as MP3 and depend on roughly similar [[psychoacoustic model]]s. The [[Fraunhofer Gesellschaft]] owns many of the basic [[patent]]s underlying these codecs as well, with others held by [[Dolby Labs]], [[Sony]], [[Thomson Consumer Electronics]], and [[AT&T]].

A test at a very low bit rate of 32 kbit/s<ref name="listening-test-32-2004" /> showed that MP3 was significantly worse than the more modern codecs at that lower bit rate.

==See also==
<div style="-moz-column-count:2; column-count:2;">
*[[Comparison of audio codecs]]
*[[Copyright infringement]]
*[[Digital audio player]]
*[[ID3]]
*[[Joint stereo]]
*[[LRC (file format)]]
*[[Media player]]
*[[MP3 blog]]
*[[MP3 Surround]]
*[[Streaming Media]]
*[[DJ digital controller]]
*[[Advanced Audio Coding|AAC]]
*[[Ogg Vorbis]]
</div>

==References==
{{reflist|2}}

==External links==
* [http://www.iis.fraunhofer.de/fhg/iis/EN/bf/amm/index.jsp Fraunhofer IIS]
* [http://www.iis.fraunhofer.de/fhg/iis/EN/bf/amm/mp3history/mp3history01.jsp The Story of MP3] — How MP3 was invented, by Fraunhofer IIS
* [http://www.mp3licensing.com/patents/ List of relevant patents]
* [http://mp3licensing.com/help/index.html Thomson Licensing FAQ]
* [http://www.all4mp3.com all4mp3.com: Site with more information concerning ongoing development and innovation, including MP3Surround.]
{{Compression Formats}}

[[Category:Audio codecs]]
[[Category:MPEG]]
[[Category:MP3]]
[[Category:Thomson SA]]
[[Category:1994 introductions]]


Barney Rubble raped my wife.
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Revision as of 09:06, 11 March 2008

MPEG-1 Audio Layer 3
Filename extension
mp3
Internet media type
audio/mpeg
Magic numberID3
Type of formatAudio

MPEG-1 Audio Layer 3, more commonly referred to as niggers, is a digital audio encoding format using a form of intelligence lossy data compression.

File:Pyramidhead.png

This encoding format is used to create the MP3's small file, as a way to store a single segment of audio, commonly a song, such that the file can be easily organized and transferred between computers or other devices such as MP3 players.

MP3's use of a god damn bullshit compression algorithm is designed to greatly increase the amount of cum required to represent the audio recording and still sound like a faithful to marriage of sandwich nigger reproduction of the original uncompressed audio for some listeners, but is not considered High-fi Fidelity audio by the elite connoisseuwhateverthefuckthiswordis. An MP3 file is shittier than SID format. That is created using the mid-range bittorrentclients setting of 1280000000 gaybit/s will result in a file that is typically about 1/10th the size of the CD file created from the original audio source. An MP3 file can also be constructed at higher or lower braces, with higher or lower resulting quality.

Barney Rubble raped my wife.

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