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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. It is a common audio format for consumer audio storage, as well as a de facto standard of digital audio compression for the transfer and playback of music on digital audio players. MP3 is an audio-specific format that was designed by the Moving Picture Experts Group. The group was formed by several teams of engineers at Fraunhofer IIS in Erlangen, Germany, AT&T-Bell Labs in Murray Hill, NJ, USA, Thomson-Brandt, and CCETT as well as others. It was approved as an ISO/IEC standard in 1991.

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, but is not considered high fidelity audio by audiophiles. An MP3 file that is created using the mid-range bit rate setting of 128 kbit/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 bit rates, with higher or lower resulting quality. 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. It internally 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 JPEG, an image compression format. The term MP3 is actually a reference shortened from the official Moving Picture Experts Group-1 Audio Layer 3 nomenclature for this lossy digital audio encoding format. The MP3 audio data compression lossy data compression algorithm takes advantage of a perceptual limitation of human hearing called auditory masking. In 1894, Mayer reported that a tone could be rendered inaudible by another tone of lower frequency. In 1959, Richard Ehmer described a complete set of auditory curves regarding this phenomenon. Ernst Terhardt et al. created an algorithm describing auditory masking with high accuracy. This work added on a variety of reports from authors dating back to Fletcher, and to the work that initially determined critical ratios and critical bandwidths.

The psychoacoustic masking codec was first proposed in 1979, apparently independently, by Manfred R. Schroeder, et al.. from AT&T-Bell Labs in Murray Hill, NJ, and M. A.Krasner both in the United States. Krasner was the first to publish and to produce hardware for speech, not usable as music bit compression, but the publication of his results as a relatively 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, and his paper had influence in acoustic and source-coding (audio data compression) research. Both Krasner and Schroeder built upon the work performed by Eberhard F. Zwicker in the areas of tuning and masking of critical bands, that in turn built on the fundamental research in the area from Bell Labs of Harvey Fletcher and his collaborators. A wide variety of (mostly perceptual) audio compression algorithms were reported in IEEE's refereed Journal on Selected Areas in Communications. That journal reported in February 1988 on a wide range of established, working audio bit compression technologies, some of them using auditory masking as part of their fundamental design, and several showing real-time hardware implementations aimed at laboratory experiences. This hardware was never used in PC audio cards. The immediate predecessors of MP3 were "Optimum Coding in the Frequency Domain" (OCF), and Perceptual Transform Coding (PXFM). These two codecs, along with block-switching contributions from Thomson-Brandt, were merged into a codec called ASPEC, which was submitted to MPEG, and which won the quality competition, but that was mistakenly rejected as too complex to implement. The first practical implementation of an audio perceptual coder (OCF) in hardware (Krasner's hardware was too cumbersome and slow for practical use), was an implementation of a psychoacoustic transform coder based on Motorola 56000 DSP chips. 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 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. The European Community financed this project, commonly known as EU-147, from 1987 to 1994 as a part of the EUREKA research program.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).

In 1991 there were two proposals available: Musicam 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. 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). 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 were approved in 1991 and finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3, 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 ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. They may use the 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 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, 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) as discussed in Brian C. J. Moore's book on the Psychology of Human Hearing, for instance.

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 (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 drives back in that time (~ 500 MB) lossy compression was essential to store non-instrument based (see tracker and MIDI) music for playback on computer.

From the first half of 1994 through the late 1990s, 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), and the Unix audio player mpg123. In 1998, the Rio PMP300, one of the first portable MP3 players was released, despite legal efforts by the RIAA. In November 1997, the website mp3.com was offering thousands of MP3s for free. The small size of MP3 files enabled widespread peer-to-peer file sharing of music ripped from compact discs, 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 argue that this free sharing of music reduces sales, and call it "music piracy". They reacted by pursuing lawsuits against Napster (which was eventually shut down) and eventually against individual users who engaged in file sharing.

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. Also, there are software and hardware solutions available that allow the user to record anything they can play.

Unauthorized MP3 file sharing continues on next-generation peer-to-peer networks. Some authorized services, such as Beatport, Bleep, Xiie.net, Juno Records, eMusic, and Amazon.com have begun selling 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. At the 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 in the raw audio (or rather its 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, 192 samples are taken instead of 576. This is done to limit the temporal spread of quantization noise accompanying the transient. 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 lower bit rates.

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 encoded data correctly.

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, as in when ripping a compact disc to MP3 format. Using a lower bit rate provides a relatively lower audio quality and produces a smaller file size. Likewise, using a higher bit rate outputs a higher quality audio, but also results in a larger file. 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 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, 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. However, in 1998, MP3 at 128 kbit/s was only providing quality equivalent to HE-AAC at 64 kbit/s and MP2 at 192 kbit/s.

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 not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate. 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, the LAME MP3 encoder scored only 1.79/5 — behind all modern encoders — with 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). 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.Others have reached the same conclusion, and some record producers have begun to mix music specifically to be heard on iPods and mobile phones.

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 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, because 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 bit rate files are slowly being replaced with higher bit rates like 192 and 256 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 because file sizes at higher bit rates approach those of lossless codecs such as FLAC. By contrast, uncompressed audio as stored on a compact disc has a bit rate of 1,411.2 kbit/s (16 bit/sample × 44100 samples/second × 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 kbit/s and sample rates of 8, 11.025, 12, 16, 22.05 and 24 kHz.

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. 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. Frames are not independent items ("byte reservoir") and therefore cannot 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 Vorbis, WMA Pro and 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 and may cause smearing of percussive sounds
* Due to the tree structure of the filter bank, pre-echo problems are made worse, as the combined impulse response of the two filter banks does not, and cannot, provide an optimum solution in time/frequency resolution
* 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
* Frequency resolution is limited by the small long block window size, which decreases coding efficiency
* There is 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 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.

Since volume levels of different audio sources can vary greatly, it is sometimes desirable to adjust the playback volume of audio files such that a consistent average volume is perceived. The idea is to control the average volume across multiple files, not the volume peaks in a single file. This gain normalization, while similar in purpose, is distinct from dynamic range compression (DRC), which is a form of normalization used in audio mastering. Gain normalization may defeat the intent of recording artists and audio engineers who deliberately set the volume levels of the audio they recorded. A few standards for storing the average volume of an MP3 file in its metadata tags, enabling a specially designed player to automatically adjust the overall playback volume for each file, have been proposed. A popular and widely implemented such proposal is "Replay Gain", which is not MP3-specific. When used in MP3s, it is stored differently by different encoders, and as of 2008, Replay Gain-aware players don't yet support all formats.

A large number of different organizations have claimed ownership of patents related to MP3 decoding or encoding. These different 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 various patents claimed to cover MP3 by different patent-holders have many different expiration dates, ranging from 2007 to 2017 in the U.S. The initial near-complete MPEG-1 standard (parts 1,2,3) was publicly available in December 6, 1991 as ISO CD 11172. Since US patents must be filed no later than a year after publication, some of the later patents are questionable, and MP3 may be patent free in the US by December of 2012. 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. Thomson has been actively enforcing these patents.

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 licensed MP3 software and led to increased focus on creating and popularizing alternatives such as Vorbis, AAC, and WMA. Microsoft chose to move away from MP3 to its own proprietary Windows Media format to avoid licensing issues associated with these patents. Until the key patents expire, unlicensed encoders and players could be 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 effects 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 DRM restrictions, which makes MP3 files easy to edit, copy and play in different portable digital players (Samsung, Apple, Creative, etc.),
* the majority of home users not knowing or not caring about the patent's 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 and open source decoders, which allows many free MP3 decoders to develop. Furthermore, while attempts have been made to discourage distribution of encoder binaries, Thomson has stated that individuals who use free MP3 encoders are not allowed to license MP3. 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. Sisvel S.p.A. and its U.S. subsidiary Audio MPEG, Inc. previously sued Thomson for patent infringement on MP3 technology, 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 licensing rights. The injunction was later reversed by a Berlin judge; but that reversal was in turn blocked the same day by another judge from the same court, "bringing the Patent Wild West to Germany" in the words of one commentator.

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 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."