MP3 is a popular digital audio encoding and lossy 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.





MP3 is a lossy compression format. It provides a representation of 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 rates, 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 and 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 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 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 rate depends on the bit depth and sampling rate of the input signal. Nevertheless, there are often published compression rates that use the 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 drives 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 OS).


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, 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 lawsuit against Napster to protect their Copyrights.


Commercial online music distribution services (like the iTunes Music Store) usually prefer other/proprietary music file formats that support Digital Rights Management (DRM) to control and restrict the use of digital music. This preference is most likely chosen in an attempt to prevent piracy of copyright protected materials, but most users with at least an intermediate understanding of computers will know that it's just a matter of time before someone else makes it easy to convert such proprietary file formats.


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 bits of encoded data that are used to represent each second of audio. Typically rates chosen are between 128 and 256 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.


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 6:1...8:1

  • Layer 3: 112...128 kbit/s, compression 10:1...12:1

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 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 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 compression algorithms, currently capable of compressing 16-bit PCM audio to 38% while leaving the audio identical to the original, such as Lossless Audio LA, Apple Lossless, FLAC, Windows Media Audio 9 Lossless (wma) and 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 kibit/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 discs), 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 Kibit/s


Variable bit rates (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.



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 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 MDCT representation in the frequency domain). This is the domain of psychoacoustics, which aims at understanding how human acoustical perception works (both in our ears 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 decoders are "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 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 file format (see 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 discs 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 codecs exist, including:

  • MPEG-1/2 Audio Layer 2 (MP2), MP3's predecessor;

  • Ogg Vorbis from the Foundation, a free software and patent free codec.

  • MPC, also known as Musepack (formerly MP+), a derivative of MP2;

  • mp3PRO from Thomson Multimedia combining MP3 with SBR;

  • AC-3, used in Dolby Digital and DVD;

  • ATRAC, used in Sony's Minidisc;

  • MPEG-4 AAC, used by Apple's iTunes Music Store and iPod

  • Windows Media Audio (WMA) from Microsoft.

  • QDesign, used in QuickTime at low bitrates;

  • 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 models. The Fraunhofer Gesellschaft owns many of the basic patents underlying these codecs, with Dolby Labs, Sony, Thomson Consumer Electronics, and AT&T holding other key patents.


There are also some 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 tests have attempted to find the best-quality lossy audio codecs at certain bitrates. The tests have suggested that for some audio samples, newer audio codecs including Ogg Vorbis, mp3PRO, AC-3, Windows Media Audio, MPC and RealAudio perform better than MP3. Generally, these codecs achieve the equivalent of MP3 128kbit/s at around 80kbit/s. At 128kbit/s, Ogg Vorbis and MPC performed marginally better than other codecs. At 64kbit/s, AAC 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 192kbit/s 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 Consumer Electronics controls licensing of the MPEG-1/2 Layer 3 patents in countries that recognize software patents, including the United States and Japan, but not 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 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 here and here.

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, 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.

Sisvel S.p.A. and Audio MPEG, Inc., are suing Thomson for patent infringement on MP3 technology. Audio MPEG also starts licensing MP3 to vendors of MP3, so legal status of MP3 is unclear.



Online music resources


Tools such as iRate try to make it easier to find music that matches the listener's tastes. There are several online music stores. Apple's iTunes store is presently the most popular commercial online music offering. Independent artists are able to use smaller sites to provide distribution. A controversial MP3 portal is the Russian site, which through their country's copyright laws can legally distribute music by any label or artist. The music industry has closed down many file sharing networks and the publics urge for free mp3 downloads has made the way for sites as and Erik Brown's MP3 Links that list links to free legal mp3 download sites.


There are also several online columnists who edit news sites focused on digital music and the grassroots community it spawned. They include Richard Menta's MP3 Newswire, an early MP3 news site started in 1998, Jon Newton's P2Pnet, and Thomas Mennecke's Other sites like and which allow artists to choose to post their own music for free download.





















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