MPEG-1 Audio Layer 3
File extension:.mp3
MIME type:audio/mpeg
Type of format:Audio
MPEG-1 Audio Layer 3, more commonly referred to as MP3, is an audio encoding format.

It uses a lossy compression algorithm that is designed to greatly reduce the amount of data required to represent the audio recording, yet still sound like a faithful reproduction of the original uncompressed audio to most listeners. It was invented by a team of European engineers at Philips, CCETT (Centre commun d'études de télévision et télécommunications), IRT and Fraunhofer Society, who worked in the framework of the EUREKA 147 DAB digital radio research program, and it became an ISO/IEC standard in 1991.

MP3 is an audio-specific format. The compression removes certain parts of sound that are outside the hearing range of most people. It provides a representation of pulse-code modulation — encoded audio in much less space than straightforward methods, by using psychoacoustic models to discard components less audible to human hearing, and recording the remaining information in an efficient manner. Similar principles are used by JPEG, an image compression format.


The psychoacoustic masking codec was first proposed, apparently independently in 1979, by Manfred Schroeder, et al.[1] in Germany and M. A.Krasner[2] 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 coder 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 immediate influence in European and specifically German circles of acoustic and source-coding (audio compression) research. Both Krasner and Schroeder built upon the work of E. F. Zwicker.[3]

The immediate predecessor of MP3, and the first practical implementation in hardware (Krasner's hardware was too cumbersome and slow for practical use), was "Optimum Coding in the Frequency Domain",[4] which was an implementation of a psychoacoustic transform coder based on Motorola 56000 DSP chips. MP3 is directly descended from OCF. MP3 represents the outcome of the collaboration of Dr. Karl Heinz Brandenburg with the Fraunhofer Society for Integrated Circuits, Erlangen, with relatively minor contributions from the Musicam (MP2) branch of psychoacoustic sub-band coders.

Modern lossy bit compression technologies, including MPEG and MP3, are based on the early work of Prof Oscar Bonello of the University of Buenos Aires, Argentina. He was involved in studio equipment design for broadcast radio automation. At the same time he taught acoustics at the University (he is the author of the "Bonello Criterion" for room acoustics design), with psychoacoustics being his main field of research. In 1983, he started researching the idea of using the Critical Band Masking principle (a property of the ear) in order to reduce the bit stream needed to encode an audio signal. The masking principle was discovered in 1924 and further developed by Egan-Hake and Richard Ehmer in 1959. Bonello's work created, in 1987, the world's first bit compression system, named ECAM, working in real time and implemented by hardware on an IBM PC computer. This plug in card and the associated control software was demonstrated for the first time in 1988 as a fully working product named Audicom and introduced to the world at the international NAB Radio Exhibition in Atlanta, USA in 1990. The basic Bonello implementation is now used in MP3 and other systems. Bonello refuses to apply for any patents around this technology.[5][6]

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).[7]

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) 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 key to settle the 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 Mussmann (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) 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 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 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 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 softness and simplicity, 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.

Going public

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) the technology was essential to store non-instrument based (see tracker and MIDI) music for listening on a computer.


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 22, 1994 (MAPlay was also ported to Microsoft Windows).

Initially the only encoder available for MP2 production was the Xing Encoder, accompanied by the program cdda2wav, a CD ripper used for extracting CD audio tracks to Waveform Audio 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.


In 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), the Unix audio player mpg123 and the peer-to-peer file sharing network Napster (released in 1999). These programs made it simple for average users to play back, create, share and collect MP3s.

The small size of MP3 files has enabled widespread peer-to-peer file sharing of music, which would previously have been nearly impossible. The major record companies, who argue that such free sharing of music reduces sales, reacted to this by pursuing law-suits against Napster, which was eventually closed down, and eventually against individual users who engaged in file sharing. Napster has now returned, albeit in a slightly different form. These legal actions have had little effect on the production and distribution of MP3 audio.

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 authorised 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. Some services, such as eMusic, continue to offer the MP3 format, which allows users to play back their music on virtually any device.

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 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. (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, which is in widespread use for encoding at higher bit rates) is not necessarily as good as other, lower bit rates.

Decoding audio

Decoding, on the other hand, is carefully defined in the standard. Most decoders are "bitstream compliant", meaning that the decompressed 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 ISO/IEC 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, comparison of decoders is usually based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process).

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 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 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 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,[8] 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 kbit/s was somewhere between annoying and acceptable with older encoders, modern MP3 encoders can provide very good quality at those bit rates[9] (January 2006). However, in 1998, MP3 at 128 kbit/s was only providing quality equivalent to AAC-LC at 96 kbit/s and MP2 at 192 kbit/s.[10]

The transparency threshold of MP3 can be estimated to be at about 128 kbit/s with good encoders on typical music as evidenced by its strong performance in the above test, however some particularly difficult material can require 192 kbit/s or higher. As with all lossy formats, some samples can not be encoded to be transparent for all users.

For digital stereophonic sounds, this transparency threshold of MP3 can be greatly reduced by using the Joint stereo coding mode based on stereo intensity redundancy removal. This feature further reduces the overall bit rate of a stereophonic sound down to 96 kbit/s. Unfortunately, in spite of a wide use of this feature in most MP3 files and all standardized encoders no official results of this transparency level were ever published due to strong lobbying and opposition of the professional music industry.

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 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,[11] the LAME MP3 encoder scored only 1.79/5 — behind all modern encoders — with Nero Digital HE AAC scoring 3.30/5.

It is also important to note that perceived quality can be influenced by listening environment (ambient noise), listener attention, and listener training.

Bit rate

Several bit rates are specified in the MPEG-1 Layer 3 standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 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 since 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 since it typically offers very good audio quality in a relatively small space. 192 kbit/s is often used by those who notice artifacts at lower bit rates. By contrast, uncompressed audio as stored on a compact disc has a bit rate of 1,411.2 kbit/s (16 bits/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, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 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, but 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.[12]

File structure

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 independent items: one can cut the frames from a file and an MP3 player would be able to play it. 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 being 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; this is also shown in the diagram.

Design limitations

There are several limitations inherent to the MP3 format that can not be overcome by any MP3 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:
  • Bit rate is limited to a maximum of 320 kbit/s (while some encoders can create higher bit rates, there is little-to-no support for these higher bit rate mp3s)
  • 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 filterbank (Layer II). Pre-echo is concealed due to the specific time-domain characteristics of the filter.
  • 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 on a frame-to-frame basis
  • 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.
Nevertheless, a well-tuned MP3 encoder can perform competitively even with these restrictions.

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 ID3v1 and ID3v2, and the more recently introduced APEv2.

APEv2 was originally developed for the MPC file format (see 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 editors dedicated to the purpose.

Volume normalization

As compact discs 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 audio track is stored in the metadata tag.

One can download audio converting software to change the formats.

Licensing and patent issues

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.[13]

Thomson Consumer Electronics claims to control MP3 licensing of the MPEG-1/2 Layer 3 patents in many countries, including the United States, Japan, Canada and EU countries.[14] Thomson has been actively enforcing these patents.

For current information about Fraunhofer IIS and Thomson's patent portfolio and licensing terms and fees see their website MP3 license revenues generated ca. 100 million Euro revenue to the Fraunhofer Society in 2005.[15]

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."[16]

These patent issues significantly slowed the development of unlicensed MP3 software and led to increased focus on creating and popularizing alternatives such as Vorbis, AAC, and WMA. 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, unlicensed encoders and players appear to be infringing articles in countries that recognize those patents.

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 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 and open source decoders, allowing many free MP3 decoders to develop.[17] Furthermore, while attempts have been made to discourage distribution of encoder binaries, Thomson has stated that individuals using free MP3 encoders are not required to pay fees. Thus while patent fees have been an issue for companies attempting to use MP3, they have not meaningfully impacted users, allowing the format to grow in popularity.

Sisvel S.p.A. and its U.S. subsidiary Audio MPEG, Inc. previously sued Thomson for patent infringement on MP3 technology,[18] 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;[19] 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.[20]

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."[21]

Alcatel-Lucent also claims ownership of several patents relating to MP3 encoding and compression. 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.[22] 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 IIS, which, it claims, is "the industry-recognized rightful licensor".[23] 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.[24]

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 that permit software patents.

Alternative technologies

Main article: List of codecs
Many other lossy and lossless audio codecs exist. Among these, mp3PRO, AAC, and MP2 are all members of the same technological family as MP3 and depend on roughly similar psychoacoustic models. The Fraunhofer Gesellschaft owns many of the basic patents underlying these codecs as well, with others held by Dolby Labs, Sony, Thomson Consumer Electronics, and AT&T.

In a 2005 listening test<ref name="listening-test-128-2006" /> that compared the performance of the LAME MP3 encoder against more modern compression formats at 128 kbit/s, it was found that there was no statistically significant difference between the results for LAME, Vorbis, several AAC encoders, and WMA. However, 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


1. ^ "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
2. ^ "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
3. ^ "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
4. ^ "OCF: Coding High Quality Audio with Data Rates of 64 KBit/sec; K. Brandenburg, D. Seitzer; Universitaet Erlangen-Nuernberg, Erlangen; Presented at the 85th Convention of the Audio Engineering Society; Los Angeles; November 3-6 1988
5. ^ Masking by Tones vs Noise Bands Richard Ehmer ASA Journal, Vol 3, Number 9, September 1959
6. ^ The invention of Audicom: Summary of some of Solidyne's contributions to Broadcast Engineering
7. ^ Jack Ewing (March 5, 2007). How MP3 Was Born. Retrieved on 2007-07-24.
8. ^ Amorim, Roberto (2003-08-03), Results of 128kbps Extension Public Listening Test, <[1] (retrieved on 2007-03-17)
9. ^ Mares, Sebastian (2006–01), Results of Public, Multiformat Listening Test @ 128 kbps, <[2] (retrieved on 2007-03-17)
10. ^ David Meares, Kaoru Watanabe & Eric Scheirer (1998–02). "Report on the MPEG-2 AAC Stereo Verification Tests" (PDF). International Organisation for Standardisation. Retrieved on 2007-03-17.
11. ^ Amorim, Roberto (2004-07-11), Results of Dial-up bit rate public Listening Test, <[3] (retrieved on 2007-03-17)
12. ^ Bouvigne, Gabriel (2006-11-28), freeformat at 640 kbps and foobar2000, possibilities?, <[4] (retrieved on 2007-03-17)
13. ^ tunequest (2007-02-26). Big List of MP3 Patents (and supposed expiration dates).
14. ^ Acoustic Data Compression -- MP3 Base Patent. Foundation for a Free Information Infrastructure (January 15, 2005). Retrieved on 2007-07-24.
15. ^ Muzinée Kistenfeger (May, 2006). The Fraunhofer Society (Fraunhofer-Gesellschaft, FhG). British Consulate-General Munich. Retrieved on 2007-07-24.
16. ^ Early MP3 Patent Enforcement. Chilling Effects Clearinghouse (September 1, 1998). Retrieved on 2007-07-24.
17. ^ Glyn Moody (June 15, 2007). Should We Fight for Ogg Vorbis?. Linux Journal. Retrieved on 2007-07-24.
18. ^ Audio MPEG and Sisvel: Thomson sued for patent infringement in Europe and the United States - MP3 players stopped by customs. ZDNet India (October 6, 2005). Retrieved on 2007-07-24.
19. ^ Erica Ogg (September 7, 2006). SanDisk MP3 seizure order overturned. CNET Retrieved on 2007-07-24.
20. ^ Sisvel brings Patent Wild West into Germany. IPEG blog (September 7, 2006). Retrieved on 2007-07-24.
21. ^ Martyn Williams (2007-02-26). Texas MP3 Technologies claims the companies infringed its patent covering 'an MPEG portable sound reproducing system'. IDG News Service.
22. ^ BBC report of the Alcatel-Lucent lawsuit verdict: Microsoft faces $1.5bn MP3 payout (February 22, 2007). Retrieved on 2007-07-24.
23. ^ Joe Wilcox (2007-02-23). Microsoft's Patent Disputes with Alcatel-Lucent, AT&T Make Waves.
24. ^ Anne Broache (2007-03-02). Microsoft wins in second Alcatel-Lucent patent suit. CNET, published on ZDNet news.

External links

A filename extension is a suffix to the name of a computer file applied to indicate its type. It is commonly used to infer information about what sort of data might be stored in the file.
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Mime or pantomime is a theatrical medium or performance art, involving the acting out of a story by a mime artist through body motions, without use of speech.


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MPEG-1 defines a group of Audio and Video (AV) coding and compression standards agreed upon by MPEG (Moving Picture Experts Group). MPEG-1 video is used by the Video CD (VCD) format and less commonly by the DVD-Video format.
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Sound recording and reproduction is the electrical or mechanical inscription and re-creation of sound waves, usually used for the voice or for music.

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lossy compression method is one where compressing data and then decompressing it retrieves data that may well be different from the original, but is close enough to be useful in some way.
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Audio compression can mean two things:
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The Institut für Rundfunktechnik GmbH (IRT) is the research centre of the German broadcasters (ARD / ZDF / DLR), Austria's broadcaster (ORF) and the Swiss public broadcaster (SRG/SSR).
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The Fraunhofer Society (German: Fraunhofer-Gesellschaft) is a German research organization with 56 institutes spread throughout Germany, each focusing on different fields of applied
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Digital Audio Broadcasting (DAB), also known as Eureka 147, is a technology for broadcasting of audio using digital radio transmission.

The original objectives of converting to digital transmission were to enable higher fidelity, more stations and more resistance to
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Digital Audio Broadcasting (DAB), also known as Eureka 147, is a technology for broadcasting of audio using digital radio transmission.

The original objectives of converting to digital transmission were to enable higher fidelity, more stations and more resistance to
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International Organization for Standardization (Organisation internationale de normalisation), widely known as ISO, is an international standard-setting body composed of representatives from various national standards organizations.
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The International Electrotechnical Commission[1] (IEC) is a not-for-profit, non-governmental international standards organization that prepares and publishes International Standards for all electrical, electronic and related technologies – collectively known
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19th century - 20th century - 21st century
1960s  1970s  1980s  - 1990s -  2000s  2010s  2020s
1988 1989 1990 - 1991 - 1992 1993 1994

Year 1991 (MCMXCI
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Pulse-code modulation (PCM) is a digital representation of an analog signal where the magnitude of the signal is sampled regularly at uniform intervals, then quantized to a series of symbols in a digital (usually binary) code.
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Psychoacoustics is the study of subjective human perception of sounds. Alternatively it can be described as the study of the psychological correlates of the physical parameters of acoustics.
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A photo of a flower compressed with successively more lossy compression ratios from left to right.
File extension: .jpeg, .jpg, .jpe
.jfif, .jfi, .

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MIT Lincoln Laboratory, also known as Lincoln Lab, is a federally funded research and development center managed by the Massachusetts Institute of Technology and primarily funded by the United States Department of Defense.
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A digital signal processor (DSP) is a specialized microprocessor designed specifically for digital signal processing, generally in real-time computing.

Characteristics of typical Digital Signal Processors

  • Designed for real-time processing

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Audicom is the name of the first PC audio system dedicated to broadcasting. Designed in the 80's by Solidyne included a compression audio card and the software for running it.
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MPEG-1 Audio Layer 2

File extension: .mp2
MIME type: audio/mpeg
Type of format: Audio

MPEG-1 Audio Layer II (MP2, sometimes Musicam) is an audio codec defined by ISO/IEC 11172-3.
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Digital Audio Broadcasting (DAB), also known as Eureka 147, is a technology for broadcasting of audio using digital radio transmission.

The original objectives of converting to digital transmission were to enable higher fidelity, more stations and more resistance to
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German Aerospace Center (DLR) (German: Deutsches Zentrum für Luft- und Raumfahrt e.V.) is the national research center for aviation and space flight of the Federal Republic of Germany and the German Space Agency. DLR is a member in the Helmholtz Association.
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"Das Lied der Deutschen" (third stanza)
also called "Einigkeit und Recht und Freiheit"
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EUREKA, often abbreviated as "E!", is a pan-european research and development funding and coordination organisation. EUREKA aims to coordinate efforts of governments and commercial companies. It does not partake in, for example, military research.
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Karlheinz Brandenburg (born June 20, 1954, in Erlangen, Germany) is an audio engineer who has contributed to the audio compression format MPEG Audio Layer 3, more commonly known as MP3.


He received a Dipl. Ing.
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19th century - 20th century - 21st century
1960s  1970s  1980s  - 1990s -  2000s  2010s  2020s
1988 1989 1990 - 1991 - 1992 1993 1994

Year 1991 (MCMXCI
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MPEG-1 Audio Layer 2

File extension: .mp2
MIME type: audio/mpeg
Type of format: Audio

MPEG-1 Audio Layer II (MP2, sometimes Musicam) is an audio codec defined by ISO/IEC 11172-3.
..... Click the link for more information.
Koninklijke Philips Electronics N.V. (Royal Philips Electronics)

Public (Euronext: PHIA , NYSE:  PHG )
Founded 1891 Eindhoven
Headquarters Amsterdam, the Netherlands

Key people Gerard Kleisterlee, CEO
Industry Electronics
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