MP3

MPEG-1 Audio Layer 3
File name extension	.mp3
Internet media type	audio/mpeg
Type of format	Audio

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 encoding for the transfer and playback of music on digital audio players.

MP3 is an audio-specific format that was co-designed by several teams of engineers at Fraunhofer IIS in Erlangen, Germany, AT&T-Bell Labs in Murray Hill, NJ, USA, Thomson-Brandt, and CCETT. It was approved as an ISO/IEC standard in 1991.

MP3's use 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 most audiophiles. An MP3 file that is created using the mid-range bitrate setting of 128 kbit/s will result in a file that is typically about 1/10th the size of the CD file created from the original audio source. An MP3 file can also be constructed at higher or lower bitrates, with higher or lower resulting quality. 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. ^[1] 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.

Contents 1 History 1.1 Development 1.2 Going public 1.3 Internet 2 Encoding audio 3 Decoding audio 4 Audio quality 5 Bit rate 6 File structure 7 Design limitations 8 ID3 and other tags 9 Volume normalization 10 Licensing and patent issues 11 Alternative technologies 12 See also 13 References 14 External links

History

Development

The psycho-acoustic masking codec was first proposed, apparently independently in 1979, by Manfred Schroeder, et al.^[2] from AT&T-Bell Labs in Murray Hill, NJ, and M. A.Krasner^[3] both in the United States. Krasner was the first to publish and to produce hardware, but the publication of his results as a relatively obscure Lincoln Laboratory Technical Report did not immediately influence the mainstream of psychoacoustic codec development. Manfred Schroeder was already a well known and revered figure in the world wide community of acoustical and electrical engineers and his paper had influence in acoustic and source-coding (audio compression) research. Both Krasner and Schroeder built upon the work of E. F. Zwicker ^[4], that in turn built on the fundamental research in the area from Bell Labs of Harvey Fletcher and his collaborators. ^[5] A wide variety of audio compression algorithms, mostly (but not completely) perceptual were reported in a refereed journal, the Journal on Selected Areas in Communications, ^[6]. That journal reported in Feb. 1988 on a wide range of established, working audio bit compression technologies, most of them using auditory masking as part of their fundamental design, and several showing real-time hardware implementations.

The immediate predecessors of MP3 were "Optimum Coding in the Frequency Domain" (OCF),^[7] and Perceptual Transform Coding (PXFM).^[8] 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. This project was financed by the European Community 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).^[9]

In 1991, there were two proposals available: Musicam, and ASPEC - (Short excerpt on German Wikipedia) (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. ^[10] 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 filterbank from Layer 2, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 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.

Going public

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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 a computer.

Internet

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

The ease of creating and sharing MP3s resulted in widespread copyright 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 MP3, online music retailers often use other proprietary formats that are encrypted (known as Digital rights management) to prevent users from using purchased music in ways not specifically authorized by the record companies. The record companies argue that this is necessary to prevent the files from being made available on peer-to-peer file sharing networks. However, this has other side effects such as preventing users from playing back their purchased music on different types of devices. The audio content of these files can be converted into an unencrypted format, however, because often the user permissions include "burn to audio CD". And even when that option is not available, many sound cards allow the user to record anything they play. Unauthorized MP3 filesharing continues on next-generation peer-to-peer networks, though some authorized services, such as eMusic, and Amazon.com sell unrestricted music in the MP3 format.

Encoding audio

The MPEG-1 standard does not include a precise specification for an MP3 encoder. 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) is not necessarily as good at lower bit rates.

Decoding audio

Decoding, on the other hand, is carefully defined in the standard. Most decoders are "bitstream compliant", which means that the decompressed output - that they produce from a given MP3 file - will be the same (within a specified degree of rounding tolerance) as the output specified mathematically in the ISO/IEC standard document. The MP3 file has a standard format, which is a frame that consists of 384, 576, or 1152 samples (depends on MPEG version and layer), and all the frames have associated header information (32 bits) and side information (9, 17, or 32 bytes, depending on MPEG version and stereo/mono). The header and side information help the decoder to decode the associated Huffman 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 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, 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 the quality, and therefore, larger the resulting file will be.

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,^[11] 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 adequate quality at those bit rates^[12] (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.^[13]

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, or music encoded for the use of people with more sensitive hearing can require 192 kbit/s or higher. As with all lossy formats, some samples cannot be encoded to be transparent for all users.

The simplest type of MP3 file uses one bit rate for the entire file — this is known as Constant 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,^[14] 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).

Bit rate

Several bit rates are specified in the MPEG-1 Layer 3 standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 192, 224, 256 and 320 kbit/s, and the available sampling frequencies are 32, 44.1 and 48 kHz. A sample rate of 44.1 kHz is almost always used, because this is also used for 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 bitrate files are slowly being replaced with higher bitrates like 192 kbit/s, with some being encoded up to MP3's maximum of 320 kbit/s. It is unlikely that higher bit rates will be popular with any lossy audio codec as higher bit rates than 320 kbit/s encroach on the domain of lossless codecs such as FLAC.

By contrast, uncompressed audio as stored on a compact disc has a bit rate of 1,411.2 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, 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.^[15]

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 not independent items ("byte reservoir") and therefore cannot be extracted on arbitrary frame boundaries. The MP3 data is the actual audio payload. The diagram shows that the MP3 header consists of a sync word, which is used to identify the beginning of a valid frame. This is followed by a bit indicating that this is the MPEG standard and two bits that indicate that layer 3 is used; hence MPEG-1 Audio Layer 3 or MP3. After this, the values will differ, depending on the MP3 file. 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 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, may cause some smearing of percussive sounds.
• Due to the tree structure of the filter bank, pre-echo issues are made worse, as the combined impulse response of the two filter banks does not, and can not, provide an optimum solution in time/frequency resolution.
• The combination of the two filter banks creates aliasing issues that must be handled partially by the "aliasing compensation" stage, but that create excess energy to be coded in the frequency domain, thereby decreasing coding efficiency
• Frequency resolution is limited by the small long block window size, decreasing coding efficiency
• No scale factor band for frequencies above 15.5/15.8 kHz
• Joint stereo is done only on a frame-to-frame basis
• Internal handling of the bit reservoir increases encoding delay
• Encoder/decoder overall delay is not defined, which means lack of official provision for gapless playback. However, some encoders such as LAME can attach additional metadata that will allow players that are aware of it to deliver seamless playback.

ID3 and other tags

Main articles: ID3 and APEv2 tag

A "tag" in a compressed audio file is a section of the file that contains metadata such as the title, artist, album, track number or other information about the file's contents.

As of 2006, the most widespread standard tag formats are 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 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 (see filerename.co.uk for a free open source example).

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.

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

Licensing and patent issues

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 products with MP3 support in countries where those patents are valid.

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.^[16]

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.^[17] 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 mp3licensing.com. MP3 license revenues generated ca. 100 million Euro revenue to the Fraunhofer Society in 2005.^[18]

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."^[19]

These patent issues significantly slowed the development of unlicensed MP3 software^{[citation needed]} 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.^{[citation needed]} 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 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, which allows many free MP3 decoders to develop.^[20] Furthermore, while attempts have been made to discourage distribution of encoder binaries, Thomson has stated that individuals who use free MP3 encoders are not required to pay fees. Thus, while patent fees have been an issue for companies that attempt to use MP3, they have not meaningfully impacted users, which allows the format to grow in popularity.

Sisvel S.p.A. and its U.S. subsidiary Audio MPEG, Inc. previously sued Thomson for patent infringement on MP3 technology,^[21] 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;^[22] 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.^[23]

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."^[24]

Alcatel-Lucent also claims ownership of several patents relating to MP3 encoding and compression, inherited from AT&T-Bell Labs. In November 2006, (prior to the companies' merger) Alcatel filed a lawsuit against Microsoft (see Alcatel-Lucent v. Microsoft), alleging infringement of seven of its patents. On February 23, 2007 a San Diego court upheld the suit, and awarded Alcatel-Lucent a record-breaking US$1.52 billion in damages.^[25] 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".^[26] 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.^[27]

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

• Partial list of Alcatel-Lucent claimed patents [1]
• Full list of Audio MPEG, Inc claimed patents [2]
• Full list of Thomson/FhG claimed patents [3]

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.

See also

References

External links

• Fraunhofer IIS
• The Story of MP3 — How MP3 was invented, by Fraunhofer IIS

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