The Info List - Mp3

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AUDIO LAYER III, more commonly referred to as MP3 (or MP3), is an audio coding format for digital audio . It uses a form of lossy data compression to encode data using inexact approximations and partial data discarding to reduce file sizes, typically by a factor of 10, in comparison with a CD while retaining a sound quality comparable to uncompressed audio. Compared to CD quality digital audio, MP3 compression commonly achieves 75 to 95% reduction in size. MP3 files are thus 1/4 to 1/20 the size of the original digital audio stream. This is important for both transmission and storage concerns. The basis for such comparison is the CD digital audio format which requires 1411200 bit/s. A commonly used MP3 encoding setting is CBR 128 kbit/s resulting in file of 1/11 (=9%) of the size of the original CD -quality file, that is with 91% compression.

The MP3 lossy compression works by reducing (or approximating) the accuracy of certain parts of a continuous sound that are considered to be beyond the auditory resolution ability of most people. This method is commonly referred to as perceptual coding or "psychoacoustics" . It uses psychoacoustic models to discard or reduce the precision of components less audible to human hearing, and then records the remaining information in an efficient manner.

MP3 was designed by the Moving Picture Experts Group
Moving Picture Experts Group
(MPEG) as part of its MPEG-1
standard and later extended in the MPEG-2
standard. The first subgroup for audio was formed by several teams of engineers at CCETT , Matsushita , Philips
, Sony
, AT&T- Bell Labs
Bell Labs
, Thomson-Brandt , and others. MPEG-1
Audio ( MPEG-1
Part 3), which included MPEG-1 Audio Layer I, II and III was approved as a committee draft of ISO /IEC standard in 1991, finalised in 1992 and published in 1993 (ISO/IEC 11172-3:1993 ). A backwards compatible MPEG-2
Audio (MPEG-2 Part 3) extension with lower sample and bit rates was published in 1995 (ISO/IEC 13818-3:1995). MP3 is a streaming or broadcast format (as opposed to a file format) meaning that individual frames can be lost without affecting the ability to decode successfully delivered frames. Storing an MP3 stream in a file enables time-shifted playback.


* 1 History

* 1.1 Development * 1.2 Standardization * 1.3 Going public * 1.4 Internet distribution

* 2 Design

* 2.1 File structure * 2.2 Encoding and decoding * 2.3 Quality * 2.4 Bit rate * 2.5 Ancillary data * 2.6 Metadata

* 3 Licensing, ownership and legislation * 4 Alternative technologies * 5 See also * 6 References * 7 Further reading * 8 External links



The MP3 lossy audio data compression algorithm takes advantage of a perceptual limitation of human hearing called auditory masking . In 1894, the American physicist Alfred M. Mayer reported that a tone could be rendered inaudible by another tone of lower frequency. In 1959, Richard Ehmer described a complete set of auditory curves regarding this phenomenon. Ernst Terhardt _et al._ created an algorithm describing auditory masking with high accuracy. This work added to a variety of reports from authors dating back to Fletcher, and to the work that initially determined critical ratios and critical bandwidths.

The psychoacoustic masking codec was first proposed in 1979, apparently independently, by Manfred R. Schroeder
Manfred R. Schroeder
, et al. from Bell Telephone Laboratories, Inc. in Murray Hill, New Jersey
Murray Hill, New Jersey
, and M. A. Krasner both in the United States. Krasner was the first to publish and to produce hardware for speech (not usable as music bit compression), but the publication of his results as a relatively obscure Lincoln Laboratory
Lincoln Laboratory
Technical Report did not immediately influence the mainstream of psychoacoustic codec development. Manfred Schroeder was already a well-known and revered figure in the worldwide community of acoustical and electrical engineers, but his paper was not much noticed, since it described negative results due to the particular nature of speech and the linear predictive coding (LPC) gain present in speech.

Both Krasner and Schroeder built upon the work performed by Eberhard F. Zwicker in the areas of tuning and masking of critical frequency bands, that in turn built on the fundamental research in the area from Bell Labs
Bell Labs
of Harvey Fletcher and his collaborators. A wide variety of (mostly perceptual) audio compression algorithms were reported in IEEE
's refereed Journal on Selected Areas in Communications. That journal reported in February 1988 on a wide range of established, working audio bit compression technologies, some of them using auditory masking as part of their fundamental design, and several showing real-time hardware implementations.

The Moving Picture Experts Group
Moving Picture Experts Group
(MPEG) was established in 1988 by the initiative of Hiroshi Yasuda ( Nippon Telegraph and Telephone
Nippon Telegraph and Telephone
) and Leonardo Chiariglione . Yasuda was leading an initiative in Japan, called the Digital Audio and Picture Architecture (DAPA), while Chiariglione was leading an initiative in Europe, called the Coding of Moving Images for Storage (COMIS). Both eventually met in May 1988 to work on a global standard.

The genesis of the MP3 technology is fully described in a paper from Professor Hans Musmann, who chaired the ISO MPEG Audio group for several years. In December 1988, MPEG called for an audio coding standard. In June 1989, 14 audio coding algorithms were submitted. Because of certain similarities between these coding proposals, they were clustered into four development groups. The first group was MUSICAM , by Matsushita , CCETT , ITT and Philips
. The second group was ASPEC, by AT&T , France Telecom , Fraunhofer Gesellschaft , Deutsche and Thomson-Brandt . The third group was ATAC, by Fujitsu , JVC , NEC and Sony
. And the fourth group was SB-ADPCM , by NTT and BTRL.

The immediate predecessors of MP3 were "Optimum Coding in the Frequency Domain" (OCF), and Perceptual Transform Coding (PXFM). These two codecs, along with block-switching contributions from Thomson-Brandt, were merged into a codec called ASPEC, which was submitted to MPEG, and which won the quality competition, but that was mistakenly rejected as too complex to implement. The first practical implementation of an audio perceptual coder (OCF) in hardware (Krasner's hardware was too cumbersome and slow for practical use), was an implementation of a psychoacoustic transform coder based on Motorola 56000 DSP chips.

Another predecessor of the MP3 format and technology is to be found in the perceptual codec MUSICAM based on an integer arithmetics 32 sub-bands filterbank, driven by a psychoacoustic model. It was primarily designed for Digital Audio Broadcasting (digital radio) and digital TV, and its basic principles disclosed to the scientific community by CCETT (France) and IRT (Germany) in Atlanta during an IEEE-ICASSP conference in 1991, after having worked on MUSICAM with Matsushita and Philips
since 1989.

This codec incorporated into a broadcasting system using COFDM modulation was demonstrated on air and on the field together with Radio Canada and CRC Canada during the NAB show (Las Vegas) in 1991. The implementation of the audio part of this broadcasting system was based on a two chips encoder (one for the subband transform, one for the psychoacoustic model designed by the team of G. Stoll (IRT Germany), later known as psychoacoustic model I) and a real time decoder using one Motorola 56001 DSP chip running an integer arithmetics software designed by Y.F. Dehery's team (CCETT , France). The simplicity of the corresponding decoder together with the high audio quality of this codec using for the first time a 48 kHz sampling frequency, a 20 bits/sample input format (the highest available sampling standard in 1991, compatible with the AES/EBU professional digital input studio standard) were the main reasons to later adopt the characteristics of MUSICAM as the basic features for an advanced digital music compression codec.

During the development of the MUSICAM encoding software, Stoll and Dehery's team made a thorough use of a set of high quality audio assessment material selected by a group of audio professionals from the European Broadcasting Union and later used as a reference for the assessment of music compression codecs . The subband coding technique was found to be efficient, not only for the perceptual coding of the high quality sound materials but especially for the encoding of critical percussive sound materials (drums, triangle, ..) due to the specific temporal masking effect of the MUSICAM sub-band filterbank (this advantage being a specific feature of short transform coding techniques).

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. MP3 is directly descended from OCF and PXFM, representing the outcome of the collaboration of Brandenburg—working as a postdoc at AT&T- Bell Labs
Bell Labs
with James D. Johnston ("JJ") of AT&T-Bell Labs—with the Fraunhofer Institute for Integrated Circuits, Erlangen (where he worked with Bernhard Grill and four other researchers – "The Original Six" ), with relatively minor contributions from the MP2 branch of psychoacoustic sub-band coders. In 1990, Brandenburg 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). The song "Tom\'s Diner " by Suzanne Vega was the first song used by Karlheinz Brandenburg to develop the MP3. Brandenburg adopted the song for testing purposes, listening to it again and again each time refining the scheme, making sure it did not adversely affect the subtlety of Vega's voice.


In 1991, there were two available proposals that were assessed for an MPEG audio standard: MUSICAM (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). As proposed by the Dutch corporation Philips
, the French research institute CCETT, and the German standards organization Institute for Broadcast Technology , the MUSICAM technique was chosen due to its simplicity and error robustness, as well as for its high level of computational efficiency. The MUSICAM format, based on sub-band coding , became the basis for the MPEG Audio compression format, incorporating, for example, its frame structure, header format, sample rates, etc.

While much of MUSICAM technology and ideas were incorporated into the definition of MPEG Audio Layer I and Layer II, the filter bank alone and the data structure based on 1152 samples framing (file format and byte oriented stream) of MUSICAM remained in the Layer III (MP3) format, as part of the computationally inefficient hybrid filter bank. Under the chairmanship of Professor Musmann of the University of Hanover , the editing of the standard was delegated to Dutchman Leon van de Kerkhof , to German Gerhard Stoll , to Frenchman Yves-François Dehery , who worked on Layer I and Layer II. ASPEC was the joint proposal of AT"> Diagram of the structure of an MP3 file (MPEG version 2.5 not supported, hence 12 instead of 11 bits for MP3 Sync Word).

An MP3 file is made up of MP3 frames, which consist of a header and a data block. This sequence of frames is called an elementary stream . Due to the "byte reservoir", frames are not independent items and cannot usually be extracted on arbitrary frame boundaries. The MP3 Data blocks contain the (compressed) audio information in terms of frequencies and amplitudes. The diagram shows that the MP3 Header consists of a sync word , which is used to identify the beginning of a valid frame. This is followed by a bit indicating that this is the MPEG standard and two bits that indicate that layer 3 is used; hence MPEG-1
Audio Layer 3 or MP3. After this, the values will differ, depending on the MP3 file. _ISO /IEC 11172-3_ defines the range of values for each section of the header along with the specification of the header. Most MP3 files today contain ID3 metadata , which precedes or follows the MP3 frames, as noted in the diagram. The data stream can contain an optional checksum.

Joint stereo is done only on a frame-to-frame basis.


The MPEG-1
standard does not include a precise specification for an MP3 encoder, but does provide example psychoacoustic models, rate loop, and the like in the non-normative part of the original standard. MPEG-2
doubles the number of sampling rates which are supported and MPEG-2.5 adds 3 more. When this was written, the suggested implementations were quite dated. Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information from the audio input. As a result, many different MP3 encoders became available, each producing files of differing quality. Comparisons were widely available, so it was easy for a prospective user of an encoder to research the best choice. Some encoders that were proficient at encoding at higher bit rates (such as LAME ) were not necessarily as good at lower bit rates. Over time, LAME evolved on the SourceForge website until it became the de facto CBR MP3 encoder. Later an ABR mode was added. Work progressed on true variable bit rate using a quality goal between 0 and 10. Eventually numbers (such as -V 9.600) could generate excellent quality low bit rate voice encoding at only 41kbit/sec using the mpeg-2.5 extensions.

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 .) Frequency resolution is limited by the small long block window size, which decreases coding efficiency. Time resolution can be too low for highly transient signals and may cause smearing of percussive sounds.

Due to the tree structure of the filter bank, pre-echo problems are made worse, as the combined impulse response of the two filter banks does not, and cannot, provide an optimum solution in time/frequency resolution. Additionally, the combining of the two filter banks' outputs creates aliasing problems that must be handled partially by the "aliasing compensation" stage; however, that creates excess energy to be coded in the frequency domain, thereby decreasing coding efficiency.

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 high standard document (ISO/IEC 11172-3). 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). Over time this concern has become less of an issue as CPU speeds transitioned from MHz to GHz. Encoder/decoder overall delay is not defined, which means there is no official provision for gapless playback . However, some encoders such as LAME can attach additional metadata that will allow players that can handle it to deliver seamless playback.


When performing lossy audio encoding, such as creating an MP3 data stream, there is a trade-off between the amount of data generated and the sound quality of the results. The person generating an MP3 selects a bit rate , which specifies how many kilobits per second of audio are desired. The higher the bit rate, the larger the MP3 data stream will be, and, generally, the closer it will sound to the original recording. 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 or an excerpt of triangle instrument with a relatively low bit rate provides a good example of compression artifacts. Most subjective testings on perceptual codecs tend to avoid using this type of extremely critical sound materials however this type of artifacts generated by percussive sounds is barely perceptible on the MP3 format due to the specific temporal masking feature of the 32 sub-band filterbank of Layer II on which the MP3 format is based.

Besides the bit rate of an encoded piece of audio, the quality of MP3 encoded sound also depends on the quality of the encoder algorithm as well as the complexity of the signal being encoded. As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders do feature quite different quality, even with identical bit rates. As an example, in a public listening test featuring two early MP3 encoders set at about 128 kbit/s, one scored 3.66 on a 1–5 scale, while the other scored only 2.22. Quality is dependent on the choice of encoder and encoding parameters.

This observation caused a revolution in audio encoding. Early on bitrate was the prime and only consideration. At the time MP3 files were of the very simplest type: they used the same bit rate for the entire file: this process is known as Constant Bit Rate (CBR) encoding. Using a constant bit rate makes encoding simpler and less CPU intensive. However, it is also possible to create files where the bit rate changes throughout the file. These are known as Variable Bit Rate The bit reservoir and VBR encoding were actually part of the original MPEG-1
standard. The concept behind them is that, in any piece of audio, some sections are easier to compress, such as silence or music containing only a few tones, 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 advanced MP3 encoders, it is possible to specify a given quality, and the encoder will adjust the bit rate accordingly. Users that desire a particular "quality setting" that is transparent to their ears can use this value when encoding all of their music, and generally speaking not need to worry about performing personal listening tests on each piece of music to determine the correct bit rate.

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). Furthermore, sufficient quality may be achieved by a lesser quality setting for lectures and human speech applications and reduces encoding time and complexity. A test given to new students by Stanford University Music Professor Jonathan Berger showed that student preference for MP3-quality music has risen each year. Berger said the students seem to prefer the 'sizzle' sounds that MP3s bring to music.

An in-depth study of MP3 audio quality, sound artist and composer Ryan Maguire 's project "The Ghost in the MP3" isolates the sounds lost during MP3 compression. In 2015, he released the track "moDernisT" (an anagram of "Tom's Diner"), composed exclusively from the sounds deleted during MP3 compression of the song "Tom's Diner", the track originally used in the formulation of the MP3 standard. A detailed account of the techniques used to isolate the sounds deleted during MP3 compression, along with the conceptual motivation for the project, was published in the 2014 Proceedings of the International Computer Music Conference.


MPEG Audio Layer III available bit rates (kbit/s) MPEG-1 Audio Layer III MPEG-2 Audio Layer III MPEG-2.5 Audio Layer III

– 8 8

– 16 16

– 24 24

32 32 32

40 40 40

48 48 48

56 56 56

64 64 64

80 80

96 96

112 112

128 128

n/a 144

160 160

192 – –

224 – –

256 – –

320 – –

Supported sampling rates by MPEG Audio Format MPEG-1 Audio Layer III MPEG-2 Audio Layer III MPEG-2.5 Audio Layer III

– – 8000 Hz

– – 11025 Hz

– – 12000 Hz

– 16000 Hz –

– 22050 Hz –

– 24000 Hz –

32000 Hz – –

44100 Hz – –

48000 Hz – –

Bitrate is the product of the sample rate and number of bits per sample used to encode the music. CD audio is 44100 samples per second. The number of bits per sample also depends on the number of audio channels. CD is stereo and 16 bits per channel. So, multiplying 44100 by 32 gives 1411200—the bitrate of uncompressed CD digital audio. MP3 was designed to encode this 1411 kbit/s data at 320 kbit/s or less. As less complex passages are detected by MP3 algorithms then lower bitrates may be employed. When using MPEG-2
instead of MPEG-1, MP3 supports only lower sampling rates (16000, 22050 or 24000 samples per second) and offers choices of bitrate as low as 8 kbit/s but no higher than 160 kbit/s. By lowering the sampling rate, MPEG-2
layer III removes all frequencies above half the new sampling rate that may have been present in the source audio.

As shown in these two tables, 14 selected bit rates are allowed in MPEG-1
Audio Layer III standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, along with the 3 highest available sampling frequencies of 32, 44.1 and 48 kHz . MPEG-2
Audio Layer III also allows 14 somewhat different (and mostly lower) bit rates of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s with sampling frequencies of 16, 22.05 and 24 kHz which are exactly half that of MPEG-1
MPEG-2.5 Audio Layer III frames are limited to only 8 bit rates of 8, 16, 24, 32, 40, 48, 56 and 64 kbit/s with 3 even lower sampling frequencies of 8, 11.025, and 12 kHz.

frames contain the most detail in 320kbit/s mode with silence and simple tones still requiring 32 kbit/s. MPEG-2
frames can capture up to 12 kHz sound reproductions needed up to 160kbit/s. MP3 files made with MPEG-2
don't have 20 kHz bandwidth because of the Nyquist–Shannon sampling theorem . Frequency reproduction is always strictly less than half of the sampling frequency, and imperfect filters require a larger margin for error (noise level versus sharpness of filter), so an 8 kHz sampling rate limits the maximum frequency to 4 kHz, while a 48 kHz sampling rate limits an MP3 to a maximum 24 kHz sound reproduction. MPEG-2
uses half and MPEG-2.5 only a quarter of MPEG-1
sample rates.

For the general field of human speech reproduction, a bandwidth of 5512 Hz is sufficient to produce excellent results (for voice) using the sampling rate of 11025 and VBR encoding from 44100 (standard) wave files.. This is easily accomplished using LAME version 3.99.5 and the command line "lame -V 9.6 lecture.WAV" English speakers average 41-42kbit/s with -V 9.6 setting but this may vary with amount of silence recorded or the rate of delivery (wpm). Resampling to 12000 (6K bandwidth) is selected by the LAME parameter -V 9.4 Likewise -V 9.2 selects 16000 sample rate and a resultant 8K lowpass filtering. For more info see Nyquist – Shannon. Older versions of LAME and FFmpeg only support integer arguments for variable bit rate quality selection parameter. The n.nnn quality parameter (-V) is documented at lame.sourceforge.net but is only supported in LAME with the new style VBR variable bit rate quality selector—not average bit rate (ABR).

A sample rate of 44.1 kHz is commonly used for music reproduction, because this is also used for CD audio , the main source used for creating MP3 files. A great variety of bit rates are used on the Internet. A bit rate of 128 kbit/s is commonly used, at a compression ratio of 11:1, offering adequate audio quality in a relatively small space. As Internet bandwidth availability and hard drive sizes have increased, higher bit rates up to 320 kbit/s are widespread. Uncompressed audio as stored on an audio-CD has a bit rate of 1,411.2 kbit/s, (16 bit/sample × 44100 samples/second × 2 channels / 1000 bits/kilobit), so the bitrates 128, 160 and 192 kbit/s represent compression ratios of approximately 11:1, 9:1 and 7:1 respectively.

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. Early MPEG Layer III encoders used what is now called Constant Bit Rate (CBR). The software was only able to use a uniform bitrate on all frames in an MP3 file. Later more sophisticated MP3 encoders were able to use the bit reservoir to target an average bit rate selecting the encoding rate for each frame based on the complexity of the sound in that portion of the recording.

A more sophisticated MP3 encoder can produce variable bitrate audio. MPEG audio may use bitrate switching on a per-frame basis, but only layer III decoders must support it. VBR is used when the goal is to achieve a fixed level of quality. The final file size of a VBR encoding is less predictable than with constant bitrate . Average bitrate is a type of VBR implemented as a compromise between the two: the bitrate is allowed to vary for more consistent quality, but is controlled to remain near an average value chosen by the user, for predictable file sizes. Although an MP3 decoder must support VBR to be standards compliant, historically some decoders have bugs with VBR decoding, particularly before VBR encoders became widespread. The most evolved LAME MP3 encoder supports the generation of VBR, ABR, and even the ancient CBR MP3 formats.

Layer III audio can also use a "bit reservoir", a partially full frame's ability to hold part of the next frame's audio data, allowing temporary changes in effective bitrate, even in a constant bitrate stream. Internal handling of the bit reservoir increases encoding delay. There is no scale factor band 21 (sfb21) for frequencies above approx 16 kHz , forcing the encoder to choose between less accurate representation in band 21 or less efficient storage in all bands below band 21, the latter resulting in wasted bitrate in VBR encoding.


The ancillary data field can be used to store user defined data. The ancillary data is optional and the number of bits available is not explicitly given. The ancillary data is located after the Huffman code bits and ranges to where the next frame's main_data_begin points to. mp3PRO uses ancillary data to encode their bits to improve audio quality.


Main articles: ID3 and APEv2 tag

A "tag" in an 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. The MP3 standards do not define tag formats for MP3 files, nor is there a standard container format that would support metadata and obviate the need for tags. However, several _de facto_ standards for tag formats exist. As of 2010, the most widespread are ID3v1 and ID3v2 , and the more recently introduced APEv2 . These tags are normally embedded at the beginning or end of MP3 files, separate from the actual MP3 frame data. MP3 decoders either extract information from the tags, or just treat them as ignorable, non- MP3 junk data.

Playing U.S. Patent 5,850,456 , expired February 2017; and U.S. Patent 5,960,037 , expired 9 April 2017. However, the last MP3 patent, U.S. Patent 5,703,999 might not expire before 30 December 2017.

In September 2006, German officials seized MP3 players from SanDisk 's booth at the IFA show in Berlin after an Italian patents firm won an injunction on behalf of Sisvel against SanDisk in a dispute over licensing rights. The injunction was later reversed by a Berlin judge, but that reversal was in turn blocked the same day by another judge from the same court, "bringing the Patent Wild West to Germany" in the words of one commentator. In February 2007, Texas MP3 Technologies sued Apple, Samsung Electronics and Sandisk in eastern Texas federal court , claiming infringement of a portable MP3 player patent that Texas MP3 said it had been assigned. Apple, Samsung, and Sandisk all settled the claims against them in January 2009.

Alcatel-Lucent has asserted several MP3 coding and compression patents, allegedly inherited from AT it was co-owned by AT codecs="vorbis"" data-title=" Ogg Vorbis" data-shorttitle=" Ogg Vorbis" data-transcodekey="ogg" data-width="0" data-height="0" data-bandwidth="109744" /> The first is uncompressed WAV file. The second is a Vorbis file encoded at 48kbit/s, and third is an MP3 encoded at 48kbit/s using LAME . -------------------------

Problems playing this file? See media help ._

Main article: List of codecs

Other lossy formats 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 Society owns many of the basic patents underlying these formats as well, with others held by Alcatel-Lucent, and Thomson Consumer Electronics . There are also open compression formats like Opus and Vorbis that are available free of charge and without any known patent restrictions. Some of the newer audio compression formats, such as AAC, WMA Pro and Vorbis, are free of some limitations inherent to the MP3 format that cannot be overcome by any MP3 encoder.

Besides lossy compression methods, lossless formats are a significant alternative to MP3 because they provide unaltered audio content, though with an increased file size compared to lossy compression. Lossless formats include FLAC (Free Lossless Audio Codec), Apple Lossless and many others.


* Information technology portal

* Comparison of audio coding formats * MP3 blog * MP3 player * MP3 Surround * MP3HD * MPEG-4 Part 14 * Podcast * Portable media player


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* Geert Lovink (2014). "Reflections on the MP3 Format: Interview with Jonathan Sterne". _Computational Culture_ (4). ISSN 2047-2390 .


_ Wikimedia Commons has media related to MP3 _.

* MP3 at DMOZ * MP3-history.com, The Story of MP3: How MP3 was invented,