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89-605: MP3 (formally MPEG-1 Audio Layer III or MPEG-2 Audio Layer III ) is a coding format for digital audio developed largely by the Fraunhofer Society in Germany under the lead of Karlheinz Brandenburg . It was designed to greatly reduce the amount of data required to represent audio, yet still sound like a faithful reproduction of the original uncompressed audio to most listeners; for example, compared to CD-quality digital audio , MP3 compression can commonly achieve

178-477: A psychoacoustic coding-algorithm exploiting the masking properties of the human ear. Further optimization by Schroeder and Atal with J.L. Hall was later reported in a 1979 paper. That same year, a psychoacoustic masking codec was also proposed by M. A. Krasner, who published and produced hardware for speech (not usable as music bit-compression), but the publication of his results in a relatively obscure Lincoln Laboratory Technical Report did not immediately influence

267-520: A 75–95% reduction in size, depending on the bit rate . In popular usage, MP3 often refers to files of sound or music recordings stored in the MP3 file format (.mp3) on consumer electronic devices. Originally defined in 1991 as the third audio format of the MPEG-1 standard, it was retained and further extended—defining additional bit rates and support for more audio channels —as the third audio format of

356-460: A bit rate, which specifies how many kilobits per second of audio is 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

445-463: A broadcasting system using COFDM modulation was demonstrated on air and in the field 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-chip 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

534-423: 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

623-481: A core part of the MP3 algorithm. Ernst Terhardt and other collaborators constructed an algorithm describing auditory masking with high accuracy in 1982. 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. In 1985, Atal and Schroeder presented code-excited linear prediction (CELP), an LPC-based perceptual speech-coding algorithm with auditory masking that achieved

712-407: A detailed technical specification document known as an audio coding specification . Some such specifications are written and approved by standardization organizations as technical standards , and are thus known as an audio coding standard . The term "standard" is also sometimes used for de facto standards as well as formal standards. Audio content encoded in a particular audio coding format

801-459: 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 postdoctoral researcher at AT&T-Bell Labs with James D. Johnston ("JJ") of AT&T-Bell Labs — with

890-539: 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, the 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 clock rates transitioned from MHz to GHz. Encoder/decoder overall delay

979-488: A home computer as full recordings (as opposed to MIDI notation, or tracker files which combined notation with short recordings of instruments playing single notes). A hacker named SoloH discovered the source code of the "dist10" MPEG reference implementation shortly after the release on the servers of the University of Erlangen . He developed a higher-quality version and spread it on the internet. This code started

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1068-483: A large reduction in file sizes when compared to uncompressed audio. The combination of small size and acceptable fidelity led to a boom in the distribution of music over the Internet in the late 1990s, with MP3 serving as an enabling technology at a time when bandwidth and storage were still at a premium. The MP3 format soon became associated with controversies surrounding copyright infringement , music piracy , and

1157-462: 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

1246-490: 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. Between 1967 and 1974, Eberhard Zwicker did work in the areas of tuning and masking of critical frequency-bands, which in turn built on

1335-401: 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 bit rate was the prime and only consideration. At the time MP3 files were of the very simplest type: they used

1424-439: A quarter of MPEG-1 sample rates. For the general field of human speech reproduction, a bandwidth of 5,512 Hz is sufficient to produce excellent results (for voice) using the sampling rate of 11,025 and VBR encoding from 44,100 (standard) WAV file. English speakers average 41–42 kbit/s with -V 9.6 setting but this may vary with the amount of silence recorded or the rate of delivery (wpm). Resampling to 12,000 (6K bandwidth)

1513-506: 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 rate , a 20 bits/sample input format (the highest available sampling standard in 1991, compatible with the AES/EBU professional digital input studio standard) were

1602-628: A significant data compression ratio for its time. IEEE 's refereed Journal on Selected Areas in Communications reported on a wide variety of (mostly perceptual) audio compression algorithms in 1988. The "Voice Coding for Communications" edition published in February 1988 reported 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 genesis of

1691-446: Is 1152 samples, divided into two granules of 576 samples. These samples, initially in the time domain, are transformed in one block to 576 frequency-domain samples by MDCT. MP3 also allows the use of shorter blocks in a granule, down to a size of 192 samples; this feature is used when a transient is detected. Doing so limits the temporal spread of quantization noise accompanying the transient (see psychoacoustics ). Frequency resolution

1780-476: Is an Indian physicist and engineer. He is a noted researcher in acoustics , and is best known for developments in speech coding . He advanced linear predictive coding (LPC) during the late 1960s to 1970s, and developed code-excited linear prediction (CELP) with Manfred R. Schroeder in 1985. In 1987, Atal was elected as a member into the National Academy of Engineering for innovative research in

1869-402: Is by removing data in ways humans can't hear, according to a psychoacoustic model ; the implementer of an encoder has some freedom of choice in which data to remove (according to their psychoacoustic model). A lossless audio coding format reduces the total data needed to represent a sound but can be de-coded to its original, uncompressed form. A lossy audio coding format additionally reduces

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1958-424: Is compressed, artifacts such as ringing or pre-echo are usually heard. A sample of applause or a triangle instrument with a relatively low bit rate provides good examples of compression artifacts. Most subjective testings of perceptual codecs tend to avoid using these types of sound materials, however, the artifacts generated by percussive sounds are barely perceptible due to the specific temporal masking feature of

2047-420: 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,

2136-545: Is normally encapsulated within a container format . As such, the user normally doesn't have a raw AAC file, but instead has a .m4a audio file , which is a MPEG-4 Part 14 container containing AAC-encoded audio. The container also contains metadata such as title and other tags, and perhaps an index for fast seeking. A notable exception is MP3 files, which are raw audio coding without a container format. De facto standards for adding metadata tags such as title and artist to MP3s, such as ID3 , are hacks which work by appending

2225-432: 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

2314-558: Is selected by the LAME parameter -V 9.4. Likewise -V 9.2 selects a 16,000 sample rate and a resultant 8K lowpass filtering. Older versions of LAME and FFmpeg only support integer arguments for the 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). Audio coding format Some audio coding formats are documented by

2403-406: 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

2492-418: Is the most advanced MP3 encoder. LAME includes a variable bit rate (VBR) encoding which uses a quality parameter rather than a bit rate goal. Later versions (2008+) support an n.nnn quality goal which automatically selects MPEG-2 or MPEG-2.5 sampling rates as appropriate for human speech recordings that need only 5512 Hz bandwidth resolution. In the second half of the 1990s, MP3 files began to spread on

2581-562: 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 the 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

2670-593: 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. A reference simulation software implementation, written in the C language and later known as ISO 11172-5 , was developed (in 1991–1996) by the members of the ISO MPEG Audio committee to produce bit-compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). It

2759-581: 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 's Heinrich Herz Institute . In 1993, he joined

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2848-500: The Institute for Broadcast Technology (Germany), and Matsushita (Japan), 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

2937-598: The Internet , often via underground pirated song networks. The first known experiment in Internet distribution was organized in the early 1990s by the Internet Underground Music Archive , better known by the acronym IUMA. After some experiments using uncompressed audio files, this archive started to deliver on the native worldwide low-speed Internet some compressed MPEG Audio files using the MP2 (Layer II) format and later on used MP3 files when

3026-467: The MPEG-2 ideas and implementation but was named MPEG-2.5 audio since MPEG-3 already had a different meaning. This extension was developed at Fraunhofer IIS, the registered patent holder of MP3, by reducing the frame sync field in the MP3 header from 12 to 11 bits. As in the transition from MPEG-1 to MPEG-2, MPEG-2.5 adds additional sampling rates exactly half of those available using MPEG-2. It thus widens

3115-476: The Nyquist–Shannon sampling theorem . Frequency reproduction is always strictly less than half of the sampling rate, 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

3204-496: The University of Washington . Atal holds more than 16 U.S. patents, and is a member of the National Academy of Engineering and National Academy of Sciences , and a fellow of the Acoustical Society of America and of the Institute of Electrical and Electronics Engineers . He received the 1986 IEEE Morris N. Liebmann Memorial Award "for pioneering contributions to linear predictive coding for speech processing", and

3293-587: The bit resolution of the sound on top of compression, which results in far less data at the cost of irretrievably lost information. Transmitted (streamed) audio is most often compressed using lossy audio codecs as the smaller size is far more convenient for distribution. The most widely used audio coding formats are MP3 and Advanced Audio Coding (AAC), both of which are lossy formats based on modified discrete cosine transform (MDCT) and perceptual coding algorithms. Lossless audio coding formats such as FLAC and Apple Lossless are sometimes available, though at

3382-445: The bitstream , called an audio frame, which is made up of 4 parts, the header , error check , audio data , and ancillary data . The MPEG-1 standard does not include a precise specification for an MP3 encoder but does provide examples of psychoacoustic models, rate loops, and the like in the non-normative part of the original standard. MPEG-2 doubles the number of sampling rates that are supported and MPEG-2.5 adds 3 more. When this

3471-426: The modified discrete cosine transform (MDCT) used by modern audio compression formats such as MP3 and AAC. MDCT was proposed by J. P. Princen, A. W. Johnson and A. B. Bradley in 1987, following earlier work by Princen and Bradley in 1986. The MDCT is used by modern audio compression formats such as Dolby Digital , MP3 , and Advanced Audio Coding (AAC). Bishnu S. Atal Bishnu S. Atal (born 1933)

3560-509: 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

3649-442: The 32 sub-band filterbank of Layer II on which the 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

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3738-643: The Department of Electrical Communication Engineering, Indian Institute of Science, Bangalore. In 1961 Atal joined Bell Laboratories , where his subsequent research focused on acoustics and speech , making major contributions in the field of speech analysis, synthesis, and coding, including low bit-rate speech coding and automatic speech recognition . He advanced and promoted linear predictive coding (1967), and developed code-excited linear prediction (1985) with Manfred R. Schroeder . He retired in 2002 to become affiliate professor of Electrical Engineering at

3827-451: 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. In short, MP3 compression works by reducing

3916-573: The MP3 format and technology is to be found in the perceptual codec MUSICAM based on an integer arithmetics 32 sub-bands filter bank, driven by a psychoacoustic model. It was primarily designed for Digital Audio Broadcasting (digital radio) and digital TV, and its basic principles were 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

4005-580: The MP3 format. It is nonetheless ubiquitous and especially advantageous for low-bit-rate human speech applications. The ISO standard ISO/IEC 11172-3 (a.k.a. MPEG-1 Audio) defined three formats: the MPEG-1 Audio Layer I, Layer II and Layer III. The ISO standard ISO/IEC 13818-3 (a.k.a. MPEG-2 Audio) defined an extended version of MPEG-1 Audio: MPEG-2 Audio Layer I, Layer II, and Layer III. MPEG-2 Audio (MPEG-2 Part 3) should not be confused with MPEG-2 AAC (MPEG-2 Part 7 – ISO/IEC 13818-7). LAME

4094-499: 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 ASPEC, by Fraunhofer Gesellschaft , AT&T , France Telecom , Deutsche and Thomson-Brandt . The second group

4183-399: 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

4272-622: The MPEG-1 Audio Layer III standard, MP3 files with a bit rate below 32 kbit/s might be played back sped-up and pitched-up. Earlier systems also lack fast forwarding and rewinding playback controls on MP3. MPEG-1 frames contain the most detail in 320 kbit/s mode, the highest allowable bit rate setting, with silence and simple tones still requiring 32 kbit/s. MPEG-2 frames can capture up to 12 kHz sound reproductions needed up to 160 kbit/s. MP3 files made with MPEG-2 do not have 20 kHz bandwidth because of

4361-512: 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 41 kbit/s using the MPEG-2.5 extensions. MP3 uses an overlapping MDCT structure. Each MPEG-1 MP3 frame

4450-427: The accuracy of certain components of sound that are considered (by psychoacoustic analysis) to be beyond the hearing capabilities of most humans. This method is commonly referred to as perceptual coding or psychoacoustic modeling. The remaining audio information is then recorded in a space-efficient manner using MDCT and FFT algorithms. The MP3 encoding algorithm is generally split into four parts. Part 1 divides

4539-522: The area of linear predictive coding of speech. Atal was born in India , and received his BS degree in physics (1952) from the University of Lucknow , a diploma in electrical communication engineering (1955) from the Indian Institute of Science , Bangalore and a PhD in electrical engineering (1968) from Brooklyn Polytechnic Institute . From 1957 to 1960, he was a lecturer in acoustics at

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4628-481: The assessment of music compression codecs. The subband coding technique was found to be efficient, not only for the perceptual coding of 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

4717-435: The audio signal into smaller pieces, called frames, and an MDCT filter is then performed on the output. Part 2 passes the sample into a 1024-point fast Fourier transform (FFT), then the psychoacoustic model is applied and another MDCT filter is performed on the output. Part 3 quantifies and encodes each sample, known as noise allocation, which adjusts itself to meet the bit rate and sound masking requirements. Part 4 formats

4806-403: The available frequency fidelity in half while likewise cutting the bit rate by 50%. MPEG-2 Part 3 also enhanced MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel. An MP3 coded with MPEG-2 results in half of the bandwidth reproduction of MPEG-1 appropriate for piano and singing. A third generation of "MP3" style data streams (files) extended

4895-430: 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

4984-557: The compression format during playbacks. This particular track has an interesting property in that the two channels are almost, but not completely, the same, leading to a case where Binaural Masking Level Depression causes spatial unmasking of noise artifacts unless the encoder properly recognizes the situation and applies corrections similar to those detailed in the MPEG-2 AAC psychoacoustic model. Some more critical audio excerpts ( glockenspiel , triangle, accordion , etc.) were taken from

5073-575: The conceptual motivation for the project, was published in the 2014 Proceedings of the International Computer Music Conference. Bit rate 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. The CD is stereo and 16 bits per channel. So, multiplying 44100 by 32 gives 1411200—the bit rate of uncompressed CD digital audio. MP3

5162-433: The cost of larger files. Uncompressed audio formats, such as pulse-code modulation (PCM, or .wav), are also sometimes used. PCM was the standard format for Compact Disc Digital Audio (CDDA). In 1950, Bell Labs filed the patent on differential pulse-code modulation (DPCM). Adaptive DPCM (ADPCM) was introduced by P. Cummiskey, Nikil S. Jayant and James L. Flanagan at Bell Labs in 1973. Perceptual coding

5251-681: 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 Leibniz University Hannover , the editing of the standard was delegated to Leon van de Kerkhof (Netherlands), Gerhard Stoll (Germany), and Yves-François Dehery (France), who worked on Layer I and Layer II. ASPEC

5340-597: 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 the listening environment (ambient noise), listener attention, listener training, and in most cases by listener audio equipment (such as sound cards, speakers, and headphones). Furthermore, sufficient quality may be achieved by

5429-475: The file- ripping and sharing services MP3.com and Napster , among others. With the advent of portable media players (including "MP3 players"), a product category also including smartphones , MP3 support remains near-universal and a de facto standard for digital audio. The Moving Picture Experts Group (MPEG) designed MP3 as part of its MPEG-1 , and later MPEG-2 , standards. MPEG-1 Audio (MPEG-1 Part 3), which included MPEG-1 Audio Layer I, II, and III,

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5518-478: The first generation of MP3 defined 14 × 3 = 42 interpretations of MP3 frame data structures and size layouts. The compression efficiency of encoders is typically defined by the bit rate because the compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. They may use the compact disc (CD) parameters as references (44.1 kHz , 2 channels at 16 bits per channel or 2×16 bit), or sometimes

5607-482: The fundamental research in the area from Harvey Fletcher and his collaborators at Bell Labs . Perceptual coding was first used for speech coding compression with linear predictive coding (LPC), which has origins in the work of Fumitada Itakura ( Nagoya University ) and Shuzo Saito ( Nippon Telegraph and Telephone ) in 1966. In 1978, Bishnu S. Atal and Manfred R. Schroeder at Bell Labs proposed an LPC speech codec , called adaptive predictive coding , that used

5696-611: The joint stereo coding of MUSICAM and created the MP3 format, which was designed to achieve the same quality at 128  kbit/s as MP2 at 192 kbit/s. The algorithms for MPEG-1 Audio Layer I, II and III 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 (a.k.a. MPEG-1 Audio or MPEG-1 Part 3 ), published in 1993. Files or data streams conforming to this standard must handle sample rates of 48k, 44100, and 32k and continue to be supported by current MP3 players and decoders. Thus

5785-481: 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 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

5874-508: The mainstream of psychoacoustic codec-development. The discrete cosine transform (DCT), a type of transform coding for lossy compression, proposed by Nasir Ahmed in 1972, was developed by Ahmed with T. Natarajan and K. R. Rao in 1973; they published their results in 1974. This led to the development of the modified discrete cosine transform (MDCT), proposed by J. P. Princen, A. W. Johnson and A. B. Bradley in 1987, following earlier work by Princen and Bradley in 1986. The MDCT later became

5963-404: The masking properties of the human ear, followed in the early 1980s with the code-excited linear prediction (CELP) algorithm which achieved a significant compression ratio for its time. Perceptual coding is used by modern audio compression formats such as MP3 and AAC . Discrete cosine transform (DCT), developed by Nasir Ahmed , T. Natarajan and K. R. Rao in 1974, provided the basis for

6052-415: The previous generation for a total of 9 varieties of MP3 format files. The sample rate comparison table between MPEG-1, 2, and 2.5 is given later in the article. MPEG-2.5 is supported by LAME (since 2000), Media Player Classic (MPC), iTunes, and FFmpeg. MPEG-2.5 was not developed by MPEG (see above) and was never approved as an international standard. MPEG-2.5 is thus an unofficial or proprietary extension to

6141-414: The recording industry approved re-incarnation of Napster , and Amazon.com sell unrestricted music in the MP3 format. 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 "bit reservoir", frames are not independent items and cannot usually be extracted on arbitrary frame boundaries. The MP3 Data blocks contain

6230-594: The reproduction of Vega's voice. Accordingly, he dubbed Vega the "Mother of MP3". Instrumental music had been easier to compress, but Vega's voice sounded unnatural in early versions of the format. Brandenburg eventually met Vega and heard Tom's Diner performed live. In 1991, two available proposals were assessed for an MPEG audio standard: MUSICAM ( M asking pattern adapted U niversal S ubband I ntegrated C oding A nd M ultiplexing) and ASPEC ( A daptive S pectral P erceptual E ntropy C oding). The MUSICAM technique, proposed by Philips (Netherlands), CCETT (France),

6319-440: 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 optimize the size of the file by creating files where the bit rate changes throughout the file. These are known as variable bit rate. The bit reservoir and VBR encoding were part of the original MPEG-1 standard. The concept behind them

6408-863: 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 rates 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 rates 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 rates of 8, 11.025, and 12 kHz. On earlier systems that only support

6497-514: The scope of MP3 to include human speech and other applications yet requires only 25% of the bandwidth (frequency reproduction) possible using MPEG-1 sampling rates. While not an ISO-recognized standard, MPEG-2.5 is widely supported by both inexpensive Chinese and brand-name digital audio players as well as computer software-based MP3 encoders ( LAME ), decoders (FFmpeg) and players (MPC) adding 3 × 8 = 24 additional MP3 frame types. Each generation of MP3 thus supports 3 sampling rates exactly half that of

6586-427: The staff of Fraunhofer HHI. An acapella version of the song " Tom's Diner " by Suzanne Vega was the first song used by Brandenburg to develop the MP3 format. It was used as a benchmark to see how well MP3's compression algorithm handled the human voice. Brandenburg adopted the song for testing purposes, listening to it again and again each time he refined the compression algorithm, making sure it did not adversely affect

6675-518: The standard was fully completed. The popularity of MP3s began to rise rapidly with the advent of Nullsoft 's audio player Winamp , released in 1997, which still had in 2023 a community of 80 million active users. In 1998, the first portable solid-state digital audio player MPMan , developed by SaeHan Information Systems, which is headquartered in Seoul , South Korea , was released and the Rio PMP300

6764-413: The subsequent MPEG-2 standard. MP3 as a file format commonly designates files containing an elementary stream of MPEG-1 Audio or MPEG-2 Audio encoded data, without other complexities of the MP3 standard. Concerning audio compression , which is its most apparent element to end-users, MP3 uses lossy compression to encode data using inexact approximations and the partial discarding of data, allowing for

6853-428: The tags to the MP3, and then relying on the MP3 player to recognize the chunk as malformed audio coding and therefore skip it. In video files with audio, the encoded audio content is bundled with video (in a video coding format ) inside a multimedia container format . An audio coding format does not dictate all algorithms used by a codec implementing the format. An important part of how lossy audio compression works

6942-688: The widespread CD ripping and digital music distribution as MP3 over the internet. 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 (a.k.a. MPEG-2 Part 3 or backward compatible MPEG-2 Audio or MPEG-2 Audio BC ), originally published in 1995. MPEG-2 Part 3 (ISO/IEC 13818-3) defined 42 additional bit rates and sample rates for MPEG-1 Audio Layer I, II and III. The new sampling rates are exactly half that of those originally defined in MPEG-1 Audio. This reduction in sampling rates serves to cut

7031-488: Was MUSICAM , by Matsushita , CCETT , ITT and Philips . The third group was ATAC (ATRAC Coding), 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

7120-483: Was able to demonstrate the first real-time hardware decoding (DSP based) of compressed audio. Some other real-time implementations of MPEG Audio encoders and decoders were available for digital broadcasting (radio DAB , television DVB ) towards consumer receivers and set-top boxes. On 7 July 1994, the Fraunhofer Society released the first software MP3 encoder, called l3enc . The filename extension .mp3

7209-481: Was approved as a committee draft for an ISO / IEC standard in 1991, finalized in 1992, and published in 1993 as ISO/IEC 11172-3:1993. An MPEG-2 Audio (MPEG-2 Part 3) extension with lower sample and bit rates was published in 1995 as ISO/IEC 13818-3:1995. It requires only minimal modifications to existing MPEG-1 decoders (recognition of the MPEG-2 bit in the header and addition of the new lower sample and bit rates). The MP3 lossy compression algorithm takes advantage of

7298-605: Was approved as a committee draft of the ISO/IEC technical report in March 1994 and printed as document CD 11172-5 in April 1994. It was approved as a draft technical report (DTR/DIS) in November 1994, finalized in 1996 and published as international standard ISO/IEC TR 11172-5:1998 in 1998. The reference software in C language was later published as a freely available ISO standard. Working in non-real time on several operating systems, it

7387-459: Was chosen by the Fraunhofer team on 14 July 1995 (previously, the files had been named .bit ). With the first real-time software MP3 player WinPlay3 (released 9 September 1995) many people were able to encode and play back MP3 files on their PCs. Because of the relatively small hard drives of the era (≈500–1000 MB ) lossy compression was essential to store multiple albums' worth of music on

7476-401: Was designed to encode this 1411 kbit/s data at 320 kbit/s or less. If less complex passages are detected by the MP3 algorithms then lower bit rates may be employed. When using MPEG-2 instead of MPEG-1, MP3 supports only lower sampling rates (16,000, 22,050, or 24,000 samples per second) and offers choices of bit rate as low as 8 kbit/s but no higher than 160 kbit/s. By lowering

7565-428: Was first used for speech coding compression, with linear predictive coding (LPC). Initial concepts for LPC date back to the work of Fumitada Itakura ( Nagoya University ) and Shuzo Saito ( Nippon Telegraph and Telephone ) in 1966. During the 1970s, Bishnu S. Atal and Manfred R. Schroeder at Bell Labs developed a form of LPC called adaptive predictive coding (APC), a perceptual coding algorithm that exploited

7654-589: Was launched in 1999. The ease of creating and sharing MP3s resulted in widespread copyright infringement . Major record companies argued that this free sharing of music reduced sales, and called it " music piracy ". They reacted by pursuing lawsuits against Napster , which was eventually shut down and later sold, and against individual users who engaged in file sharing. Unauthorized MP3 file sharing continues on next-generation peer-to-peer networks . Some authorized services, such as Beatport , Bleep , Juno Records , eMusic , Zune Marketplace , Walmart.com , Rhapsody ,

7743-458: Was sold afterward in 1998, despite legal suppression efforts by the RIAA . In November 1997, the website mp3.com was offering thousands of MP3s created by independent artists for free. The small size of MP3 files enabled widespread peer-to-peer file sharing of music ripped from CDs, which would have previously been nearly impossible. The first large peer-to-peer filesharing network, Napster ,

7832-454: Was the joint proposal of AT&T Bell Laboratories, Thomson Consumer Electronics, Fraunhofer Society, and CNET . It provided the highest coding efficiency. A working group consisting of van de Kerkhof, Stoll, Leonardo Chiariglione ( CSELT VP for Media), Yves-François Dehery, Karlheinz Brandenburg (Germany) and James D. Johnston (United States) took ideas from ASPEC, integrated the filter bank from Layer II, added some of their ideas such as

7921-571: Was written, the suggested implementations were quite dated. Implementers of the standard were supposed to devise 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

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