MPEG-1 Audio Layer 3, more commonly referred to as MP3, is a digital audio encoding format using a form of lossy data compression.
This encoding format is used to create the MP3's small file, as a way to store a single segment of audio, commonly a song, such that the file can be easily organized and transferred between computers or other devices such as MP3 players.
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 some listeners, but is not considered High Fidelity audio by the elite connoisseur. 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.
MP3 is an audio-specific format. It was invented by a team of international engineers at Philips, CCETT (Centre commun d'études de télévision et télécommunications), IRT, AT&T-Bell Labs and Fraunhofer Society, and it became an ISO/IEC standard in 1991. 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 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, say, JPEG, an image compression format.
This encoding format is used to create the MP3's small file, as a way to store a single segment of audio, commonly a song, such that the file can be easily organized and transferred between computers or other devices such as MP3 players.
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 some listeners, but is not considered High Fidelity audio by the elite connoisseur. 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.
MP3 is an audio-specific format. It was invented by a team of international engineers at Philips, CCETT (Centre commun d'études de télévision et télécommunications), IRT, AT&T-Bell Labs and Fraunhofer Society, and it became an ISO/IEC standard in 1991. 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 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, say, JPEG, an image compression format.
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.
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 which 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 Musicam (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 Union as a part of the EUREKA research program where it was commonly known as EU-147, which ran from 1987 to 1994.
As a doctoral student at Germany's University of Erlangen-Nuremberg, Karlheinz Brandenburg began working on digital music compression in the early 1980s, focusing on how people perceive music. He completed his doctoral work in 1989 and became an assistant professor at Erlangen-Nuremberg. While there, he continued to work on music compression with scientists at the Fraunhofer Society (in 1993 he joined the staff of the Fraunhofer Institute).[9]
In 1991, there were two proposals available: Musicam (known as Layer 2), 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. The Musicam format, based on sub-band coding, was a basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame). Its technology and ideas were fully incorporated into the definition of ISO MPEG Audio Layer I and Layer II and further on of the Layer III (MP3) format. Under the chairmanship of Professor 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 rate depends on the bit depth and sampling rate of the input signal. Nevertheless, there are often published compression rates that use the CD parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit). Sometimes the Digital Audio Tape (DAT) SP parameters are used (48 kHz, 2×16 bit). Compression ratios with this reference are higher, which demonstrates the problem of the term compression ratio for lossy encoders.
Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess the MP3 compression algorithm. This song was chosen because of its 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).[citation needed]
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.
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 which 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 Musicam (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 Union as a part of the EUREKA research program where it was commonly known as EU-147, which ran from 1987 to 1994.
As a doctoral student at Germany's University of Erlangen-Nuremberg, Karlheinz Brandenburg began working on digital music compression in the early 1980s, focusing on how people perceive music. He completed his doctoral work in 1989 and became an assistant professor at Erlangen-Nuremberg. While there, he continued to work on music compression with scientists at the Fraunhofer Society (in 1993 he joined the staff of the Fraunhofer Institute).[9]
In 1991, there were two proposals available: Musicam (known as Layer 2), 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. The Musicam format, based on sub-band coding, was a basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame). Its technology and ideas were fully incorporated into the definition of ISO MPEG Audio Layer I and Layer II and further on of the Layer III (MP3) format. Under the chairmanship of Professor 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 rate depends on the bit depth and sampling rate of the input signal. Nevertheless, there are often published compression rates that use the CD parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2×16 bit). Sometimes the Digital Audio Tape (DAT) SP parameters are used (48 kHz, 2×16 bit). Compression ratios with this reference are higher, which demonstrates the problem of the term compression ratio for lossy encoders.
Karlheinz Brandenburg used a CD recording of Suzanne Vega's song "Tom's Diner" to assess the MP3 compression algorithm. This song was chosen because of its 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).[citation needed]
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, beause 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. Gabriel Bouvigne, a principal developer of the LAME project, says that the freeformat option is compliant with the standard but, according to the standard, decoders are only required to be able to decode streams up to 320 kbit/s
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, beause 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. Gabriel Bouvigne, a principal developer of the LAME project, says that the freeformat option is compliant with the standard but, according to the standard, decoders are only required to be able to decode streams up to 320 kbit/s
File structure
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