Information AboutMp3 |
MPEG-1 Audio Layer 3, more commonly referred to as '''MP3''', is an Audio encoding format. It uses a Lossy Compression Algorithm that is designed to greatly reduce the amount of data required to represent the audio recording, yet still sound like a faithful reproduction of the original uncompressed audio to most listeners. It was invented by a team of European engineers at Philips , CCETT (Centre commun d'études de télévision et télécommunications), IRT and Fraunhofer Society , who worked in the framework of the EUREKA 147 DAB digital radio research program, and it became an ISO / IEC standard in 1991 . MP3 is an audio-specific format. The compression removes certain parts of sound that are outside the normal human hearing range so cannot be heard by the listener. It provides a representation of Pulse-code Modulation — encoded audio in much less space than straightforward methods, by using Psychoacoustic models to discard components less audible to human hearing, and recording the remaining information in an efficient manner. Similar principles are used by JPEG , an image compression format. DEVELOPMENT The psychoacoustic masking codec was first proposed, apparently independently in 1979, by Manfred Schroeder, ''et. al.''"Optimizing Digital Speech Coding by Exploiting Masking Properties of the Human Ear"; M. R. Schroeder, B. S. Atal and J. L. Hall; J. Acoust. Soc. Am.; received 8 June 1979; accepted for publication 13 August 1979; Dec. 1979 in Germany and M. A.Krasner"Digital Encoding of Speech and Audio Signals Based on the Perceptual Requirements of the Auditory System"; M. A. Krasner; Massachusetts Institute of Technology Lincoln Laboratory Technical Report 535; 18 June 1979 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 laboratories Technical Report did not immediately influence the mainstream of psychoacoustic coder development. Manfred Schroeder was already a well known and revered figure in the world wide community of acoustical and electrical engineers and his paper had immediate influence in European and specifically German circles of acoustic and source-coding (audio compression) research. Both Krasner and Schroeder built upon the work of E. F. Zwicker."On the Psychoacoustical Equivalent of Tuning Curves"; E. F. Zwicker; Proceedings of the Symposium on Psychophysical Models and Physiological Facts in Hearing; held at Tuzing, Oberbayern, April 22-26, 1974 The immediate predecessor of MP3, and the first practical implementation in hardware (Krasner's hardware was too cumbersome and slow for practical use), was "Optimum Coding in the Frequency Domain","OCF: Coding High Quality Audio with Data Rates of 64 KBit/sec; K. Brandenburg, D. Seitzer; Universitaet Erlangen-Nuernberg, Erlangen; Presented at the 85th Convention of the Audio Engineering Society; Los Angeles; November 3-6 1988 which was an implementation of a psychoacoustic transform coder based on Motorola 56000 DSP chips. MP3 is directly descended from OCF. MP3 represents the outcome of the collaboration of Dr. Karl Heinz Brandenburg with the Fraunhofer Society for Integrated Circuits, Erlangen, with relatively minor contributions from the Musicam (MP2) branch of psychoacoustic sub-band coders. Modern lossy bit compression technologies, including MPEG and MP3, are based on the early work of Prof Oscar Bonello of the University of Buenos Aires, Argentina. He was involved in studio equipment design for broadcast radio automation. At the same time he taught acoustics at the University (he is the author of the "Bonello Criterion" for room acoustics design), with psychoacoustics being his main field of research. In 1983, he started researching the idea of using the Critical Band Masking principle (a property of the ear) in order to reduce the bit stream needed to encode an audio signal. The masking principle was discovered in 1924 and further developed by Egan-Hake and Richard Ehmer in 1959. Bonello's work created, in 1987, the world's first bit compression system, named ECAM, working in real time and implemented by hardware on an IBM PC computer. This plug in card and the associated control software was demonstrated for the first time in 1988 as a fully working product named Audicom and introduced to the world at the international NAB Radio Exhibition in Atlanta, USA on 1990. The basic Bonello implementation is now used in MP3 and other systems. Bonello refuses to apply for any patents around this technology. Masking by Tones vs Noise Bands Richard Ehmer ASA Journal, Vol 3, Number 9, September 1959 The invention of Audicom : Summary of some of Solidyne's contributions to Broadcast Engineering 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).1 In (known as ''Layer 2''), and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (The Netherlands), CCETT (France) and Institut Für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated with the encoding of high quality compressed audio. The Musicam format, based on Sub-band Coding , was a key to settle the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame). Its technology and ideas were fully incorporated into the definition of ISO MPEG Audio Layer I and Layer II and further on of the Layer III (MP3) format. Under the chairmanship of Professor Mussmann ( University Of Hannover ) the editing of the standard was made under the responsibilities of Leon Van De Kerkhof (Layer I) and Gerhard Stoll (Layer II). A Working Group consisting of Leon Van De Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) took ideas from Musicam and ASPEC, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 Kbit/s as MP2 at 192 kbit/s. All algorithms were approved in 1991, finalized in 1992 as part of MPEG-1 , the first standard suite by MPEG , which resulted in the international standard '' ISO / IEC 11172-3'', published in 1993. Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, MPEG-2 , more formally known as international standard ''ISO/IEC 13818-3'', originally published in 1995. Compression efficiency of encoders is typically defined by the bit rate, because compression rate depends on the bit depth and Sampling Rate of the input signal. Nevertheless, there are often published compression rates that use the CD parameters as references (44.1 KHz , 2 channels at 16 bits per channel or 2×16 bit). Sometimes the Digital Audio Tape (DAT) SP parameters are used (48 kHz, 2×16 bit). Compression ratios with this reference are higher, which demonstrates the problem of the term ''compression ratio'' for lossy encoders. Karlheinz Brandenburg used a CD recording of Suzanne Vega 's song " Tom's Diner " to assess the MP3 Compression Algorithm . This song was chosen because of its softness and simplicity, making it easier to hear imperfections in the compression format during playbacks. Some jokingly refer to Suzanne Vega as "The mother of MP3". Some more critical audio excerpts ( Glockenspiel , Triangle , Accordion , etc.) were taken from the EBU V3/SQAM reference compact disc and have been used by professional sound engineers to assess the subjective quality of the MPEG Audio formats. GOING PUBLIC A reference simulation software implementation, written in the C language and known as ''ISO 11172-5'', was developed by the members of the ISO MPEG Audio committee in order to produce bit compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). Working in non-real time on a number of operating systems, it was able to demonstrate the first real time hardware decoding (DSP based) of compressed audio. Some other real time implementation of MPEG Audio encoders were available for the purpose of digital broadcasting (radio DAB, television DVB) towards consumer receivers and set top boxes. Later, on July 7 1994 the Fraunhofer Society released the first software MP3 encoder called L3enc . The Filename Extension ''.mp3'' was chosen by the Fraunhofer team on July 14 , 1995 (previously, the files had been named ''.bit''). With the first real-time software MP3 player Winplay3 (released September 9 , 1995 ) many people were able to encode and playback MP3 files on their PCs. Because of the relatively small Hard Drive s back in that time (~ 500 MB ) the technology was essential to store non-instrument based (see Tracker and MIDI ) music for listening on a computer. MP2 In October 1993, MP2 (''MPEG-1 Audio Layer 2'') files appeared on the Internet and were often played back using the '' Xing MPEG Audio Player'', and later in a program for Unix by Tobias Bading called MAPlay , which was initially released on February 22 , 1994 (MAPlay was also ported to Microsoft Windows ). Initially the only encoder available for MP2 production was the Xing Encoder, accompanied by the program Cdda2wav , a CD Ripper used for extracting CD audio tracks to Waveform Audio Files . The Internet Underground Music Archive (IUMA) is generally recognized as the start of the on-line music revolution. IUMA was the Internet's first high-fidelity music web site, hosting thousands of authorized MP2 recordings before MP3 or the web was popularized. INTERNET In the first half of 1995 through the late 1990s , MP3 files began to spread on the Internet . MP3's popularity began to rise rapidly with the advent of Nullsoft 's audio player Winamp (released in 1997), the Unix audio player Mpg123 and the peer-to-peer file sharing network Napster (released in 1999). These programs made it simple for average users to play back, create, share and collect MP3s. The small size of MP3 files has enabled widespread Peer-to-peer File Sharing of music, which would previously have been nearly impossible. The major record companies, who argue that such free sharing of music reduces sales, reacted to this by pursuing law-suits against Napster , which was eventually closed down, and eventually against individual users who engaged in file sharing. Napster has now returned, albeit in a slightly different form. These legal actions have had little effect on the production and distribution of MP3 audio. Despite the popularity of MP3, online music retailers often use other proprietary formats that are encrypted (known as Digital Rights Management ) to prevent users from using purchased music in ways not specifically authorised by the record companies. The record companies argue that this is necessary to prevent the files from being made available on peer-to-peer file sharing networks. However, this has other side effects such as preventing users from playing back their purchased music on different types of devices. Some services, such as EMusic , continue to offer the MP3 format, which allows users to playback their music on virtually any device. ENCODING AUDIO The MPEG-1 standard does not include a precise specification for an MP3 encoder. The decoding algorithm and file format, as a contrast, are well defined. Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information in the raw audio (or rather its MDCT representation in the frequency domain). During encoding 576 time domain samples are taken and are transformed to 576 frequency domain samples. If there is a Transient , 192 samples are taken instead of 576. This is done to limit the temporal spread of quantization noise accompanying the transient. (See Psychoacoustics .) As a result, there are many different MP3 encoders available, each producing files of differing quality. Comparisons are widely available, so it is easy for a prospective user of an encoder to research the best choice. It must be kept in mind that an encoder that is proficient at encoding at higher bit rates (such as LAME , which is in widespread use for encoding at higher bit rates) is not necessarily as good at other, lower bit rates. DECODING AUDIO Decoding, on the other hand, is carefully defined in the standard. Most Decoder s are " Bitstream compliant", meaning that the decompressed output they produce from a given MP3 file will be the same (within a specified degree of Rounding tolerance) as the output specified mathematically in the ISO/IEC standard document . The MP3 file has a standard format, which is a frame consisting of 384, 576, or 1152 samples (depends on MPEG version and layer) and all the frames have associated header information (32 bits) and side information (9, 17, or 32 bytes, depending on MPEG version and stereo/mono). The header and side information help the decoder to decode the associated Huffman encoded data correctly. Therefore, comparison of decoders is usually based on how computationally efficient they are (i.e., how much Memory or CPU time they use in the decoding process). AUDIO QUALITY When creating an MP3 file, there is a trade-off between the amount of space used and the sound quality of the result. Typically, the creator of the MP3 file is allowed to set a Bit Rate , which specifies how many Kilobits the file may use per second of audio, for example, when Ripping a Compact Disc to this Format . The lower the bit rate used, the lower the audio quality will be, but the smaller the file size. Likewise, the higher the bit rate used, the higher quality, and therefore, larger the file size the resulting MP3 will be. As described, MP3 files encoded with a lower bit rate will generally play back at a lower quality. With too low a bit rate, " Compression Artifact s" (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 nominal bitrate provides a good example of compression artifacts.
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| Author | David Meares, Kaoru Watanabe & Eric Scheirer |
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| Title | Report on the MPEG-2 AAC Stereo Verification Tests |
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| Publisher | International Organisation For Standardisation |
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| Date | 1998–02 |
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| Url | http://soundmediamitedu/mpeg4/audio/public/w2006pdf |
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| Format | PDF |
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