MP3: История, Принципы и Влияние на Цифровую Музыку

Digital audio format

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MP3 (formally MPEG 1 Audio Layer III or MPEG 2 Audio Layer III) with support from other digital scientists in other countries. Originally defined 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 the subsequent MPEG 2 standard. A third version, known as MPEG 2.5—extended to better support lower bit rates—is commonly implemented but is not a recognized standard. MP3 (or 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 (the aspect of the standard most apparent to end users and for which it is best known), MP3 uses lossy data compression to encode data using inexact approximations and the partial discarding of data. This allows 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 mid to 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 the file ripping and sharing services MP3. com and Napster, among others. With the advent of portable media players, a product category also including smartphones, MP3 support remains near universal. MP3 compression works by reducing (or approximating) 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. In the early 2000s, compact disc players increasingly adopted support for playback of MP3 files on data CDs. 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, was approved as a committee draft for an ISO/IEC standard in 1991, 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 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. did not immediately influence 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 a core part of the MP3 algorithm. Ernst Terhardt and other collaborators constructed an algorithm describing auditory masking with high accuracy in 1982. 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 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. after having worked on MUSICAM with Matsushita and Philips since 1989. 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 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 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 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 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. ), 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 the staff of Fraunhofer HHI. was chosen due to its simplicity and error robustness, as well as for its high level of computational efficiency. 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. 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 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 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. 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.

Further work on MPEG audio 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 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 was sold afterward in 1998, despite legal suppression efforts by the RIAA. 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 is limited by the small long block window size, which decreases 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, 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 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.