How does MP3 work?


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How does MP3 work?

mp3 compression

1. The common name for portable MP3 player. a portable player
used to play music in MP3 format (now compatible with wma, wav and other formats).

MP3 Compression

Portable MP3 Player was originally developed by Korean Wenguang Su and Huang Dingxia (Moon & Hwang) Invented in 1997 and applied for related patents Detailed explanation of the
technology development of the MP3 format Format, which is designed to drastically reduce the amount of audio data, while for most users the playback quality is not appreciably degraded from the original uncompressed audio. It was invented and standardized in 1991 by a group of engineers from the Fraunhofer-Gesellschaft research organization in Erlangen, Germany. MPEG-1 Audio Layer 3, often referred to as MP3, is one of the most popular lossy compression and digital audio encoding formats today. There is no noticeable drop in sound quality compared to the original uncompressed audio. It was invented and standardized in 1991 by a group of engineers from the Fraunhofer-Gesellschaft research organization in Erlangen, Germany. General information MP3 is a data compression format. It discards pulse code modulation (PCM) audio data that is not important to the human ear (similar to how JPEG is lossy image compression), resulting in a much smaller file size. Various techniques are used in MP3, including psychoacoustics, to determine which parts of the audio can be discarded. MP3 audio can be compressed at different bit rates, providing a variety of trade-offs between data size and sound quality. The MP3 format uses a hybrid conversion mechanism to convert the time domain signal to the frequency domain signal: * 32-Band Polyphase Integrating Filter (PQF)
* Modified 36 or 12 tap discrete cosine filter (
MDCT); each subband size is independently selectable between 0…1 and 2…31 However, due to the unprecedented popularity of MP3, the success of any other format is currently unlikely. MP3 not only has extensive client software support, but also has a lot of hardware support, such as portable media players (referring to MP3 players), DVD and CD players.
The development of
MPEG-1 Audio Layer 2 encoding started with the German Deutsche Forschungs- und Versuchsanstalt für Luft- und Raumfahrt (later called Deutsches Zentrum für Luft- und Raumfahrt, German Space Center) Digital Audio Broadcasting (DAB) managed by Egon Meier-Project Engelen . This project is funded by the European Union as a EUREKA research project, and its name is commonly known as EU-147. The study period for EU-147 was from 1987 to 1994.
By 1991 there were already two proposals: Musicam (called Layer 2) and ASPEC (Adaptive Spectrum Sensing Entropy Coding). The Musicam method proposed by Philips of the Netherlands, CCETT of France, and the Institut für Rundfunktechnik of Germany was chosen due to its simplicity, robustness against errors, and lower computational effort in high-quality compression. The Musicam format based on subband coding is a key factor in determining the MPEG audio compression format (sampling rate, frame structure, data header, sample points per frame). This technology and its design philosophy are fully integrated into the definition of ISO MPEG Audio Layer I, II and later Layer III (MP3) formats. The standard was developed by Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II) under the auspices of Prof. Mussmann (University of Hannover).
A working group consisting of Leon Van de Kerkhof from the Netherlands, Gerhard Stoll from Germany, Yves-François Dehery from France, and Karlheinz Brandenburg from Germany absorbed design ideas from Musicam and ASPEC and added their own design ideas to develop MP3 , which can play MP2. Sound quality from 192kbit/s to 128kbit/s.
All of these algorithms eventually became part of the first group of MPEG standards, MPEG-1, in 1992, resulting in the ISO/IEC 11172-3 international standard published in 1993. Further work on MPEG audio eventually became part of the MPEG-2 standard, a second group of MPEG standards developed in 1994, officially known as ISO/IEC 13818-3, first published in 1995.
The compression efficiency of an encoder is usually defined by the bit rate, since the compression rate depends on the number of bits.


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MP3: features and alternatives

The peculiarities of the MP3 format and some clues about other solutions of equal or even higher quality.

Impossible to deny, the MP3 format is the most common and most enjoyable to listen to music on the go or, as it has been for some years, streaming. We use it everywhere now and any device can play it today.

MP3 is part of the family of audio files called “lossy”, that is, the types of formats that can also reduce the amount of data that should contain a sound, in any case try to maintain at least an acceptable quality.

The peculiarities of the MP3 format and some clues about other solutions of equal or even higher quality.
The parameters that determine the quality level of an MP3 file are: the sampling rate, bit rate, encoder and of course the source. Now let’s move on to the order.
At the origin of everything is the source, that is, the support or source from which the MP3 file can be downloaded. The higher the quality of the source, the greater the end result: purchasing MP3s from particularly reliable sites or extracting them from compact discs in good condition is the basis for a successful MP3. What becomes crucial is the encoder (the most famous and free is LAME) or the software that takes care of creating the file after properly configuring its parameters.

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The sampling rate is measured in Herz and expresses the number of times per second. Second, as the analog signal is measured and digitized; for MP3 it must be as faithful as possible on a CD, ie 44 100 Hz (44.1 KHz).

Bitrate is the number of binary units flowing, measured every second. The value of the bit rate is not fixed: as it increases, the similarity to the original file will also increase proportionally. The higher the bit rate, the higher the quality, the larger the file size. The bit rate range ranges from 32 kbps to 320 kbps, the maximum that can be obtained from an MP3 file.

The ones we’ve just listed are an important part of the tricks that allow us to have an MP3 quality; however, be aware that a lost file is by no means faithful in all respects to the original source. The most famous lost alternatives are: AAC (the format Apple uses to sell music in the iTunes Store and since July to stream audio from the Apple Music service); WMA; MPC; OGG (excellent quality open source format).

If you are looking for maximum faith in digital audio, give up MP3 and its loss-free alternatives to switch to “loss-free” audio formats, ie loss-free quality. Overall, this file type compresses the original sound while keeping the number of bits intact. Needless to say, quality comes at a cost in terms of the space taken up: a lossless file takes about half of the original audio file, but “weighs” nearly three times as much as a 320Kbps MP3. Of these, the most famous and used are: FLAC; ALAC (Apple Lossless Format); BEE; WavPack. The “lossy” and “lossless” file distinctions are extremely applicable to images and videos as well, not just audio files.

On several occasions it has been said how absolutely difficult it is to distinguish an MP3 at 320 kbps, obtained under the best conditions, from its original version on CD or in lossless files; It is only possible to notice it with instruments at a certain level and with a good ear. When noted, the MP3 format is excellent for listening on the move, as highlighted above; On the other hand, to better preserve our music or listen to it on systems of a certain level, it is better to resort to lossless formats such as FLAC or ALAC.