MP3 Sound Quality Part 2
The MP3 format uses a hybrid conversion mechanism to convert the time domain signal to the frequency domain signal:
* 32 band polyphase integrator filter (PQF)
* 36- or 12-lead Modified Discrete Cosine (MDCT) filter; each subband size can be set to 0 Independently selectable between …1 and 2…31
* alias attenuation post-processing
According to the MPEG specification, AAC (Advanced Audio Coding) in MPEG-4 will be the next generation of the MP3 format, although there are many significant efforts to create and promote other formats. 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. Developing
historical MPEG-1 Audio Layer 2 encoding began as Digital Audio Broadcasting (DAB) managed by Egon Meier-Engelen at the German Deutsche Forschungs- und Versuchsanstalt für Luft- und Raumfahrt (later known as Deutsches Zentrum für Luft- und Raumfahrt, German Space Center). )Project. 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 (sample rate, frame structure, 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 international standard ISO/IEC 11172-3 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 (:in:bit depth) and the sampling rate of the input signal. However, there are often products that use CD parameters (44.1 kHz, two channels, 16 bits per channel, or 2×16 bits) as the compression ratio reference, and the compression ratio using this reference is usually higher, which which also shows that the compression ratio is very important for lossy compression problems.
Karlheinz Brandenburg used Suzanne Vega’s song Tom’s Diner on CD to test MP3 compression algorithms. This song is used because the song’s smooth and simple melody makes it easier to hear glitches in the compressed format during playback. Some jokingly refer to Suzanne Vega as “the mother of MP3”. The most serious and critical audio extracts (glockenspiel, triangle, accordion, …) from the EBU V3/SQAM Reference CD are used by professional audio engineers to assess the subjective perceived quality of the MPEG audio format.
