
Principle of mp3 and file format analysis.

Principle of mp3 and file format analysis

1. Introduction
With the rapid development of file compression technology, MP3 has become the most popular music format today. High-quality music spreads rapidly around the world with the arrangement of 0 and 1, which shakes people’s hearts. What is MP3? The full name of MP3 is MPEG Audio Layer 3, which is an efficient computer audio coding scheme. It converts audio files into smaller files with an .MP3 extension with a higher compression ratio, basically maintaining the sound quality of the original file. MP3 is part of the ISO/MPEG standard, which describes audio compression using a high-performance perceptual coding scheme. This standard has been continuously updated to meet the pursuit of “high quality and low quality”, and has now formed MPEG Layer 1, Layer 2, Layer 3 three audio encoding and decoding schemes. MPEG Layer 3 compression ratio can reach 1:10 to 1:12, 1M of MP3 file can be played for 1 minute and 1 minute of CD-quality WAV file (44100Hz, 16bit, dual channel, 60 seconds) occupies 10M space, so Calculated, the playing time of a 650M MP3 disc should be more than 10 hours, and the playing time of a CD of the same capacity is about 70 minutes. The advantage of MP3 is that the CD is incomparable.
2 Analysis of the principle of MP3
2.1 audio standard
MPEG MPEG (Moving Picture Experts Group) is a group of dynamic picture experts under ISO, the MPEG standard which makes it widely used in various multimedia. The MPEG standards include audio and video standards, of which the audio standards have been established as MPEG-1, MPEG-2, MPEG-2 AAC, and MPEG-4.
The MPEG-1 and MPEG-2 standards use the same family of audio codecs: Layer 1, 2, 3. A new feature of MPEG-2 is the use of low sample rate expansion to reduce the data stream, and another feature is multichannel expansion, which increases the number of main channels to 5. The MPEG-2 AAC (MPEG-2 Advanced Audio Coding) standard was released by Fraunhofer IIS and AT&T in 1997 to significantly reduce data traffic. The MDCT (Modified Discrete Cosine Transform) algorithm adopted by MPEG-2 AAC has a sampling frequency between 8KHz and 96KHz, the number of channels can be between 1-48.
The three layers of MPEG Audio Layer 1, 2, and 3 use the same filter bank, bitstream structure, and header information, and the sampling frequency is 32KHz, 44.1KHz, or 48KHz. Layer 1 is designed for DCC (Digital Compact Cassette) compressed digital tape, the data rate is 384kbps, Layer 2 has made a compromise between complexity and performance, and the data rate is reduced to 256kbps-192 kbps. Layer 3 is designed for low data traffic from the start, and the data traffic is 128Kbps-112Kbps. Layer 3 adds MDCT transformation to make its frequency resolution 18 times that of layer 2. Layer 3 also uses average information similar to MPEG video. Entropy Encoding reduces redundant information. The vast majority of MP3s use the MPEG-1 standard.
2.2 Purpose of audio compression
The MP3 format began in the mid-1980s, when the Fraunhofer Institute in Erlangen, Germany, dedicated itself to encoding high-quality, low-data-rate sound. Let’s look at an example: you want to sample a song you like that is about 4 minutes long, store it on a disk, sample it in CD-quality WAV format, at a sample rate of 44.1 kHz, that is, receive a value of 44100 per second, stereo, each sampled data is 16 bits (2 bytes), so the space this song occupies is:
44100 x 2 channels x 2 bytes x 60 seconds x 4 minutes = 40.4 MB
If you download this song from the Internet, assuming the transmission speed is 56 kbps, the download time is:
40.4x106x8/56x103x60=96 minutes
Even a 1M broadband network requires more than 5 minutes, it can be seen that audio compression is particularly important to reduce audio data storage space.
2.3 Encoding and decoding
MP3 MP3 audio compression consists of two parts: encoding and decoding. Encoding converts the data in a WAV file into a highly compressed bitstream, and decoding takes the bitstream and reconstructs it into a WAV file.














