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1. Introduction to MP3:<\/p>\n
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The full name of MP3 is MPEG 1 audio layer 3, of which the MPEG (Moving Picture Experts Group) standards include video and audio standards.
\n, of which MPEG-1, MPEG-2, MPEG-2 AAC and MPEG-4 audio standards have been formulated.
\nMPEG-1 and MPEG-2 standards use the same family of audio codecs: Layer 1, Layer 2, Layer 3. It is divided according to compression quality
\nand the complexity of encoding, corresponding to MP1, MP2, MP3, three types of sound files, and uses different levels
\nencoding for different purposes. The higher the level of MPEG audio encoding, the more complex the encoder and the higher the compression rate.
\nA new feature of MPEG-2 is the use of low sample rate expansion to reduce data throughput, and another feature is multi-channel expansion. , which increases the number of main channels to 5 .
\nAll three layers of MPEG Audio Layer 1, Layer 2, and Layer 3 use the same filter bank, bitstream structure, and header information, and the sampling rate is
\n32KHz, 44.1KHz, 48KHz
\nLayer 1 is designed for DCC (digital compact cassette) compressed digital tape, 384 kbps data stream; compression ratio 4:1;
\nLayer 2 trade-offs between complexity and performance, data throughput drops to 256kbps-192kbps; compression ratio 6:1-8:1;
\nLayer 3 is designed for low data traffic from the start, the data traffic is 128kbps-112kbps, and the compression ratio is as high as 10:1-12:1;
\nLayer 3 adds the MDCT transform, making its frequency resolution 18 times that of Layer 2. Layer 3 also uses
\nentropy coding, which is similar to MPEG video, to reduce redundant information. The vast majority of MP3s use the MPEG-1 standard.
\nMP3 audio quality depends on its bit rate and sample rate, as well as the quality of the encoder. Typical MP3 speeds are between
\n128 and 320 kb per second (problem here). The sample rate also has three frequencies: 32, 44.1 and 48 kHz. The most common is to use the
\nCD sample rate: 44.1 kHz. The commonly used encoder is LAME, which fully follows the LGPL MP3 encoder and has good speed and sound quality.
\nMP3 uses a lossy compression method for audio signals. In order to reduce the degree of sound distortion, MP3 adopts “sensory coding technology”, that is,
\ndiscards data that is not important to the human ear in pulse code modulation (PCM) Audio data Higher compression ratio, i.e. the file
\naudio , and then the noise level is filtered out with a filter, and then each remaining bit is spread out and organized by quantization, and finally
\na higher compression ratio is formed .MP3 files, so that compressed files can achieve sound effects closer to the original sound source during playback.
\n3. MP3 encoding and decoding process
\nMP3 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. MP3 uses perceptual audio coding (Perceptual Audio Coding) this distortion algorithm. The frequency range of sound perceived by the human ear is 20 Hz to 20 kHz. MP3 cuts out a lot of redundant signals and irrelevant signals. The encoder transforms the original sound into the frequency domain through a mixed filter bank and uses a psychoacoustic model. to estimate that it may be only The perceived noise level is quantized and converted to Huffman coding to form an MP3 bitstream. The decoder is much simpler, its task is to extract the sound signal from the encoded spectral line components through inverse quantization and inverse transformation.
\nMP3 decoding can be divided into 9 processes in general: bitstream analysis, Huffman encoding, inverse quantization processing, stereo processing, spectral rearrangement, anti-aliasing processing, IMDCT transformation, subband synthesis, PCM output.
\nBriefly describe the MP3 compression process: Sound is an analog signal, and sound is sampled, quantized, and encoded to obtain PCM data.<\/p>\n","protected":false},"excerpt":{"rendered":"