Tag: mp3 to wav format

MP3 File Structure Analysis Part 2

Sounds in nature are very complex and waveforms are extremely complex.

Usually we use pulse code modulation coding, that is, PCM coding. PCM converts continuously changing analog signals into digital codes through three steps of sampling, quantizing, and encoding.

u Decode:

Reverse encoding process

1.1.2 Brief introduction of MP3
The full name of MP3 is MPEG Audio Layer 3. It is an efficient computer audio coding scheme. It converts audio files into smaller files with a .mp3 extension with a higher compression ratio, essentially maintaining the sound quality of the source file. MP3 is part of the ISO/MPEG standard,

The ISO/MPEG standard 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”. Three audio codec schemes, MPEG Layer1, Layer2 and Layer3, have been formed, respectively, corresponding to the three sound files MP1, MP2 and MP3

MPEG (Moving Picture Experts Group) is a group of moving picture experts under ISO. The MPEG standard it specifies is widely used in various multimedia. The MPEG standard includes video and audio standards. Audio standards have developed MPEG-1, MPEG -2, MPEG-2 ACC, MPEG-4. The MPEG-1 and MPEG-2 standards use the same family of Layer1, 2, 3 audio codecs, and most MP3s use the MPEG1 standard.

MP3 audio compression consists of two parts: encoding and decoding. Encoding is the process of converting the original signal to a level signal, and decoding is the reverse process. MP3 uses the PerceptualAudio Coding 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 bit stream. 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.

MP3 file data consists of multiple frames, and a frame is the smallest unit of an MP3 file. Each frame, in turn, consists of a frame header, additional information, and sound data. The playback time of each frame is 0.026 seconds and its duration varies with the bit rate. Some MP3 files have extra bytes at the end that contain description information for non-audio data.

MP3 file structure analysis

ü ID3:

Usually located in several bytes at the beginning or end of an mp3 file, it records the singer, title, album name, era, style, and other mp3 file information.

ID3 is divided into two versions, the V1 ID3 version is fixed at the end of the 128-word file section, it begins with the TAG character, if there is no ID3V1 information, it is considered that there is no ID3V1 information, the V2 ID3 version is found. at the beginning of mp3 and the length is variable.

ü Sampling rate:

The number of samples extracted from a continuous signal to form a discrete signal per second. It is expressed in Hertz (Hz). Sampling rate refers to the sampling frequency when converting an analog signal to a digital signal, i.e. how many points are sampled per unit of time. The higher the sample rate, the more realistic and natural the sound will be. On today’s major capture cards, the sample rate is generally divided into three levels: 22.05 KHz, 44.1 KHz, and 48 KHz. 22.05 KHz can only achieve the sound quality of FM radio, and 44.1 KHz is the theoretical limit of CD sound quality, and 48 KHz is more accurate.

ü Bit rate:

Bit rate refers to the number of bits (bits) transmitted per second. The unit is bps (bit per second). The higher the bit rate, the more information transmitted. In the audio and video fields, bit rate often translates to bit rate. The bit rate indicates how many bits per second the encoded (compressed) audio and video data should represent, and a bit is the smallest unit in binary. 0 or 1. The relationship between bitrate and audio and video compression is simply that the higher the bitrate, the better the quality of the audio and video, but the larger the encoded file; if the bitrate is lower, the situation is just the opposite.

Bit rate = sample rate * number of samples * number of channels

ü Bitrate/Stream/Bitrate:

It refers to the data stream used by audio and video files in a unit of time. The popular understanding is the sample rate, which is the most important part of quality control in audio and video encoding. Generally, the units we use are Kb/s and Mb/s. . Generally speaking, the higher the code stream, the lower the compression ratio and the higher the quality. The higher the code stream, the higher the sampling rate per unit time, the higher the data stream, the higher the accuracy, and the closer the processed file is to the original file.

ü Code:

From the point of view of information theory, the data that describes the source of information is the sum of the redundancy of information and data, namely: data = information + data redundancy. The audio signal has correlation in the time domain and the frequency domain, that is, there is data redundancy. Taking audio as the source, the essence of audio encoding is to reduce redundancy in the audio.