MP3: complete analysis of the audio format
Although MP3 technology has been fully disclosed now, its specific technical details still have some depth.
Therefore, this editor will provide a detailed explanation of some technologies and hope that netizens will laugh at it.
Sampling rate:
A digital audio system reproduces the original sound by converting the waveform of the sound wave into a series of binary data. The equipment used to accomplish this step is an analog-to-digital (A/D) converter, which samples the sound wave at a rate of tens of thousands of times per second. , each sample records the state of the original analog sound wave at a given time, called a sample.
A series of samples can be connected to describe a sound wave. The number of samples per second is called the sample rate or sample rate, and the unit is HZ (Hertz). The higher the sample rate, the higher the frequency of the sound wave that can be described. For each sampling system, a certain storage bit (number of bits) is allocated to express the sound wave amplitude state of the sound wave, which is called the sampling resolution or sampling precision. Increase the dynamic range of 6db, that is, the dynamic range of 6db, a 2-bit digital audio system expresses thousands of states, that is, the dynamic range of 12db, and so on. If you keep increasing the number of bits, the sampling precision will increase at a very fast rate. It can be calculated that 16 bits can express 65536 states, corresponding to 96db, and 20 bits can express 1048576 states, corresponding to 120db. 24bit can express up to 16777216 states. Corresponding to the dynamic range of 144db, the higher the sampling precision, the more delicate the restoration of sound waves. (Note: dynamic range refers to the range of the sound from the weakest to the loudest.) The hearing range of the human ear is usually 20HZ~20KHZ.
According to the Nyquist sampling theorem, sampling at twice the frequency of a sine wave can completely restore the waveform, so the sampling frequency of a digital recording wave is directly related to its rate of sampling. higher refresh rate. For example, sampling with a sampling frequency of 44.1 KHZ can restore the highest frequency of 22.05 KHZ — this value is slightly higher than the hearing limit of the human ear (Note: MD can be recorded, please For example, the sampling rate of R900 is 44.1KHZ, it also has a sampling frequency converter, which can convert the input 32KHz/44.1KHZ/48KHZ into the machine’s standard sampling rate 44.1KHZ. The refresh rate is enough to truly record and reproduce the sound that everyone can distinguish, so the sampling specification of CD audio is defined as 16bit 44KHZ, even if 16bit recording is actually done with high precision electronic components that are almost impossible to manufacture in real life in the most ideal environment there will still be issues like sound filtering and positioning people can still detect some tiny distortions so much Many professional digital audio systems have used 18-bit or even 24-bit for recording and playback.
Existing sampling methods:
MP3: The full name of MP3 should be MPEG1 Layer 3 audio file. MPEG (Moving Picture Experts Group) translates into Chinese as Moving Picture Experts Group, which refers to the compression standard for moving video . The MPEG audio file is the sound part in the MPEG1 standard. It is called MPEG audio layer, which is divided into three layers according to compression quality and encoding complexity, namely Layer-1, Layer2, Layer3, and corresponds to the three sound files of MP1, MP2 and MP3 , respectively. , and use different layers according to different purposes. The higher the MPEG audio encoding level, the more complex the encoder and the higher the compression ratio. The compression ratio of MP1 and MP2 is 4:1 and 6:1-8:1 respectively, while the compression ratio MP3 compression ratio is up to 10:1-12:1, that is, one minute of CD-quality music requires 10 MB of storage space without compression, and only about 1 MB after encoding with MP3 compression. However, MP3 uses a lossy compression method for audio signals. To reduce the degree
