The digital audio system reproduces the original sound by converting the acoustic waveform into a series of binary data, and the process of converting the analog audio into digital audio is called sampling. The device used to implement this step is an analog-to-digital converter (A/D) that samples the sound waves at a rate of tens of thousands of times per second. Each sample records the state of the original analog sound wave at a certain moment, called for the sample. By connecting a series of samples, you can describe a sound wave. The number of samples sampled per second is called the sampling frequency or recovery rate, and the unit is HZ (Hertz). The higher the sampling frequency is, the higher the frequency of the sound wave that can be described, and the better the sound quality. Most MP3s support 44.1kHz, and there are very few MP3 players that support 48kHz MP3 files.
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