Music sampling-digital audio system reproduces the original sound by converting the sound waveform into a series of binary data. The device used to achieve this step is an analog/digital converter (A/D), which converts the sound waveform into a series of binary data. The sound wave is sampled at a rate of tens of thousands of times per second. Each sampling records the state of the original simulated sound wave at a certain moment, which is called a sample. A sound wave can be described by connecting a series of samples. The number of samples per second is called the sampling frequency or sampling rate. The unit is HZ (Hertz). The higher the sampling frequency, the higher the frequency of sound waves that can be described. The sampling rate determines the range of sound frequencies (equivalent to pitch) that can be represented by a digital waveform. The range of frequencies represented by a waveform is often called the bandwidth. To correctly understand audio sampling can be divided into the number of sampling bits and sampling frequency.