音频采样 解释（Audio sampling interpretation）.doc

音频采样 解释（Audio sampling interpretation） Audio sampling interpretation Digital audio system is the acoustic waveform into a series of binary data to reproduce the original sound, the steps of used equipment is the analog / digital converter (A/D) with tens of thousands of times per second rate of acoustic sampling, each sampling record the original analog sound in a moment, known as sample. A string of samples can be connected to describe a section of sound wave. The number of samples sampled per second is called the sampling frequency or recovery rate. The unit is HZ (Hz). The higher the sampling frequency, the higher the acoustic frequency can be described. The sampling rate determines the range of the sound frequency (equivalent to the pitch) and can be represented by a digital waveform. The frequency range represented by waveform is usually called bandwidth. To understand correctly, audio sampling can be divided into the number of bits sampled and the frequency of the samples. 1. sample digit The number of bits sampled can be interpreted as the acquisition cards resolution of the sound. The bigger the value, the higher the resolution, the more realistic the recording and playback sounds. The first thing we need to know is that the sound files in a computer are represented by numbers 0 and 1. So the essence of recording on a computer is to convert analog sound signals into digital signals. On the contrary, when playing, the digital signal is reduced to analog sound signal output. The location of the acquisition card refers to the binary digits of the digital sound signal used by the acquisition card when collecting and playing audio files. The location of the acquisition card objectively reflects the accuracy of the digital sound signal in describing the input sound signal. The 8 represents 2, the 8 Party --256, and the 16 represents the 16 --64K. For comparison, the same piece of music information, 16 sound card can put it into the 64K precision processing unit,