The process of converting analog audio signals into digital audio signals is called sampling. Simply put, it means how many data points are needed to record a 1-second sound through waveform sampling.
eg: The sound with a 44.1KHz sampling rate takes 44,000 data points to describe the sound waveform for 1 second. In principle, the higher the sampling rate, the better the sound quality.
Sampling frequencies are generally divided into three levels: 22.05KHz, 44.1KHz, and 48KHz; 22.05KHz can only achieve the sound quality of FM radio, 44.1KHz is the theoretical limit of CD sound quality, and 48KHz It has reached DVD sound quality.
The sampling rate refers to the sampling frequency when converting sound (analog signal) into mp3 (digital signal), that is, how many points of data are sampled per unit time.
Bit rate refers to the number of bits transmitted per second. The unit is bps (Bit Per Second). The higher the bit rate, the larger the data transmitted and the better the sound quality.
Bit rate refers to the number of bits transmitted per second. The unit is bps (Bit Per Second). The higher the bit rate, the more data is transmitted per second and the clearer the image quality.
The bit rate in sound refers to the amount of binary data per unit time after converting the analog sound signal into a digital sound signal. It is an indirect indicator of audio quality. The principle of bit rate (code rate) in video is the same as that in sound. They both refer to the amount of binary data per unit time after converting an analog signal into a digital signal.
It can be said that the sampling rate and bit rate are like the horizontal and vertical coordinates on the coordinate axis. The sampling rate on the abscissa represents the sampled data points per second. The bit rate on the ordinate represents the accuracy when using digital quantities to quantize analog quantities.
Extended information:
The meaning of sampling:
Sound is actually a kind of energy wave, so it also has the characteristics of frequency and amplitude. The frequency corresponds to the time axis. , the amplitude corresponds to the level axis. The wave is infinitely smooth, and the string can be seen as composed of countless points. Since the storage space is relatively limited, the points of the string must be sampled during the digital encoding process.
The process of sampling is to extract the frequency value of a certain point. Obviously, the more points extracted in one second, the richer the frequency information obtained. In order to restore the waveform, there must be 2 For sampling points, the highest frequency that the human ear can detect is 20kHz.
Therefore, to meet the hearing requirements of the human ear, it is necessary to sample at least 40k times per second, expressed in 40kHz, and this 40kHz is the sampling rate. Our common CD has a sampling rate of 44.1kHz.
It is very important to synchronize video and audio during the acquisition process. Frequency information alone is not enough. We must also obtain the energy value of the frequency and quantify it to represent the signal strength. The number of quantization levels is an integer power of 2. Our common CD bit sampling size of 16 levels is 2 to the 4th power.
Sampling size is more difficult to understand than sampling rate because it needs to be abstract. Let’s take a simple example: Suppose a wave is sampled 8 times, and the energy values ??corresponding to the sampling points are A1-A8, but We only use a 2-bit sampling size, and as a result we can only retain the values ??of 4 points in A1-A8 and discard the other 4.
If we use a 3-bit sampling size, all the information of 8 points will be recorded. The larger the values ??of sample rate and sample size, the closer the recorded waveform is to the original signal.
Baidu Encyclopedia-Audio Sampling Rate
Baidu Encyclopedia-Bit Rate