Pulse code modulation (PCM). The most commonly used method for digital signal encoding of analog data is pulse code modulation (PCM), which is usually used to encode sound signals. Pulse code modulation is based on the sampling theorem. It is mathematically proved that if the continuously changing analog signal is sampled periodically, as long as the sampling frequency is greater than or equal to twice the maximum frequency of the effective signal or its bandwidth, the sampled value can contain all the information of the original signal, and the original signal can be reconstructed from these samples by using the low-pass filter.