PCM – which is short for Pulse Code Modulation – is often mistakenly referred to as an audio format but it is in fact something completely different. It is a method used to digitally represent analogue signals as well as convert analogue signals into digital and back. Most digital audio devices (except DSD equipment) use PCM encoding and transcoding since it is simple to use with digital signal processors. PCM-signals are made by regularly measuring the amplitude of a signal. Every measurement is called a sample and every sample represents an amplitude value (quantization level) in binary numbers. The more bits there are, the higher the dynamical range and the better the resolution.
For example, an audio file with a sample rate of 44.1 kHz and a bitrate of 16 bit has 44100 amplitude measurements with a 16 bit binary resolution. For audio purposes a specific type of PCM is used, namely LPCM which stands for Linear Pulse Code Modulation. LPCM is used because of the linearity of quantization levels when working with audio. Usually when PCM is mentioned for audio purposes, it’s assumed to be LPCM.