Digital Audio
Pulse Code Modulation, or PCM, is a method of converting analogue audio signals into digital form by sampling the amplitude of the signal at uniform intervals and representing each sample with a numerical value. It is the fundamental process behind nearly all forms of digital audio recording and playback.
Pulse Code Modulation (PCM) is the earliest and most widely used technique for digitally representing analogue signals. In audio applications, the analogue waveform is measured (sampled) at a fixed rate, such as 44.1 kHz or 48 kHz, and each sample is assigned a binary number that describes its amplitude at that instant. The precision of these numbers is determined by the bit depth, commonly 16-bit, 24-bit or higher.
The process involves three stages: sampling, quantisation and encoding. Sampling determines how frequently the waveform is measured; quantisation converts each measurement into the nearest available digital step; and encoding stores or transmits these values as a digital data stream.
PCM does not employ compression or data reduction; it preserves the original waveform as accurately as the sampling rate and bit depth allow. Because of this, PCM serves as the reference standard for high-quality audio in systems such as Compact Disc, Digital Audio Tape, professional digital recorders like the Sony PCM-3324 and Mitsubishi X-80, and virtually all modern DAWs and audio interfaces.
Variants such as linear PCM, used in most professional applications, and non-linear PCM, used in telephony, differ only in how they allocate quantisation steps. PCM forms the basis for later developments like Delta-Sigma conversion, which oversamples the signal before noise-shaping and decimation to a PCM stream.
