Use this tag for questions about the Pulse-Code Modulation representation of sampled signals.
Pulse-Code Modulation (PCM) is a method used to digitally represent sampled analog signals. It is the standard form for digital audio in computers and various Blu-ray, DVD and Compact Disc formats, as well as other uses such as digital telephone systems. A PCM stream is a digital representation of an analog signal, in which the magnitude of the analog signal is sampled regularly at uniform intervals, with each sample being quantized to the nearest value within a range of digital steps. PCM streams have two basic properties that determine their fidelity to the original analog signal: the sampling rate, which is the number of times per second that samples are taken; and the bit depth, which determines the number of possible digital values that each sample can take.
Pulse-code modulation is an uncompressed audio encoding method. PCM is used in the .WAV, and .PCM file formats, and for encoding audio CDs, among many other uses. Digital signal processing algorithms like filters or FFT interact with the data when its in PCM format. When compressed audio format conversions are made, say from aac to flac, the data is converted from the source codec into PCM and then into the target codec. PCM can be considered the fundamental representation of digital time series data (EG. audio, financial ticker prices).
An alternative to PCM, with its sample rate and bit depth measurements, is the notion of storing the curve as a series of one bit samples. Instead of the ADC generating a sample of the absolute location of the height of the input analog curve when using the PCM approach, with this alternative one bit sample approach the ADC generates the relative position of the input analog curve. This one bit approach is memory efficient when performing very high sample rates.
