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In digital recording, an audio or video signal is converted into a stream of discrete numbers representing the changes over time in air pressure for audio, or chroma and luminance values for video. This number stream is saved to a storage device. To play back a digital recording, the numbers are retrieved and converted back into their original analog audio or video forms so that they can be heard or seen. In a properly matched analog-to-digital converter (ADC) and digital-to-analog converter (DAC) pair the analog signal is accurately reconstructed per the constraints of the Nyquist–Shannon sampling theorem dependent on the sampling rate and quantization error dependent on the audio or video bit depth. Because the signal is stored digitally, assuming proper error detection and correction, the recording is not degraded by copying, storage or interference. October 3, 1938: British telephone engineer Alec Harley Reeves files at the French Patent Office the first patent describing the technique known today as pulse-code modulation (PCM). On November 22, 1939, Reeves files also in the US. It was first proposed as a telephony technology. 1943: Bell Telephone Laboratories develops the first PCM-based digital scrambled speech transmission system, SIGSALY, in response to German interception of military telephone traffic during World War II. The twelve transmission points were retired after the war. June 1950: Differential pulse-code modulation (DPCM) developed by C. Chapin Cutler at Bell Labs. 1957: Max Mathews of Bell Labs recorded the first computer-generated music, a 17-second piece called "The Silver Scale" composed by his co-worker Newman Guttman. 1967: The first commercial PCM encoder (monaural) was developed by NHK's research facilities in Japan. The 30 kHz 12-bit device used a compander (similar to DBX Noise Reduction) to extend the dynamic range, and stored the signals on a video tape recorder. 1969: NHK expands the PCM encoder's capabilities to two-channel stereo and 32 kHz 13-bit resolution.

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