Sample frequency

Overview

In order to digitize analog audio, most contemporary systems use a system referred to as "sampling" to repeatedly measure the voltage of the analog audio waveform at a regular time interval. Each voltage measurement results in a binary number of a given wordlength. The series of binary “words” are typically stored consecutively in a file for later reconstruction of the analog voltage waveform by a digital to analog converter. The sample frequency is the rate at which the samples are generated and is measured in Hertz (cycles per second). The term sample rate is used interchangeably with sample frequency.

Basics

Virtually all contemporary analog audio equipment operates on the principle of an analog voltage waveform being analogous to the the original sound's air pressure "waveform." Typically; the original sound is translated from the pressure variation to electrical variations by a microphone ( a type of transducer). The resulting voltage waveform can be transmitted on wires to an amplifier and a power amplifier; then translated back into sound pressure variations by a speaker.

One important consideration is how this analog waveform can be stored for later reproduction or transmission. All analog storage and transmission schemes are prone to loss of signal quality; with storage being particularly problematic. As the technology became available, digital audio systems were developed to address these issues.