Introduction to Computer Music: Volume One

3. Samples and sampling

Sounds from the real world can be recorded and digitized using an analog-to-digital converter (ADC). As in the diagram below, the circuit takes a sample of the instantaneous amplitude (not frequency) of the analog waveform. Alternatively, digital synthesis software can also create samples by modeling and sampling mathematical functions or other forms of calculation. A sample in either case is defined as a measurement of the instantaneous amplitude of a real or artificial signal. Frequencies will be recreated later by playing back the sequential sample amplitudes at a specified rate. It is important to remember that frequency, phase, waveshape, etc. are not recorded in each discrete sample measurement, but will be reconstructed during the playback of the stored sequential amplitudes.

Samples are taken at a regular time interval. The rate of sample measurement is called the sampling rate (or sampling frequency). The sampling rate is responsible for the frequency response of the digitized sound.

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