Chapter Five: Digital Audio

### 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.