Sound is an analogue signal — a continuous wave that varies in amplitude and frequency. Computers can only store digital (binary) data, so analogue sound must be converted to digital through a process called sampling.
Sampling is the process of measuring the amplitude (height) of a sound wave at regular time intervals. Each measurement is called a sample. Each sample is stored as a binary number.
| Factor | Higher value | Lower value |
|---|---|---|
| Sample rate | Better quality, more accurate sound reproduction, larger file | Lower quality, fewer sound details captured, smaller file |
| Bit depth | More amplitude values possible, less distortion, larger file | Less accurate, more distortion (quantisation noise), smaller file |
Audio file size (bits) = sample rate × bit depth × duration (seconds)
For stereo audio (2 channels): multiply by 2
Example: 30 seconds of mono audio at 44 100 Hz sample rate, 16-bit depth:
An ADC (Analogue-to-Digital Converter) converts analogue sound signals into digital binary data. A DAC (Digital-to-Analogue Converter) converts digital data back into analogue sound for playback through speakers.
5 questions · 11 marks
| Term | Definition |
|---|
10 minutes · 9 marks