Sound is naturally an analogue (continuous) signal — a pressure wave in air. Computers store digital (discrete) values, so sound must be sampled (digitised) at regular intervals.
A microphone converts sound waves to an electrical signal. The ADC then samples this signal at fixed time intervals, measuring the amplitude at each moment. Each measurement is stored as a binary number. On playback, a DAC (Digital-to-Analogue Converter) reconstructs the wave.
| Property | Definition | Effect of increasing |
|---|---|---|
| Sample rate | Number of samples taken per second (Hz) | Better frequency reproduction, larger file |
| Bit depth | Number of bits per sample (resolution of each measurement) | Better dynamic range/amplitude accuracy, larger file |
| Channels | Mono (1) or stereo (2) or surround | More channels = larger file |
To accurately reproduce a sound of frequency f, the sample rate must be at least 2f. CD audio uses 44,100 Hz (44.1 kHz) — capable of capturing up to 22,050 Hz (above the human hearing range of ~20 kHz). If the sample rate is too low, high-frequency sounds are lost or distorted.
Formula: File size (bits) = sample rate × bit depth × channels × duration (seconds)
Stereo audio, 44,100 Hz sample rate, 16-bit depth, 3 minutes:
| Format | Compression | Use case |
|---|---|---|
| WAV | None (lossless) | High quality, large files, professional audio |
| AIFF | None (lossless) | Apple equivalent of WAV |
| MP3 | Lossy | Music streaming, reduced file size |
| FLAC | Lossless | Archival, audiophile quality |
Higher sample rate and bit depth produce better audio quality but larger files. Compression (MP3) reduces file size by discarding inaudible frequencies (psychoacoustics) but introduces quality loss. This is the key trade-off: quality vs storage/bandwidth efficiency.
8 questions · instantly marked · AQA 7517 standard
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10 questions · 10 minutes