In our daily lives, we experience sound as a continuous flow of vibrations. However, computers are fundamentally different from the human ear; they cannot process the continuous, flowing nature of the physical world. To store music, voice recordings, or any other audio in a computer, we must undergo a process of translation from the analogue world to the digital world.
Understanding Sound Waves
To understand digital sound, we must first understand what sound is in its natural state. Sound is perceived by our ears as waves, which are actually vibrations traveling through the air. These waves are considered analogue because they vary continuously over time.
Two primary characteristics define these waves:
- Amplitude: This refers to the height of the sound wave. In practical terms, amplitude represents the volume of the sound; the higher the amplitude, the louder the sound.
- Frequency: This refers to the number of waves that occur within a specific period of time. Frequency determines the pitch of the sound. High-frequency waves (where waves are closer together) result in a high pitch, while low-frequency waves (where waves are further apart) result in a low pitch.
From Analogue to Digital: The ADC Process
Because computers cannot work with analogue data, the signal must be converted into a format they can understand: binary (a series of 0s and 1s). This conversion is performed by a piece of hardware known as an Analogue to Digital Converter (ADC).
The ADC works through a process called sampling. Instead of trying to record the entire continuous wave, the converter measures the amplitude of the sound wave at specific, discrete moments in time. Each of these individual measurements is called a sample. Once a sample is measured, its value is converted into a binary number so it can be stored in the computer's memory.
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