Sound is recorded digitally using sampling. The diagram shows an analogue sound wave being sampled at regular intervals (shown by the dots). (a) Describe ho...

(a) The process of converting an analogue sound wave into a digital file involves four key steps:

1. A microphone converts sound waves (pressure variations in the air) into a continuous analogue electrical signal whose voltage varies with the sound's amplitude. [1]
2. At regular intervals determined by the sample rate (measured in Hz), the amplitude of the analogue signal is measured. Each measurement captures the height of the wave at that instant. [1]
3. Each sampled amplitude value is rounded to the nearest available level, as determined by the bit depth. This rounding process is called quantisation. The rounded value is then encoded as a binary number. [1]
4. The sequence of binary values is stored as a digital sound file that can be played back, edited or transmitted. [1]

(b) Two factors affecting quality and file size:

1. Sample rate (the number of samples taken per second, measured in Hz): a higher sample rate captures more points of the original wave per second, producing a more accurate digital representation. However, more samples means more data, increasing the file size. [1]
2. Bit depth (the number of bits used to store each sample): a higher bit depth provides more possible amplitude levels, allowing each sample to represent the wave's amplitude more precisely. This improves quality (reducing quantisation noise) but increases file size because each sample value occupies more bits. [1]