Computers have the ability to change analogue sounds and converts them into digital sounds because computers can only interpret binary information (0’s and 1’s). That is why input devices such as microphones are used to provide the computer with binary information about how the analogue sounds so that it can be recreated by the computer digitally.
Sound is measured via waves and those wave need to be converted into binary patterns; this is done through a method called sampling. It consists of recording the points on the wave at regular, tiny intervals so that the computer can map out the wave electronically. In order to be able to recreate it fully, metadata is once again required to do so and it consists of the audio codec (a function that help map the wave points) and the sample rate (the waves sampled per second - in Hertz [Hz])
The sample rate is often described by the sample interval which refers to the time between each sample that is taken. A higher sample rate results in a lower sample interval because there is less time between each sample as more are being recorded. When sound is sampled at a low rate:
When sound is sampled at a high rate:
The bit rate is the amount of bits per second that can be transmitted along a digital network. A standard MP3 track stores sound at 128 kbits per second whereas an audio CD has the capability of storing up to 1411.2 kbits per second .
When sound is sampled at a high bit rate:
When sound is sampled at a low bit rate: