What is the principle of binary audio coding?

Sound is vibrations of a medium, such as air, that propagate in waves. They act on the ear membrane and are perceived by humans. This is a purely analog signal. For computer representation, the sound converted by a microphone into an analog voltage must be represented in discrete form and binary code. This transformation is called sampling.

To perform sampling of a continuous sound signal, it is necessary to store the sound intensity in binary numerical form at fixed time intervals. During playback, discrete sound is converted into an analog signal and fed to the speakers, that is, the reverse process is formed. To obtain high quality sound, it is necessary to carry out measurements as often as possible and to allocate as many binary digits as possible to record the sound volume.
The principle of binary audio coding is that an analog audio signal is converted into a discrete sequence of signal levels – the so-called depth of sound. So, for a 16-bit sound card, the number of volume levels will be two to the sixteenth power, or N = 65538.

The sound quality also directly depends on the number of loudness measurements per unit of time. The number of sound volume counts per second is called the sampling rate. The higher the sampling rate and the depth of the sound, the higher the quality of the audio encoding. It should be remembered that as the sound quality increases, the volume of the sound file increases significantly.
So, the quality of binary audio coding is determined by two parameters: sound depth and sampling frequency.