
Sampling frequency (audio)

Time sampling is a process in which, during encoding of a continuous audio signal, the sound wave is divided into small separate time sections, and a certain amplitude value is set for each section. The greater the amplitude of the signal, the louder the sound.
Sound depth (encoding depth): the number of bits per sound encoding.
Volume levels (signal levels): Sound can have different volume levels. The number of different loudness levels is calculated by the formula N = 2 I where I is the depth of the sound.
Sampling rate: the number of measurements of the input signal level per unit of time (for 1 second). The higher the sampling rate, the more accurate the binary encoding procedure will be. Frequency is measured in Hertz (Hz). 1 measurement in 1 second -1 Hz.
1000 measurements in 1 second 1 kHz. Let the sample rate of the letter D. One of three frequencies is selected for encoding: 44.1 KHz, 22.05 KHz, 11.025 KHz.
The range of frequencies a person hears is believed to be 20 Hz to 20 kHz.
The quality of the binary encoding is a value that is determined by the encoding depth and the sample rate.
Audio adapter (sound card) – A device that converts electrical vibrations from an audio frequency to a numeric binary code when inputting sound and vice versa (from a numerical code to electrical vibrations) when playing sound.
Audio adapter characteristics: sampling rate and recording capacity).
The register size is the number of bits in the audio adapter register. The higher the capacity of the digit, the smaller the error of each individual conversion of the value of electric current into a number and vice versa. If the bit width is I, then by measuring the input signal, 2 I = N different values can be obtained.
The size of a digital mono audio file (A) is measured by the formula:
A = D * T * I / 8, where D is the sampling frequency (Hz), T is the resonance time or the recording of the sound, I register bit (resolution). This formula measures the size in bytes.
The size of a digital stereo audio file (A) is measured by the formula:
A = 2 * D * T * I / 8, the signal is recorded for two speakers, since the left and right sound channels are encoded separately.



