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Simple about what is **sample rate** and **Nyquist–Shannon sampling theorem**

Sound is air pressure oscillation.

The oscillation is fast changing of air pressure in time.

Air oscillations become to microphone and changing of air pressure transform to changing of electrical voltage.

Analog-digital converter make measurements of voltage (samples) at each instant.

The measurements become to digital audio recording device.

**Sample rate** is number samples per second.

More sample rate - more samples per second.

**Complex signal always consists some simple sine signals.**

Sine signal is periodic. Changing of voltage (values 0 - 1 - 0 - -1 or UP - GROUND - DOWN) repeat via time period.

As swing of the pendulum: center - left - center - right, center - left - center - right.

Sine voltage: 0 - 1 - 0 - -1, 0 - 1 - 0 - -1.

**Frequency of sine** is number repeating (0 - 1 - 0 - -1) per second.

More sine frequency - more high sound.

What is Nyquist–Shannon sampling theorem?

We have period between the samples.

Designate this period as **sampling period**.

Imagine sine with the one repeating during the two sampling periods.

**Nyquist–Shannon sampling theorem** say:

It most high frequency (high sound) which be able to converted to digital form.

At first sight, digital signal like to "stairs". But, really, it is not so.

"Stairs" exists after conversion digital values to electrical voltage.

But next stage is analog filtering.

The filtering smooth stairs (interpolate). It is like to connecting voltage values by steel ruler.

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