Digital signals sample rate/frequency

Sampling rate — The speed at which samples are read and converted into a voltage or a digital expression. The Nyquisfs theorem establishes that the sampling rate must be greater than twice the maximum frequency component in the signal to be acquired to avoid the aliasing of data [i]. 

This concept is one of the most difficult to quantitate. There are some relatively explicit definitions of information content for electronic communications. (For example, the Nyquist theorem defines the minimum sampling rate required in order to preserve the maximum frequency information in a periodic signal. And, the relationships between digital encoding formats and information content of a data base can be quantitated.) However, for the general problem of evaluating the results of instrumental measurements of chemical systems, the definitions for information content of data are very clear. 

Fig. 2.3.9 [Deri] Digital sampling of analog signals. The continuous signal is digitized at an adequate sampling rate, the broken signal is not. After sampling it appears at the frequency of the continuously drawn signal.

A digital signal has an upper bound on the frequencies it can represent. This is called the Nyquist frequency and is exactly half the sample rate. 

Use of normalised frequency, d) = co/Fj, avoids having to keep track of the sampling rate in digital signal processing expressions. 

Using n and normalised angular frequency sample rate. To convert to real frequency, the normalised frequency is simply multiplied by the sample rate. In general, we use the same terms for amplitude and phase for both analogue and digital signals. 

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