Sampling of a Analog Signal

A Sampling of analog signal is converting a frequency domain signal into time domain signal.

You must have  read about the Shannon's theorem during the graduation.

Shannon’s Sampling Theorum ->
A band limited continuous-time signal, with highest frequency BHz can be uniquely recovered from its samples provided that the sampling rate Fs is greater than 2B samples per second.

Hope you remember.   "Sampling of Analog Signal "

First understand the basic terms.

Sample :

A sample is a value or set of values at a point in time and/or space.

Under-sampling : When a bandpass signal is sampled slower than its Nyquist rate, the samples are indistinguishable from samples of a low-frequency alias of the high-frequency signal. That is often done purposefully in such a way that the lowest-frequency alias satisfies the Nyquist criterion, because the bandpass signal is still uniquely represented and recoverable. Such undersampling is also known as bandpass sampling, harmonic sampling, IF sampling, and direct IF to digital conversion.


Over-sampling : Oversampling is used in most modern analog-to-digital converters to reduce the distortion introduced by practical digital-to-analog converters, such as a zero-order hold instead of idealizations like the Whittaker–Shannon interpolation formula

A good example to understand is PCIe , where data comes through the differential channel which is in analog form then it goes through ADC and sampler where data will convert into digital form and processed it further. 

Eye diagram in PCIe protocol is related to this. It says , if eye is wide open , error rate will be less during data sample.