UNIT 5

What is the Sampling Theorem?

A continuous signal or an analog signal can be represented in the digital version in the form
of samples. Here, these samples are also called as discrete points. In sampling theorem, the
input signal is in an analog form of signal and the second input signal is a sampling signal,
which is a pulse train signal and each pulse is equidistance with a period of “Ts”. This
sampling signal frequency should be more than twice of the input analog signal frequency. If
this condition satisfies, analog signal perfectly represented in discrete form else analog signal
may be losing its amplitude values for certain time intervals. How many times the sampling
frequency is more than the input analog signal frequency, in the same way, the sampled
signal is going to be a perfect discrete form of signal. And these types of discrete signals are
well performed in the reconstruction process for recovering the original signal.

sampling Theorem Definition

The sampling theorem can be defined as the conversion of an analog signal into a discrete
form by taking the sampling frequency as twice the input analog signal frequency. Input
signal frequency denoted by Fm and sampling signal frequency denoted by Fs.

The output sample signal is represented by the samples. These samples are maintained with a
gap, these gaps are termed as sample period or sampling interval (Ts). And the reciprocal of
the sampling period is known as “sampling frequency” or “sampling rate”. The number of
samples is represented in the sampled signal is indicated by the sampling rate.

Sampling Theorem Statement

Sampling theorem states that “continues form of a time-variant signal can be represented in
the discrete form of a signal with help of samples and the sampled (discrete) signal can be
recovered to original form when the sampling signal frequency Fs having the greater
frequency value than or equal to the input signal frequency Fm.

Fs ≥ 2Fm
If the sampling frequency (Fs) equals twice the input signal frequency (Fm), then such a
condition is called the Nyquist Criteria for sampling. When sampling frequency equals twice
the input signal frequency is known as “Nyquist rate”.

Fs=2Fm
If the sampling frequency (Fs) is less than twice the input signal frequency, such criteria
called an Aliasing effect.

Fs<2Fm
So, there are three conditions that are possible from the sampling frequency criteria. They are
sampling, Nyquist and aliasing states. Now we will see the Nyquist sampling theorem.
Nyquist Sampling Theorem
In the sampling process, while converting the analog signal to a discrete version, the chosen
sampling signal is the most important factor. And what are the reasons to get distortions in
the sampling output while conversion of analog to discrete? These types of questions can be
answered by the “Nyquist sampling theorem”.

Nyquist sampling theorem states that the sampling signal frequency should be double the
input signal’s highest frequency component to get distortion less output signal. As per the
scientist’s name, Harry Nyquist this is named as Nyquist sampling theorem.

There are few applications of sampling theorem are listed below. They are

 To maintain sound quality in music recordings.
 Sampling process applicable in the conversion of analog to discrete form.
 Speech recognition systems and pattern recognition systems.
 Modulation and demodulation systems
 In sensor data evaluation systems
 Radar and radio navigation system sampling is applicable.
 Digital watermarking and biometric identification systems, surveillance systems.

Spectra of sampled signal

Signal Reconstruction:
The process of reconstructing a continuous time signal x (t) from its samples is known
as interpolation. 
Ideal Interpolator
The ideal interpolator is a (sinc) function

Aliasing and its effects:

 
In reconstructing a signal from its samples, there is another practical difficulty. The sampling
theorem was proved on the assumption that the signal x(t) is bandlimited. All practical signals
are time limited, i.e., they are of finite duration. As a signal cannot be timelimited and
bandlimited simultaneously. Thus, if a signal is timelimited, it cannot be bandlimited and vice
versa (but it can be simultaneously non timelimited and non bandlimited). Clearly it can be
said that all practical signals which are necessaily timelimited, are non bandlimited, they have
infinite bandwidth and the spectrum X'(f) consists of overlapping cycles of X(f) repeating
every fs Hz (sampling frequency). Because of infinite bandwidth, the spectral overlap will
always be present regardless of what ever may be the sampling rate chosen. Because of the
overlapping tails, X'(f) has not complete information about X(f) and it is not possible, even
theoretically to recover x(t) from the sampled signal x'(t). If the sampled signal is passed
through an ideal lowpass filter, the output is not X(f) but a version of X(f) distorted as a result
of some causes:
 
 The loss of the tail of X(f) beyond  |f | > fs/2  Hz.
 
 The reappearance of this tail inverted or folded onto the spectrum. The spectra cross
at frequency fs/2 = 1/2T   Hz. This frequency is called the folding frequency. The
spectrum folds onto itself at the folding frequency. For instance, a component of
frequency (fs/2)+ fx shows up as or act like a component of lower frequency (fs/2)- fx  in
the reconstructed signal. Thus the components of frequencies above fs/2 reappear as
components of frequencies below fs/2. This tail inversion is known as spectral
folding or aliasing which is shown in Fig. 5. In this process of aliasing not only we
are losing all the components of frequencies above fs/2 Hz, but these very components
reappear as lower frequency components. This reappearance destroys the integrity of
the lower frequency components.