Sampling and Reconstruction

Any continuous or an analog signal can be represented in the digital version in the form of
samples. These samples are also called 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 equidistant with a period of ‘Ts’. This sampling signal
frequency should be more than twice that of the input analog signal frequency. If this
condition satisfies, the analog signal can be perfectly represented in discrete form, otherwise,
the 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 do well in the reconstruction process for recovering the original signal.
The output sample signal is represented by the samples. These samples are maintained with a
gap and these gaps are termed as sample period or sampling interval (Ts). The reciprocal of
the sampling period is known as ‘sampling frequency’ or ‘sampling rate’. The number of
samples represented in the sampled signal is indicated by this sampling rate.
Sampling frequency Fs=1/Ts

The Sampling theorem states that “continuous 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
Thus, If the sampling frequency (Fs) equals twice the input signal frequency (Fm), then such
a condition is called the ‘Nyquist Criteria’ for sampling.
When the sampling frequency equals twice the input signal frequency it is known as ‘Nyquist
rate’.
Fs = 2Fm
If the sampling frequency (Fs) is less than twice the input signal frequency (Fm), such criteria
is called an ‘Aliasing effect’.
Fs < 2Fm
Possibility of sampled frequency spectrum with different conditions are given by the following
diagrams:

Aliasing Effect

It is the effect in which overlapping of a frequency component takes place at the frequency
higher than Nyquist rate i.e. fs > 2fm. Loss of signal may occur due to aliasing effect. We can
say that aliasing is the phenomenon in which a high frequency component in the frequency
spectrum of a signal takes the identity of a lower frequency component in the same spectrum
of the sampled signal.
Aliasing is generally avoided by applying low pass filters or anti-aliasing filters (AAF) to the
input signal before sampling and when converting a signal from a higher to a lower sampling
rate.
Signal Reconstruction from Sinc Interpolation.
Bandlimited interpolation of discrete-time signals is a basic tool having extensive application

in digital signal processing. In general, the problem is to correctly compute signal values at

arbitrary continuous times from a set of discrete-time samples of the signal amplitude. In

other words, we must be able to interpolate the signal between samples. Since the original

signal is always assumed to be bandlimited to half the sampling rate, (otherwise aliasing

distortion would occur upon sampling), Shannon's sampling theorem tells us the signal can be

exactly and uniquely reconstructed for all time from its samples by bandlimited interpolation.

Consequently, Shannon's sampling theorem gives us that x(t) can be uniquely reconstructed

from the samples x(nT)via

(5.13)

sin ⁡(π f s t )
where h s ( t ) ≜
π f st

sin ⁡( πt)
A plot of the sinc function sinc to the left and right of the origin t=0 is shown in
(πt)

Note that peak is at amplitude , and zero-crossings occur at all nonzero integers. The name

sinc function derives from its classical name as the sine cardinal (or cardinal sine) function.

The sinc function plotted for seven zero-crossings to the left and right.

If `` '' denotes the convolution operation for digital signals, then the summation in Eq.

(4.13) can be written as (x*hs) .

Equation can be interpreted as a superpositon of shifted and scaled sinc functions hs. A sinc

function instance is translated to each signal sample and scaled by that sample, and the

instances are all added together. Note that zero-crossings of sinc(z) occur at all integers

except z=0. That means at time t=nT, (i.e., on a sample instant), the only contribution to the

sum is the single sample x(nT). All other samples contribute sinc functions which have a

zero-crossing at time t=nT. Thus, the interpolation goes precisely through the existing

samples, as it should.

A plot indicating how sinc functions sum together to reconstruct bandlimited signals is shown

in Figure. The figure shows a superposition of five sinc functions, each at unit amplitude, and

displaced by one-sample intervals. These sinc functions would be used to reconstruct the

bandlimited interpolation of the discrete-time signal x={….0,1,1,1,1,1,0…..}.Note that at

each sampling instant , the solid line passes exactly through the tip of the sinc

function for that sample; this is just a restatement of the fact that the interpolation passes

through the existing samples. Since the nonzero samples of the digital signal are all , we

might expect the interpolated signal to be very close to over the nonzero interval; however,

this is far from being the case. The deviation from unity between samples can be thought of

as ``overshoot'' or ``ringing'' of the lowpass filter which cuts off at half the sampling rate, or it

can be considered a ``Gibbs phenomenon'' associated with band limiting.

Figure 4.22: Bandlimited reconstruction of the signal from its samples

 .

The dots show the signal samples, the dashed lines show the component sinc functions, and

the solid line shows the unique bandlimited reconstruction from the samples obtained by

summing the component sinc functions.