Course Number: EEE 310 Group Number: 04

Experiment No: 04

Name of the Experiment:

SAMPLING AND SIGNAL RECONSTRUCTION

Date of Performance: 05.03.16 Name: Towsif Taher

Date of Submission: 12.03.16 Department: EEE (A1)

Student ID: 1206025

Level-3 Term-2
Objective:
The objective of this experiment is to,
investigate methods used to sample a signal and to recognize the signal that
results from each method.
investigate a method used to reconstruct the intelligence from a sampled signal
and demonstrate how the sampling signal frequency and the low-pass filter
characteristic affect reconstruction.

Apparatus used in this experiment:

Power source
Oscilloscope
AF generator
Frequency counter
Pulse Modulation Trainer Board
Patch chord

Theory:
Sampling means to convert an analog signal into a corresponding
sequence of samples that are usually spaced uniformly in time. For doing this, a proper
sampling rate has to be chosen, so that the sequence of samples uniquely defines the
original analog signal.

In this experiment two types of sampling method have been applied to sample a signal,

1. Natural sampling:
It is a type of sampled signal in which the top of each sample pulse follows
the intelligence signal during the pulse width time of the sampling signal.

2. Flat-topped sampling:
It is a type of sampled signal in which the top of each sample pulse
represents a signal level of intelligence during the pulse-width time of the
sampling signal.
Sampling principle:

According to sampling principle, the intelligence can be totally reconstructed by

filtering when the sampling signal frequency ( ) is greater than twice the maximum
intelligence signal frequency( ).

= 2

is called the Nyquist rate. Now to reconstruct the original signal from the sampled
signal, sampled signal is passed through a low pass filter. The frequency response of
the low-pass filter must be capable of passing the maximum intelligence signal to
reconstruct the intelligence signal frequency while rejecting side band frequencies of
the sampled signal to reconstruct the intelligence free of distortion.

Natural sampling:

Block diagram:
Circuit diagram:

Observed wave-shapes:

Original Intelligence Signal: Sine wave of 3V p-p and 1KHz frequency

Pulse train representing the sampling signal at J9: 15KHz frequency

Natural sampling, sampled output at J8: sampled signal at 15KHz

Flat-topped sampling:

Block diagram:
Circuit diagram:

Observed wave-shapes:

Original Intelligence Signal: Sine wave of 3V p-p and 1KHz frequency

Pulse train representing the sampling signal at J9: 15KHz frequency

Output of the sampler/hold circuit at J16: sampled signal at 15KHz

Reconstruction of the Sampled Signal:

Block diagram:

Circuit diagram:
Observed wave-shapes:

Original Intelligence Signal: Sine wave of 3V p-p and 5KHz frequency

Reconstructed signal at 16 kHz sampling frequency

Reconstructed signal at 3kHz sampling frequency

Reconstructed signal at 9.5kHz sampling frequency

Reconstructed signal at 10kHz sampling frequency

Reconstructed signal at 50kHz sampling frequency

Reconstructed signal at 100kHz sampling frequency:

Observations:
Increase in sampling frequency increases the similarity of the reconstructed
signal shapes more and more towards the original intelligence signal.
Satisfying Nyquist criterion does not guarantee distortion-less reconstructed
signal.
Reconstructed signal has fractional gain (Av<1).
Cascading a second low-pass filter removes distortion in the reconstructed
signal quite a lot.

Discussion:
Reason behind distortion of reconstructed signal when Fs>3Fm:
If a signal is sampled at the Nyquist rate, the spectrum consists of repetitions of
() without any gap between successive cycles. To recover the original signal
the sampled signal should be passed through an ideal low-pass filter, but such a
filter is unrealizable. A practical solution to this problem is to sample the signal
at a rate higher than the Nyquist rate. But even in this case the filter gain is
required to be zero beyond the first cycle of (). By Paley-Wiener criterion it
is impossible to realize even this filter.

Another fundamental practical difficulty in reconstructing a signal is Aliasing.

All practical signals are time-limited. A signal cannot be time-limited and band-
limited simultaneously. If a signal is time-limited, it cannot be band-limited and
vice-versa. So for practical signals, because of infinite bandwidth, spectral
overlap is a constant feature, regardless of the sampling rate. That is why, even
if the sampled signal is passed through an ideal low-pass filter, the output will
contain some amount of distortion.
Advantage of cascading filters:

When two filters are cascaded, their responses are multiplied. Cascaded amplifier
has steeper roll-off. So, within the bandwidth cascaded filter can more accurately
filter distorted signal as in this case the probability of aliasing and overlapping
decreases. As a result we get a better signal.

Required characteristics of the low pass filter and the sampling

frequency to reconstruct the intelligent signal.

If sampler output is () , then () = () () (in time domain).

Here () is a pulse train.

So, () = +
= ( ) in time domain

2
() = +
= ( ) in frequency domain.

As a result, in frequency domain () can be expressed as

+
1 1
() = () () = ( )
2
=

So when the sampled signal is passed through an ideal low pass filter which passes
only those frequencies contained in (), the spectrum of the filter output will be
1
identical to () except for the amplitude which is multiplied by a factor .To

recover (), () is passed through a filter with frequency response of

, || <
() = {
0,

() = ( )

When this characteristics is converted to time domain then it becomes a sinc
function as shown in the figure

This filter is called an ideal reconstruction filter. Signal reconstruction is shown in

the following figure:

Now if sampling frequency is reduced then the different components in the

spectrum get closer and ultimately they overlap.

Three conditions may arise,

If = 0 components will touch each other.

 If < , components will overlap. So a completely new signal of
another frequency is obtained. In this case the original signal cannot be
recovered. This overlapping phenomena is called Aliasing.
If > > 2 , components will not overlap. Signal can be
totally recovered from its samples. This condition is known as Nyquist
Theorem or Sampling Theorem.

For the above stated reasons, we get distorted signal when the sample frequency is
less than 10kHz.