RIZAL TECHNOLOGICAL UNIVERSITY

Boni Avenue, Mandaluyong City, Philippines

SAMPLING AND RECONSTRUCTIONS OF

SIGNALS
EXPERIMENT NO.1

COMM02
Prof. Jenny Aruta

CEIT-29-501E

MEMBERS:

Aquilizan, Julianna Clarice A.

Llaguno, Karl Ian T.
Labadia, Dexter N.
Deus, Keanno P.
Gardiola, Daniel S.
Experiment 1 – Sampling and reconstruction

Preliminary Discussion

Digital transmission is fast replacing analog in commercial communications applications. There

are several reasons for this including the ability of digital signals and systems to resist interference
caused by electrical noise. Many digital transmission systems have been devised and several are
considered in later experiments. Conversion from analog signal to digital signal is the key. Figure 1a
below shows a pure sinewave for the message. Beneath the message is the digital sampling signal used
to tell the sampling circuit when to measure the message. Beneath that is the result of “naturally”
sampling the message at the rate set by the sampling signal. This type of sampling is “natural” because,
during the time that the analog signal is measured, any change in its voltage is measured too. For some
digital systems, a changing sample is unacceptable. Figure 1b shows an alternative system where the
sample’s size is fixed at the instant that the signal measured. This is known as Flat-top scheme (and is
also referred to as pulse amplitude modulation).

Regardless of the sampling method used, by definition it captures only pieces of the message.
So, how can the sampled signal be used to recover the whole message? This question can be answered
by considering the mathematical model that defines the sampled signal:

Sampled message = the sampling signal × the message

 As you can see, sampling is actually the multiplication of the message with the sampling signal.
And, as the sampling signal is a digital signal which is actually made up of a DC voltage and many
sinewaves (the fundamental and its harmonics) the equation can be rewritten as:

Sampled message = (DC + fundamental + harmonics) × message

When the message is a simple sinewave (like in Figure 1) the equation’s solution (which
necessarily involves some trigonometry that is not shown here) tells us that the sampled signal consists
of:

 A sinewave at the same frequency as the message

 A pair of sinewaves that are the sum and difference of the fundamental and message

frequencies

 Many other pairs of sinewaves that are the sum and difference of the sampling signals’

harmonics and the message

This ends up being a lot of sinewaves but one of them has the same frequency as the message.
So, to recover the message, all that need be done is to pass the sampled signal through a low-pass filter.
As its name implies, this type of filter lets lower frequency signals through but rejects higher frequency
signals. That said, for this to work correctly, there’s a small catch which is discussed in Part C of the
experiment.

Equipment

 Emona Telecoms-Trainer 101 (plus power-pack)

 Dual channel 20MHz oscilloscope

 two Emona Telecoms-Trainer 101 oscilloscope leads

 assorted Emona Telecoms-Trainer 101 patch leads

Part A – Sampling a Simple message

The Emona Telecoms-Trainer 101 has a Dual Analog Switch module that has been designed for
sampling. This part of the experiment lets you use the module to sample a simple message using two
techniques.

Procedure

1. Gather a set of the equipment listed above.

2. Connect the set-up shown in Figure 2 below.

The set-up in Figure 2 can be represented by the block diagram in Figure 3 below. It uses an
electronically controlled switch to connect the message signal (the 2kHz SINE output from the Master
Signals module) to the output. The switch is opened and closed by the 8kHz DIGITAL output of the
Master Signals module

3. Set up the scope per the

instructions in Experiment 1.

Ensure that:

 the Trigger Source control is

set to the CH1 (or INT) position.

 the Mode control is set to the

CH1 position.

4. Adjust the scope’s Timebase

control to view two or so cycles of the
Master Signals module’s 2kHz SINE output.

5. Set the scope’s Mode control to the DUAL position to view the sampled message out of the Dual
Analog Switch module as well as the message.

6. Set the scope’s Vertical Attenuation controls to the 1V/div position.

7. Draw the input and output waveforms to scale in the space provided.
Question 1 What type of sampling is this an example of?

 Natural

 Flat-top

Question 2 What two features of the sampled signal confirm this?

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8. Modify the set-up as shown in Figure 4 below

The set-up in Figure 4 can be represented by the block diagram in Figure 5 below. The electronically
controlled switch in the original set-up has been substituted for a sample-and-hold circuit. However, the
message and sampling signals remain the same (that is, a 2kHz sinewave and an 8kHz pulse train).

9. Draw the new sampled message to scale in the space that you left on the graph paper.
Question 3 What two features of the sampled signal confirm that the set-up models the Flat-top
scheme?

Question 4 Comparing Figure 3 and Figure 5, what is the difference between their outputs? Explain
why this is so.

_____________________________________________________________________________________

Part B – Reconstructing a Sampled Message

Recall that the sampled message is made up of many sinewaves. Importantly, for every
sinewave in the message, there’s a sinewave in the sampled message with the same frequency. So
“reconstructing” the original message involves passing the sampled message signal through a low-pass
filter. This lets the sinewave (or sinewaves) with the same frequency as the message through while
rejecting the other sinewaves. The next part of the experiment lets you do this.

10. Return the scope’s Timebase control to the 0.1ms/div position.

11. Locate the Tuneable Low-pass Filter module and set its Gain control to about the middle of its travel.

12. Turn the Tuneable Low-pass Filter module’s Cut-off Frequency Adjust control fully anticlockwise.

13. Disconnect the plugs to the Speech module’s output.

14. Modify the set-up as shown in Figure 7 below

 The Tuneable Low-pass Filter module is used to recover the message. The filter is said to be
“tuneable” because the point at which frequencies are rejected (called the cut-off frequency) is
adjustable. At this point there should be nothing out of the Tuneable Low-pass Filter module. This is
because it has been set to reject almost all frequencies, even the message. However, the cut-off
frequency can be increased by turning the module’s Cut-off Frequency Adjust control clockwise.

15. Slowly turn the Tuneable Low-pass Filter module’s Cut-off Frequency control clockwise and stop the
moment the message signal has been reconstructed

16. Draw the input and output waveform for Figure 7.

Part C – Aliasing

At present, the filter is only letting the message signal through to the output. It is comfortably
rejecting all of the other sinewaves (called aliases) that make up the sampled message. This is only
possible because the frequency of these other sinewaves is high enough. But, this isn’t an accident.
Their frequency is determined by the sampling rate (that is, the sampling signal’s frequency).

To explain, recall that the sampled message consists of the following:

 A sinewave at the same frequency as the message

  A pair of sinewaves that are the sum and difference of the fundamental and message
frequencies

 Many other pairs of sinewaves that are the sum and difference of the sampling signals’
harmonics and the message. In your set-up, a 2kHz sinewave message is sampled using an 8kHz
sampling signal.

That being the case, the lowest frequency components of the sampled message are:

 2kHz

 6kHz

 10kHz

Now, suppose the frequency of the sampling signal is lowered. You’d still get the message but
the frequency of the aliases would go down as well. So for example, if the sampling signal is 7kHz, the
lowest frequency components of the sampled message become 2kHz, 5kHz and 9kHz. Clearly, if the
sampling signal’s frequency is low enough, one or more of the lower frequency aliases can pass through
the filter along with the reconstructed message. Obviously, this would distort the reconstructed
message which is a problem known as aliasing.

To avoid aliasing, the sampling signal’s theoretical minimum frequency is twice the message
frequency (or twice the highest frequency in the message if it contains more than one sinewave and is a
baseband signal). This principle is known as the Nyquist Sample Rate. So for this set-up, the minimum
sampling rate is 4kHz and the lowest frequency components would be 2kHz, 2kHz and 6kHz.

That said, filters aren’t perfect. Their rejection of frequencies beyond the cut-off is gradual
rather than instantaneous. So, in practice the sampling signal’s frequency needs to be a little higher than
the Nyquist Sample Rate.

The next part of the experiment lets you vary the sampling signal’s frequency to observe
aliasing.

17. Locate the VCO module and set its Frequency Adjust control fully clockwise.

18. Set the VCO module’s Range control to the LO position.

19. Modify the set-up as shown in Figure 9 below.

 This set-up can be represented by the block diagram in Figure 10 below. Notice that the
sampling signal is now provided by the VCO module which has a manually adjustable frequency.

At this point, the sampling of the message and its reconstruction should be working normally.

20. Draw the input and output

21. Slowly reduce the frequency of the VCO module’s output (by turning its Frequency Adjust control
anti-clockwise) while watching the reconstructed message signal.

22. Draw the input and output

Question 5 What’s the name of the distortion that appears when the VCO module’s Frequency Adjust
control is turned far enough?

- As a result of the oscilloscope taking photos of the signal too slowly, aliasing distortion causes
quick parts of the signal to seem jumbled and produce odd patterns on the screen.
Question 6 Given the message is a 2kHz sinewave, what’s the theoretical minimum frequency for the
sampling signal?

- The sampling signal's minimum theoretical frequency is 4 kHz.

Question 7 Can you still use the Master Signal to provide sampling pulse in Figure 9 or 10, instead of a
VCO and get the same result? Explain.

- Yes, even if you use master signal the resulting sampling pulse will also be distorted.

_____________________________________________________________________________________

23. Turn the VCO module’s Frequency Adjust control clockwise and stop the moment the reconstructed
message is no-longer distorted.

24. Connect the scope’s Channel 1 input to the VCO module’s DIGITAL output.

25. Set the scope’s Mode control to the CH1 position.

26. Adjust the scope’s Timebase control to view two or so cycles of the VCO module’s DIGITAL output.

27. Measure the signal’s period and record this in Table 1 on the next page.

28. Use the period to calculate and record the signal’s frequency.

Question 8 Why is the actual minimum sampling frequency higher than the theoretical minimum that
you calculated for Question 6?

- The actual minimum sampling frequency is higher than the theoretical minimum because
filters cannot roll off infinitely fast.

_____________________________________________________________________________________
QUESTIONS:

1. What type of sampling is this an example of?

 Natural
 Flat-top
2. What two features of the sampled signal confirm this?
- The two features of the sampled signal that confirmed this is the characteristics of the
Natural Signal that it consists of, and the resemblance of the signal in the figure 1a where it
says that the analog signal is measured along with the voltage.
3. What two features of the sampled signal confirm that the set-up models the Flat-top scheme?
- As you can see in the figure above, which is the figure 1B, it has the same representation of
signal waves as shown in the equipment. The feature of the signal wave can be recognized
or described as a stair but side by side.
4. Comparing Figure 3 and Figure 5, what is the difference between their outputs? Explain why
this is so.
- Upon comparing the signals from figure 3 and 5, we can see that in figure 3 for their output,
a natural signal is formed while in figure 5 a flat-top signal is formed. The reason behind this
difference is that flat-top signals has flattened amplitude peaks while natural signal is the
modified version.

Conclusion:

In conclusion, the experiment successfully demonstrates how the sampling and reconstruction of
the message signal work in digital transmissions. It was proven that different sampling methods affect the
message signal in the output. In the natural sampling method, it was shown that the top of each pulse in
the sampled signal preserved the shape of the analog signal during the pulse interval, while in the flat-top
method, the amplitude of each pulse remained the same. The experiment also proved how the sampled
signal was reconstructed back to its original form when it was introduced to the low-pass filter. To which
it was shown that the low-pass filter lets the lower frequency of the sampled signal pass through while
attenuating the higher frequency, recovering the original signal in the process. Lastly, in aliasing, the
experiment proved its anomalies when reconstructing the message signal. It was shown that when the
VCO produced a frequency that exceeded half of the sampling rate, the frequency appeared low in the
digital representation, making the intended output distorted.