Experiment 4 - Sampling and PCM

Department of Electrical Engineering & Electronics

September 2022, Ver. 3.5

Important: Marking of all coursework is anonymous. Do not include your name,

student ID number, lab number, email or any other personal information anywhere
in the report. A penalty will be applied to submissions that do not meet this
requirement.

Experiment specifications
Module(s) ELEC224 / ELEC273
Experiment code 4
Semester 1
Level 2
Lab location Electronics lab, third floor, check the lab timetable
Work Individual
Timetabled time 7 hrs
Subject(s) of relevance Sampling, Signals and Systems
Assessment method This workbook submitted to CANVAS
Submission deadline One week after the lab day

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Instructions:
ˆ Read this script carefully before attempting the experiment.
ˆ The Pre-Lab Questions should be answered before the lab day. They are
available on CANVAS (Online) and worth 10%.
ˆ The script questions should be answered while carrying out the experiment.
ˆ At each part, you should show your connections and output to one of the
demonstrators and get approval.
ˆ This script should be completed with the graphs and answers of the questions.
ˆ A report with the script should be submitted to CANVAS one week after the
lab experiment. For the report, you can use the template for this experiment,
providing your answers along with drawings or photos of the lab equipment;
or bring your own hard copy of the workbook, complete it during the lab
and submit a scanned/photographed version in Canvas. Whatever option is
chosen, the submission needs to be clear and legible or we will not be able to
mark it.

1 Objectives
The objectives of this experiment are:
ˆ to study and practically test some important telecommunications concepts like sampling,
PCM encoding and decoding and bandwidth limitations.
ˆ to get hands-on experience on using the Emona Telecoms-Trainer 101 kit.

2 Apparatus
ˆ Emona Telecoms-Trainer 101 kit.
ˆ Oscilloscope TDS 210.
ˆ Different wires and BNC cables.

3 Introduction
In modern telecommunications, digital transmission has continually increased since its intro-
duction in 1962. This is due, in large part, to the fact that most of the communication providers
require a high degree of accuracy in the information they are transmitting through their net-
works. With digital transmission (as compared to analogue), systems are better switching
interfaces, easier to multiplex and producing clearer signals. A digital signal is depicted as
discontinuous discretely variable on/off pulses, as opposed to an analogue signal which is con-
tinuously variable.

Each pulse is known as a bit. A bit is the most common digital signal in the telecommunication
industry. The number of bits transmitted per second is the bit rate of the signal. To convert
analogue signals to digital, a coding system called Pulse Code Modulation (PCM) is used. This
process requires other pre- and post-processing steps, among which sampling is needed.

This experiment introduces some of these concepts, offering hands-on experience and practical
tests.

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4 Part A: Introduction to the Emona Telecoms-Trainer 101 (15 Marks)
The Emona Telecoms-Trainer 101 is used to study the fundamentals of telecommunications
principles at the block diagram level. It contains many telecommunications’ functional build-
ing units that can be patched to implement a wide variety of systems like basic modulators
(Figure 1), encoders and other important sub-systems associated with telecommunications the-
ory, using only one piece of lab hardware and without worrying about how the internal circuit
works. Examples of common blocks include adder, filter and phase shifter.

The kit offers the required components to practically implement a wide range of the basic
analogue communications concepts like AM, FM, DSB, SSB, PM, PAM, TDM, PWM, PLL,
QAM and SNR, and digital communications concepts like PCM, PCM-TDM, ASK, BPSK,
FSK, GFSK, DPSK, QPSK, spread spectrum, line coding and noise generation.

Figure 1: Example of a system built from component blocks.

4.1 The master signals and the buffer modules

ˆ Connect the master signal from the kit to the oscilloscope’s Ch-1. Sketch Ch-1 output in
the grid below (Figure 2), recording amplitude, period and frequency. [1 mark]

Figure 2: Master signal.

ˆ Locate the Buffer module and connect the setup in Figure 3. Calculate the buffer’s gain
mentioning the uncertainty in this measurement. [1 mark]

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 Figure 3: Master signal and buffer block.

Buffer module’s gain =

ˆ Connect the master signal 100 kHz SIN to Ch-1 and 100 kHz COS to Ch-2, as in Figure 4.
Sketch both signals in the same grid of the figure, recording amplitude, period, frequency
and phase shift, making note of the uncertainty in each of these measurements. [1 mark]

– Amplitude:
– Period:
– Frequency:
– Phase shift:

Figure 4: SIN and COS master signals.

Question 1
ˆ What do you expect the theoretical phase shift between SIN and COS to be, and why
your measurement value does not match it? [2 marks]

4.2 The Speech module

ˆ Locate the speech module on the kit that contains a microphone (Figure 5). Connect its
output to Ch-1.

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 Figure 5: Speech module.

ˆ Talk while observing the signal on the oscilloscope screen.

Question 2 [1 mark]
ˆ Can you sketch the signal? . . . . . . . . . . . . . . . . Why? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . ..

4.3 The Adder module

The adder module has two inputs: A and B. Each input has its own variable gain key, labelled
with G for input A and g for input B.
ˆ Locate the adder module and connect the setup shown in Figure 6. Find out the maximum
gain for input A, and then connect the setup in Figure 7. Find out the maximum gain for
input B, noting the uncertainty in this measurement..

Max Gain for input A = . . . . . . . . . . . . . . . . . . . . . [1 mark]

Figure 6: Adder example.

Max Gain for input B = . . . . . . . . . . . . . . . . . . . . . [1 mark]

Figure 7: Adder example.

ˆ Connect the setup in Figure 8 and adjust the gains of input A to 1 and input B to 0.5.

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 Figure 8: Adder example.

Question 3 [1 mark]
ˆ Write down the equation that represent the output to Ch-2.

Question 4 [2 marks]
ˆ What is the maximum amplitude of the resulting signal (the signal of the summation)?

Question 5 [2 marks]
ˆ Is it right to say:
Max amp of the summation output signal (A+B) = max amp of A + max amp of B? Why?

Question 6 [2 marks]
ˆ What will happen if there is phase shift between the two added signals?

5 Part B: Sampling and reconstruction (25 Marks)

Objectives:

In this part, sampling (by natural sampling and a sample-and-hold schemes) and reconstruction
will be studied. The effect of aliasing will be examined as well.

Theory:

In communication systems like AM and FM, the instantaneous value of the message signal is
used to change certain parameter of the carrier signal. Pulse modulation systems differ from
these systems in a way that they transmit a limited number of discrete states of a signal at a

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predetermined time. Sampling can be defined as measuring the value of a message signal at
predetermined time intervals. The rate of which the signal is sampled is known as the sam-
pling rate or sampling frequency. It is the major parameter, which decides the quality of the
reproduced signal. If the signal is sampled quite frequently (whose limit is specified by Nyquist
Criterion) then it can be reproduced exactly at the receiver with no distortion. Sampling is the
first step in digitising an analogue signal.

When the message signal is a simple sine wave, the sampled signal in this case consists of the
following (why?):
ˆ A sine wave at the same frequency as the message.
ˆ A pair of sine waves that are the sum and difference of the fundamental and message
frequencies.
ˆ Many other pairs of sine waves that are the sum and difference of the sampling signals’
harmonics and the message.
Nyquist Criterion: The lowest sampling frequency that can be used without the side bands
overlapping is twice the highest frequency component present in the message signal. If we
reduce this sampling frequency even further, the side bands and the message signal will overlap
and the message signal can not be recovered simply by low pass filtering. This phenomenon is
known as fold-over distortion or aliasing.

5.1 Sampling a simple message

ˆ Locate the Dual Analogue Switch module and connect the setup shown in Figure 9 using
one of the two inputs. It uses an electronically controlled switch to connect the message
signal (the 2 kHz SINE from the Master Signals module) to the output. The switch is
controlled (opened and closed) by the 8 kHz digital output of the Master Signals module.
ˆ Draw the two signals of Ch-1 and Ch-2 to the same scale in the space provided below
using different colours. [2 marks]

Question 7 [2 marks]
– What is the name of this sampling type?

Question 8 [2 marks]
– Why it is called by that name?

ˆ In the space provided below in Figure 10, draw the frequency domain view of the sampled
signal. [1 mark]
ˆ Modify the setup as shown in Figure 11 below.
The electronically controlled switch in the original setup has been substituted by a sample-
and-hold circuit. The message and sampling signals remain the same (2 kHz sine wave
and 8 kHz pulse train, respectively).

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 Figure 9: Natural sampling system.

ˆ Draw the two signals of Ch-1 and Ch-2 to the same scale in the space provided below
using different colours. [2 marks]

Question 9 [2 marks]
– What is the name of this sampling type?

Question 10 [2 marks]
– Why it is called by that name?

5.2 Speech sampling

ˆ Sample the speech signal provided from the speech module by 8 kHz signal. Show your
connections and results to one of the demonstrators.

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 Figure 10: Frequency domain view.

Figure 11: Sample-and-hold sampling system.

Question 11 [2 marks]

ˆ From your general knowledge, what is the theoretical frequency range for the speech signal?

.....................

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5.3 Reconstructing a sampled message
Theory:

Recall that the sampled message is made up of many sine waves. Importantly, for every sine
wave in the message, there’s a sine wave 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 sine wave(s) with the same frequency as the message pass through
while rejecting other sine waves.
ˆ Locate the low-pass filter and modify the setup as shown in Figure 12 below.
The Tunable Low-pass Filter module is used to recover the message. The filter is tun-
able because the point at which frequencies are rejected (called the cut-off frequency) is
adjustable. At this point and after, there should be nothing out of the Tunable Low-pass
Filter module. This is because it has been set to reject almost all frequencies, even the
message. However, the cutoff frequency can be increased by turning the module’s Cut-off
Frequency adjust control clockwise.
ˆ Slowly turn the the Cut-off Frequency control clockwise and stop the moment the message
signal has been reconstructed.
ˆ Draw in the space provided in Figure 12 the signal from Ch-1 and Ch-2 overlapped and
to the same scale with different colours. [2 marks]

Question 12 [2 marks]

ˆ What is the bandwidth of the Low-pass filter? Why?

.....................

5.4 Aliasing
The filter is only letting the message signal pass through to the output and rejecting all other
frequencies (or sine waves) that make up the sampled message. This is only possible because
the frequency of these sine waves is high enough. Their frequency is set by the sampling rate
(that is, the sampling signal’s frequency).

Now, suppose the frequency of the sampling signal is lowered. You’d still get the message but
the other sine waves would have a lower frequency. If the sampling signal’s frequency is low
enough, one or all of the other sine waves could pass through the filter along with the mes-
sage. Obviously, this would distort the reconstructed message which is a known problem called
aliasing.

To avoid aliasing, the sampling signal’s theoretical minimum frequency is chosen to be twice the
message frequency (or twice the highest frequency in the message if it contains more than one
sine wave). This figure is known as the Nyquist Sample Rate and helps to ensure that the
frequencies of the non-message sine waves in the sampled signal are higher than the message’s
frequency. 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.

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Figure 12: Reconstruction system.

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 ˆ Locate the VCO module and modify the circuit in Figure 12 to get the circuit in Figure 13,
that is to replace the fixed 8 kHz sampling signal by a variable frequency signal from the
VCO.

Figure 13: Reconstruction system-Aliasing.

ˆ By starting from the maximum frequency that the VCO can generate (fully clockwise),
reduce the frequency slowly while observing the reconstructed signal.

Question 13 [2 marks]
ˆ Measure the minimum sampling frequency without getting aliasing?

.....................

Question 14 [2 marks]
ˆ What is the minimum theoretical sampling frequency at which the message signal (2 kHz)
can be reconstructed without distortion? Why?

.....................

Question 15 [2 marks]
ˆ Why is the measured value larger than the theoretical value?

6 Part C: PCM encoding (25 Marks)

Digital transmission systems are steadily replacing analogue ones in commercial communica-
tions applications. This is especially true in telecommunications. Hence, an understanding of
digital transmission systems is crucial for technical people these industries.

PCM is a system for converting analogue message signals to a serial stream of 0s and 1s. The
conversion process is called encoding. In general, encoding involves:
ˆ Sampling the analogue signal message at regular intervals using a sample-and-hold scheme.

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 ˆ Comparing each sample to a set of reference voltages called quantisation levels.
ˆ Deciding which quantisation level the sampled voltage is closest to.
ˆ Generating the binary number for that quantisation level.
ˆ Transmitting the binary number one bit at a time (that is, in serial form).
ˆ Taking the next sample and repeating the process.

An factor that is crucial to the performance of the PCM system is the encoder’s clock frequency.
The clock tells the PCM encoder when to sample and, as the previous experiment shows, this
must be at least twice the message frequency to avoid aliasing. Another important PCM per-
formance factor is the difference between the sample voltage and the quantisation levels that
it is compared to. To explain, most sampled voltages will not be the same as any of the quan-
tisation levels. As mentioned above, the PCM Encoder assigns to the sample the quantisation
level that is closest to it. However, the original sample value is lost and the difference is known
as quantisation error.

The PCM Encoder module built on the kit uses a PCM encoding and decoding chip (called
a codec) to convert analogue voltages between -2 V and +2 V to an 8-bit binary number (a
word). With eight bits, it’s possible to produce 256 different numbers between 00000000 and
11111111 inclusive. This in turn means that there are 256 quantisation levels (one for each
number).

Each binary number is transmitted in frames serially. The most significant bit of each word
(called bit-7) is sent first, bit-6 is sent next and so on to the least significant bit (bit-0). The
PCM Encoder module also generates a separate Frame Synchronisation signal (FS) that goes
high at the same time that bit-0 is transmitted. The FS signal has been included to help in
PCM decoding, but it can also be used to help trigger another sub-system when looking at the
signals that the PCM Encoder module generates.

Figure 14 below shows an example of three frames of a PCM Encoder module’s output data
(each bit could be either 0 or 1) together with its clock input and its FS output.

Figure 14: Three frames PCM data.

Objectives:

In this part, PCM Encoder module will be used to convert the following messages to PCM: a
0 V message, a DC voltage message and a continuously changing signal. In the process, PCM
encoding will be verified and the corresponding quantisation error will be investigated.

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6.1 Encoding a 0 V message
ˆ Locate the PCM Encoder module and set its Mode switch to the PCM position. Connect
the setup shown in Figure 15.

Figure 15: PCM Encoder-0 V message.

ˆ Draw in the space provided of Figure 16 the FS, PCM data and the clock signals, dividing
the graphs into three areas, one for each signal. Annotate in your drawings the start and
the end of each bit and frame, and indicate bit-0 and bit-7 of each frame. [3 marks]

Question 16 [2 marks]
ˆ What is the binary number that the PCM Encoder module is generating?

.....................

Question 17 [2 marks]
ˆ Why does the code change even though the input voltage is fixed?

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Figure 16: FS, PCM data and the clock signal.

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Question 18 [2 marks]
ˆ Why does the PCM Encoder module output this code for 0 V DC and not 00000000?

6.2 Encoding a DC voltage

ˆ Modify the setup as shown in Figure 17 using variable DC voltage instead of 0 V (GND)
signal and keep changing the DC value from the minimum to the maximum and observe
the PCM data output.
What is the analogue voltage for 11111111 binary output? [2 marks]

.....................

What is the analogue voltage for 00000000 binary output? [2 marks]

.....................

Figure 17: PCM encoder-variable DC voltage message.

Question 19 [2 marks]
ˆ What is the maximum amplitude of the analogue signal that can be transmitted by this
PCM system?

.....................

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Question 20 [2 marks]
ˆ What is the resolution of this PCM encoder? Why?

Question 21 [2 marks]
ˆ What is the name of the difference between a sampled voltage and its closest quantisation
level?

Question 22 [3 marks]

ˆ How could you reduce this difference?

6.3 Encoding continuously changing voltages

Let us see what happens when the PCM encoder is used to convert continuously changing
signals like speech.
ˆ Modify the setup as shown in Figure 18 below.

Figure 18: PCM encoder-Speech message.

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Question 23 [3 marks]
ˆ Why does the code on PCM Encoder module’s output change even when you are not
making a sound?

7 Part D: PCM decoding (10 Marks)

Theory:

The previous part introduced you to the basics of PCM which is a system for converting message
signals to a continuous serial stream of binary numbers (encoding). Recovering the message
from the serial stream of binary numbers is called decoding.
In general, decoding involves:
ˆ Identifying each new frame in the data stream.
ˆ Extracting the binary numbers from each frame.
ˆ Generating a voltage that is proportional to the binary number.
ˆ Holding the voltage on the output until the next frame has been decoded (forming a pulse
amplitude modulation (PAM) version of the original message signal).
ˆ Reconstructing the message by passing the PAM signal through a low-pass filter.

Objectives:

In this part, sine wave and speech messages will be converted to a PCM data stream then to
a PAM signal using the PCM Decoder module. For this to work correctly, the decoder’s clock
and frame synchronisation signals are simply taken (stolen) from the PCM Encoder module.
The message will be recovered then using the Tunable Low-pass Filter module.

7.1 Decoding the PCM data

ˆ Connect the setup shown in Figure 19 below.

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 Figure 19: PCM decoder.

Question 24 [2 marks]
– What must be done to the PCM Decoder module’s output to reconstruct the message
properly? Why?

ˆ Modify the setup as shown in Figure 20 below.

Slowly turn the Tunable Low-pass Filter module’s Cut-off Frequency control clockwise and
stop the moment the message signal has been reconstructed (ignoring the phase shift).

Figure 20: Modified PCM decoder.

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Question 25 [2 marks]
ˆ Why isn’t the reconstructed message a perfect replica of the original message?

7.2 Encoding and decoding of speech [6 marks]

ˆ Draw in the space provided a diagram to encode and decode the speech signal, and connect
your suggested setup. Show your system and output to one of the demonstrators.
ˆ Add proper module to make the reconstructed (decoded) signal more like the original
message signal.

Figure 21: Speech encode-decoder system.

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8 Part E: Bandwidth limitation and restoring digital signals (15 Marks)
Theory:

In the classical communications model, a useful message moves from a transmitter to a receiver
over a channel. A number of transmission media can be used as channels including: metal
conductors (such as twisted-pair or coaxial cable), optical fibre and free-space. Regardless of
the medium used, all channels have a certain bandwidth. That is, the medium lets a range
of frequencies pass relatively unaffected while frequencies outside the range are made smaller
(or attenuated). In this way, the channel acts like a filter, an issue that in fact has important
implications. Recall that the modulated signal in analogue and digital modulation schemes
consists of many sine waves. If the medium’s bandwidth isn’t wide enough then some of the
sine waves are attenuated and others may be completely lost. In both cases, this causes the
demodulated signal (the recovered message) to be a non-faithful reproduction of the original
version. Making the matter even worse, the channel is like a filter in that it shifts the phase
of the sine waves by different amounts (why?). Imagine the difficulty a digital receiver circuit
such as a PCM decoder would have trying to interpret the logic levels of a received signal that
has been tackled by the above factors. In this case, codes will be misinterpreted and incorrect
voltages are generated. This makes the recovered message noisy which is obviously a problem.

Objectives:

In this part, a PCM communication system will be setup. Then, bandwidth limitation of the
channel will be modelled by introducing a low-pass filter. The effect of bandwidth limitation
on the PCM data will be observed using an oscilloscope as well as listening to the effect it has
on the recovered voice message. Finally, a comparator will be used to recover a digital signal
and observe its limitations.

8.1 The effects of bandwidth limitation on PCM decoding

Bandwidth limitation in a channel can distort digital signals and upset the operation of the
receiver. This part of the experiment demonstrates this using a PCM transmission system.

ˆ Connect the setup shown by the block diagram in Figure 22 below. The Tunable Low-pass
Filter module models the bandwidth limitation of the channel.

Figure 22: Channel bandwidth limitation model.

ˆ Turn the Variable DCV module’s VDC control left and right. At the same time, slowly
turn the Tunable Low-pass Filter module’s Cut-off Frequency Adjust control anti-clockwise
and stop turning it once the PCM Decoder module’s output becomes corrupted.

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Question 26 [2 marks]
ˆ Why does bandwidth limitation of the channel cause the PCM Decoder module to gener-
ate incorrect output?

8.2 The effects of bandwidth limitation on a digital signal shape

Use the Sequence Generator module (32-bit) to model a digital data signal.
ˆ Locate the Sequence Generator module and set its dip switches to 00. Connect the setup
shown in Figure 23 below. Use SYNC output to trigger the oscilloscope to provide a stable
display.

Figure 23: Bandwidth limitation on a digital signal.

ˆ Investigate the effect of the Tunable LPF (the effect of the channel) by making the channel’s
bandwidth narrower by turning the Tunable Low-pass Filter module’s Cut-off Frequency
Adjust control anti-clockwise.

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 Question 27 [2 marks]
– If reducing the channel bandwidth is distorting the signal, how could this be compen-
sated for?

An obvious solution to the problem of channel bandwidth limitation is to use a transmission

medium that has a sufficiently wide bandwidth for the digital data. In principle, this is a
good idea yet could be impractical or not possible always. Certain types of cable design
have better bandwidths than others. However, as digital technology spreads, there are
demands to push more data through the existing channels. To do so without slowing
things down requires that the transmission bit rate be increased. This ends up having the
same basic effect as reducing the channel’s bandwidth. The next part of the experiment
illustrates this.
ˆ Modify the setup as shown in Figure 24 below.
Turn the Tunable Low-pass Filter module’s Cut-off Frequency Adjust control to get the
maximum possible bandwidth. To model increasing the transmission bit-rate, continue
turning the VCO module’s Frequency Adjust control clockwise while observing the oscil-
loscope’s display. Show your connection and output to one of the demonstrators.

Figure 24: Studying bit-rate effect.

8.3 Restoring digital signals

As it was seen before, bandwidth limitation distorts digital signals. In fact, bandwidth limita-
tion is almost inevitable and its effects get worse as the transmission bit-rate increases.

To manage this problem, the received digital signal must be cleaned-up or restored before it is
decoded. A device that is ideal for this purpose is the comparator. The comparator amplifies
the difference between the voltages on its two inputs by large amount. This always produces

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a heavily clipped or squared-up version of any AC signal connected to one input if it swings
above and below a DC voltage on the other input (reference voltage).

This part of the experiment lets you restore a bandwidth limited digital signal using a com-
parator.
ˆ Modify the setup as shown in Figure 25.

Figure 25: Restoring digital signals using a comparator.

Question 28 [2 marks]
ˆ What is the difference between the original and the restored signal?

Slowly turn the Variable DC module’s DC Voltage control to fully clockwise and fully anti-
clockwise positions and observe the effect.

Question 29 [3 marks]
ˆ Why do some DC voltages cause the comparator to output wrong information?

Return the Variable DCV module’s Variable DC control to about the middle of its range.
Slowly make the channel’s bandwidth narrower by turning the Tunable Low-pass Filter mod-
ule’s Cut-off Frequency Adjust control anti-clockwise.

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Question 30 [3 marks]
ˆ Why does the comparator begin to output wrong information when this control is turned
far enough?

Question 31 [3 marks]
ˆ How can the comparator restore the bandwidth limited digital signal when it is so dis-
torted?

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9 Marking Scheme
The marking scheme for the report is as follows:

ˆ Results of Part A: 15 Marks

ˆ Results of Part B: 25 Marks
ˆ Results of Part C: 25 Marks
ˆ Results of Part D: 10 Marks
ˆ Results of Part E: 15 Marks
ˆ The pre-lab test: 10 Marks

10 Plagiarism and Collusion

Plagiarism and collusion or fabrication of data is always treated seriously, and action appro-
priate to the circumstances is always taken. The procedure followed by the University in
all cases where plagiarism, collusion or fabrication is suspected is detailed in the University’s
Policy for Dealing with Plagiarism, Collusion and Fabrication of Data, Code of Practice on
Assessment, Category C, available on https://www.liverpool.ac.uk/media/livacuk/tqsd/
code-of-practice-on-assessment/appendix_L_cop_assess.pdf.

Follow the following guidelines to avoid any problems:

(1) Do your work yourself.
(2) Acknowledge all your sources.
(3) Present your results as they are.
(4) Restrict access to your work.

References
[1] Emona Telecoms-Trainer 101 kit manual, 2008.

Version history
Name Date Version
Dr A Garcı́a-Fernández September 2022 Ver. 3.5
Dr M López-Benı́tez September 2019 Ver. 3.4
Dr A Al-Ataby August 2014 Ver. 3.3
Dr A Al-Ataby October 2013 Ver. 3.2
Dr A Al-Ataby and Dr W Al-Nuaimy October 2012 Ver. 3.1
Dr A Al-Ataby and Dr W Al-Nuaimy October 2011 Ver. 3.0
Dr Nael Al-Zubi October 2009 Ver. 2.0
Dr Nael Al-Zubi August 2008 Ver. 1.0

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