Principles of Communications

Principles of Communications
TABLE OF CONTENTS
TABLE OF CONTENTS ………………………..………………………………………………1
REGULATIONS AND SAFETY RULES ………………………………………………………. 2
TIMS OVERVIEW ………………………………………………………………………………. 4
LABORATORY EXERCISE NO 1 ACTIVE FILTER ……………………………………….…7
LABORATORY EXERCISE NO 2 MODELING EQUATION ………………………………..15
LABORATORY EXERCISE NO 3 COMMUNICATION SYSTEM USING MATLAB ….….21
LABORATORY EXERCISE NO 4 NOISE …………………………………………………. 26
LABORATORY EXERCISE NO 5 DSBSC GENERATION ………………………………….36
LABORATORY EXERCISE NO 6 AMPLITUDE MODULATION ……………….……….. 39
LABORATORY EXERCISE NO 7 SSB GENERATION ………………………………….. 46
LABORATORY EXERCISE NO 8 FREQUENCY MODULATION ………………………. 53
LABORATORY EXERCISE NO 9 ENVELOPE RECOVERY ………………………………. 60
LABORATORY EXERCISE NO 10 FREQUENCY DEMODULATION …………………. 65
LABORATORY EXERCISE NO 11 AMPLITUDE MODULATION SIMULATION …….. 70
College of Engineering and Architecture – Electronics Engineering Department 1

REGULATIONS AND SAFETY RULES

The following Regulations and Safety Rules must be observed in all concerned laboratory locations.
1. It is the duty of all concerned parties who use any electronics and communications laboratory to take all
reasonable steps to safeguard the HEALTH and SAFETY of themselves and all other users and visitors.
2. Be sure that all equipment is properly working before using them for laboratory exercises. Any defective
equipment must be reported immediately to the Lab. Instructors or Lab. Technical Staff.
3. Students are allowed to use only the equipment provided in the experiment manual or equipment used
for senior project laboratory.
4. Power supply terminals connected to any circuit are only energized with the presence of the Instructor or
Lab. Staff.
5. Students should keep a safe distance from the circuit breakers, electric circuits or any moving parts
during the experiment.
6. Avoid any part of your body to be connected to the energized circuit and ground.
7. Switch off the equipment and disconnect the power supplies from the circuit before leaving the
laboratory.
8. Observe cleanliness and proper laboratory housekeeping of the equipment and other related
accessories.
9. Wear proper clothes and safety gloves or goggles required in working areas that involves fabrications of
printed circuit boards, chemicals process control system, antenna communication equipment and laser
facility laboratories.
10. Double check your circuit connections specifically in handling electrical power machines, AC motors
and generators before switching “ON” the power supply.
11. Make sure that the last connection to be made in your circuit is the power supply and first thing to be
disconnected is also the power supply.
12. Equipment should not be removed, transferred to any location without permission from the laboratory
staff.
13. Software installation in any computer laboratory is not allowed without the permission from the
Laboratory Staff.
14. Computer games are strictly prohibited in the computer laboratory.

15. Students are not allowed to use any equipment without proper orientation and actual hands on
equipment operation.
16. Smoking and drinking in the laboratory are not permitted.
All these rules and regulations are necessary precaution in Electronics and Communications Laboratory to
safeguard the students, laboratory staff, the equipment and other laboratory users.

TIMS Overview
Throughout the course, we will be using the laboratory equipment 301C PC-based from TIMS® to
complement and demonstrate the theoretical part of the course. We will devote this experiment to introduce
the equipment and get familiar with its usages.
TIMS is a telecommunications modeling system that models block diagrams representing
telecommunications systems. Physically, TIMS is a dual rack system; the upper rack accepts up to 12 plug-
in cards, or modules; the lower rack houses a number of fixed modules, as well as the system power
supply.
The modules are simple electronic circuits, which serve as basic communications building blocks. Each
module, fixed or plug-in, has a specific function; functions fall into three categories:
1. Signal Generation - oscillators, variable DC, etc

2. Signal Processing - multipliers, filters, etc
3. Signal Measurement - frequency counter, PC-based instrument inputs.

Some of those modules are classified as basic modules while others are advanced modules. The fixed
modules are all basic. They include: BUFFER AMPLIFIERS, FREQUENCY AND EVENT COUNTER,
HEADPHONE AMPLIFIER, MASTER SIGNALS, TRUNK PANEL, VARIABLE DC and PC-BASED
INSTRUMENT INPUT. The list of available plug-in modules is shown in the table below.
A data sheet for each module describing its input(s), output(s), configurable parameters and function can
be found in the User Manuals (Basic and Advanced) available in the lab bench drawers. A soft copy is also
available on all laboratory computers‟ desktop.
All TIMS modules conform to the following conventions:

• Signal interconnections are made via front panel sockets
• Sockets on the left hand side are for module inputs.
• Sockets on the right hand side are for module outputs.
• Yellow sockets are for analog signals.
• Red sockets are for digital signals.
• Analog signals are held near the level of 4V p-p.
• Digital signals are TTL level, 0 to 5 V.
• The green socket is the system Ground.
• Any plug-in module may be placed in any of the 12 positions of the upper rack.
• All modules use the back plane bus to obtain power supply.
• The modules can be plugged-in or removed without turning off the power.

It is important to note that:

• The plug-in modules are not firmly locked in the rack, and need to be held in position while
interconnecting leads are removed.
• When removing the leads, hold them from their solid heads and DO NOT PULL them from the flexible
segment, in order not to damage the wires.
• There are 22 plug-in modules. Make sure you leave them in sequence in the storage shelves.

Laboratory Exercise No 1
Active Filter
Course: ECE004 Experiment No.: 1

Group No.: 1 Section: EC42FA1
Group Members: Abustan Date Performed: Nov. 8, 2017
Agosto, Bangcoleng, Cabobos, Date Submitted: Nov. 22, 2017
Calderon A.J. , Calderon A. , Calpotura Instructor: Engr. Daryl David
1. Objective(s):
The activity aims to show how the cut – off frequency/ ies affect the passage of different signals.
2. Intended Learning Outcomes (ILOs):

The students shall be able to:
2.1 Distinguish the different types of filters.
3. Discussion:
Active Filters
Active filters use amplifying elements, especially op amps, with resistors and capacitors in their
feedback loops, to synthesize the desired filter characteristics. Active filters can have high input impedance,
low output impedance, and virtually any arbitrary gain. They are also usually easier to design than passive
filters Possibly their most important attribute is that they lack inductors, thereby reducing the problems
associated with those components. Still, the problems of accuracy and value spacing also affect capacitors,
although to a lesser degree. Performance at high frequencies is limited by the gain-bandwidth product of
the amplifying elements, but within the amplifier's operating frequency range, the op amp-based active filter
can achieve very good accuracy, provided that low-tolerance resistors and capacitors are used. Active
filters will generate noise due to the amplifying circuitry, but this can be minimized by the use of low-noise
amplifiers and careful circuit design. In Figure 1.1, low – pass, high – pass and band – pass filters are
shown.
Figure 1.1: Filter response vs frequency for low, high and band pass filters.

1. Low – Pass Filter
A low – pass filter passes all frequencies from DC (zero frequency) up to its cutoff frequency f c . At
frequencies above the cutoff frequency, the output is greatly attenuated.
There are a number of active low-pass filter circuits, and one of the more commonly used is shown in
Figure 1.2. This circuit is a second-order low – pass active filter, and is a type of voltage – controlled –
voltage – source (VCVS). It is also known as a Sellen – Key filter. Because there are two R – C pairs that
control the frequency response, it is second order. These are R1, C1 and R2, C2.
Figure 1.2. Active Low – Pass Filter
If we set R1 = R2 and C1 = 2C2, the cutoff frequency is given by:

1
fc 
2 ( R1)( R 2)(C1)(C 2)
This is a unity – gain VCVS filter. It provides a maximally flat passband response (Butterworth) below the
cutoff frequency, and a response that rolls off (attenuates) at 40dB/decade above the cutoff frequency. For
R1 = R2, R3 = 2R1 minimizes the offset.
Figure 1.3. Frequency response of second order VCVS low-pass filter

2. High – Pass Filter
The complement of the low – pass filter is the high – pass filter. A high – pass version of the second order
VCVS filter is shown in Figure 1.4.
Figure 1.4 - High – pass filter
For C1 = C2 and R2 = 2R1, the cutoff frequency is given by:

1
fc 
2 ( R1)( R 2)(C1)(C 2)
To minimize DC offset, R3 = R2. This circuit has unity gain above f c ,and rolls off at 40 dB per decade
below f c . In Figure 1.5, the frequency response of the high – pass filter vs frequency is given.
Figure 1.5 - High – pass filter

3. Band – Pass Filter
A band – pass filter is a circuit designed to pass signals only in a certain band of frequencies while rejecting
all signals outside this band. Looking at Figure 1.2. 1 - C, the bandwidth (BW) of the filter is defined as the
difference between the upper cutoff frequency f H and the lower cutoff frequency f L . The center
frequency is f o . Actually f o is a geometric mean, because the frequency scale of Figure 2. 1 – C is
logarithmic.
fo  f H  f L
The ratio of the center frequency to the bandwidth is called the Q or quality factor. It is a measure of the
selectivity of the circuit.
f
Q O
(BW )
A band – pass filter may readily be formed by cascading a low – pass filter and a high – pass filter so that
their pass – bands overlap. This is often done to form filters with a wide bandwidth.
A very narrow band – pass filter may be constructed by using a notch filter as a feedback element .The
notch filter blocks the negative feedback signal over a very narrow band of frequencies and creates a very
high gain over the narrow range of frequencies. The notch filter supplies heavy feedback – low amplifier
gain – outside its pass band and attenuates frequencies in these ranges.
4. Equipment:
3 DMM
1 Digital Oscilloscope
2 Probes
1 Power Dual Polarity
1 Function Gen.
De Lorenzo Board DL3155E23 (Operational Amplifier)
5. Procedure:
1. Set the function Generator to 200 Hz with 2Vpp.

2. For the supply voltage, calibrate the dual polarity power supply to -15 and +15 volts. It must be
connected to the -15 and +15 supply of the board located at the upper left part. All common
grounds are interconnected.

Figure 1.6
3. Wire the low-pass filter illustrated on the image.
Figure 1.7
4. Apply a sinusoidal input voltage with Vpp=2 V and position the input signal above the output signal
on the oscilloscope screen.
5. Sweep the input frequency over the range of 200Hz to 10KHz and observe the effect that this has
an output amplitude.
6. Measure the cutoff frequency. This is the frequency where Vo/Vin = 0.707 .
7. Observe and record the filter gain at the indicated frequencies.
Frequency Measured Gain (Vo/Vin)

200 Hz 1
500 Hz 0.99
1000 Hz 0.73
1100 Hz 0.64
1200 Hz 0.6
2000 Hz 0.24
10 kHz 0.028
Table 1.1

8. Remove all connections except for the supplied voltage.

9. Wire the high – pass filter and repeat step 4 to 7.
Figure 1.8

200 Hz
0.037
500 Hz
0.24
1000 Hz
0.71
1100 Hz
0.78
1200 Hz
0.8
2000 Hz
0.96
10 kHz
1
Table 1.2
10. Remove all connections except for the supplied voltages.

11. Wire the band pass filter circuit.

„
Figure 1.9
12. Apply power and position the input signal above the output signal on the oscilloscope screen.
Sweep the frequency over the range of 200Hz to 2 kHz and observe the effect that this has on the
output amplitude.
13. Measure the center frequency. This is the frequency at which the output reaches maximum
amplitude.
14. Observe and record the filter gain at the indicated frequencies.

200 Hz
0.03942
400 Hz
0.073461
600 Hz
0.113461
800 Hz
0.15192
1200 Hz
0.230769
1400 Hz
0.26358
2000 Hz
0.4038
Table 1.3
15. Remove all connections. Turn off signal generator, power supply and oscilloscope.

6. Observation:
Based on the circuit operation, we have observed that on a low-pass filter, if the resulting gain
exceeded the cut off frequency there exist a great attenuation which causes a decreasing output
voltage and gain. The graph of the low pass-filter, gain will gradually decrease starting from the
exceeded gain. However, we observed that once the gain exceeded the cut off frequency, this
will result to a low attenuation causing the output voltage and gain to be higher.
7. Interpretation:
8. Conclusion:
9. Assessment (Rubric for Laboratory Performance):

Modeling Equation
Course: ECE-004 Experiment No.: 2

Group No.: 1 Section: EC42FA1
Group Members: Abustan, Agosto, Bangcoleng Date Performed: Nov. 22, 2017
Cabobos, Calderon A.J., Calderon A. , Calpotura Date Submitted: Nov. 29, 2017
Instructor: Engr. Daryl David
1. Objective(s):
The activity aims to enhance and develop analytical and experimental skills in using TIMS modeling
equipment.

2.1 Formulate mathematical equation based on illustration of TIMS equipment.
3. Discussion:
This experiment illustrates how TIMS is use to model a mathematical equation. It will serve to
introduce you to the TIMS system, and prepare you for more serious experiments to follow.
An Equation to Model
In this experiment, you are going to demonstrate that the two AC signals of the same frequency,
equal amplitude and opposite phase, when added, will sum to zero.
The equation you are going to model is:
Y(t) = V1sin(2πf1t) + V2sin(2πf2t + α) .....1
= V1(t) + V2 (t) .....2
Here y(t) is described as the sum of two sine waves.
Each is of the same frequency: f1 = f2 Hz .....3
Each is of the same amplitude: V1 = V2 volts .....4
and the 180° out of phase: α = 180 degrees .....5
then: y(t) = 0 .....6

Block diagram to represent eqn. (1) is suggested in figure 1.
Figure 2.1
In the block diagram of figure 1, it is assumed by convention that the ADDER has unity gain
between each input and the output. Thus the output is y(t) of eq. 2
A suitable arrangement is illustrated in block diagram form in figure 2.
Figure 2.2
4. Equipment:
TIMS – 301C System Unit

Adder
Audio Oscillator
Phase Shifter

5. Procedure:
1. Set the on board range switch of the PHASE SHIFTER to „LO‟. Its circuitry is designed to give a
wide phase shift in either the audio frequency range (LO), or the 100 kHz range (HI). Plug the
three modules (audio oscillator, phase shifter, adder) into the TIMS SYSTEM UNIT.
2. Set the front panel switch of the FREQUENCY COUNTER to a GATE TIME of 1s.
3. Set the frequency f1 with the knob on the front panel of the AUDIO OSCILLATOR, to
approximately 1 kHz using the FREQUENCY COUNTER.
4. Connect a patch lead from the lower yellow output of the AUDIO OSCILLATOR to the ext. trig.
terminal of the oscilloscope. Make sure the oscilloscope controls are switched so as to accept
this external trigger; use the automatic sweep mode if it is available.
5. Set the sweep speed of the oscilloscope to 0.5 ms/cm.
6. Do the connection as shown in figure 2.
7. Patch a lead from the input of g of the ADDER to CH1-A of the SCOPE SELECTOR. Set the
lower SCOPE SELECTOR toggle switch up.
8. Patch a lead from the input of G of the ADDER to CH1-A of the SCOPE SELECTOR. Set the
lower SCOPE SELECTOR toggle switch up.
9. Patch a lead from the output of the ADDER to CH1-B of the SCOPE SELECTOR. This signal,
y(t) will be examined later on.
10. Find a sine wave on CH1-A using the oscilloscope and place it in the upper half of the screen.
Find a sine wave on CH2-A and place it in the lower half of the screen.
Draw the diagram.
GRAPH:

11. Vary the coarse control of the PHASE SHIFTER, and slow that the relative phases of these two
signals may be adjusted. Observe the effect of the ±180° toggle switch on the front panel of the
PHASE SHIFTER. Describe what happens.
____________________________________________________________________________
The output shifts to the right because the coarse control are used for varying degree of
of phase shifting. The switch makes the phase shift 180 degrees.
____________________________________________________________________________
_______________________________________________________
12. Switch the SCOPE SELECTOR from CH1-A to CH1-B. CH1-B displays the ADDER output.
13. Remove the patch cords from g input of the ADDER. This sets the amplitude V1 at the ADDER output
to zero, adjust the G gain control of the ADDER until the signal displayed on CH1-B is
about 4 volt peak-to-peak. This is V2.
14. Remove the patch cord from the G input of the ADDER. This sets the V2 output from the
ADDER to zero, and so it will not influence the adjustment of g, replace the patch cords
previously removed from the g input of the ADDER thus restoring V1.
15. Adjust the g gain control of the ADDER until the signal at the output of the ADDER, displayed
on CH1-B of the oscilloscope, is about 4 volts peak-to-peak. This is V1
Draw the diagram; see if the amplitudes of V1 and V2 are equal.
GRAPH:
NULLING of two Signals
16. Set the fine control of the PHASE SHIFTER to its central position. On CH1-B, vary the coarse
control of the PHASE SHIFTER. “rotation in one direction will increase the amplitude, while the
other will reduce it. Continue in the direction which produces a decrease until a minimum is
reached.
“leave the PHASE SHIFTER controls in the position which gives the minimum”

17. Select the G control on the ADDER front panel to vary V2, and rotate it in the direction which
produces a deeper null.
18. Reverse the position of the PHASE SHIFTER toggle switch. Record the amplitude of y(t), which
is now the absolute sum of V1 plus V2. Set the signal to fill the upper half of the screen. What
happens when the 180° switch is flipped back to the null condition, with the oscilloscope gain
unchanged? Graph the output.
The output is out of phase but the amplitude is reversed to each other.
____________________________________________________________________________
____________________________________________________________________________
_____________________________________________________
Graph:
6. Observation:
In this experiment, the group performed different skills using modeling skills as well as using new kind of
module such as oscillator and TIMS. On the first part of our experiment, we do first the assigned
connections and adding two signals and as the experiment goes on we observed that when we toggled-
up the phase shifter it becomes in-phase and when toggled-down it becomes out of phase and also
when we adjusted the coarse, the input and output waves move either left or right. On the second part of
the experiment, in the adder part, when adjusting the max voltage of the "g" the V1 and V2 are both
equal while adjusting the max voltage of the "G" the scale of V1 and V2 becomes smaller.
7. Interpretation:

8. Conclusion:

Communication System using Matlab
Course: ECE-004 Experiment No.: #3

Group No.: Section: EC42FA1
Group Members: Dean Louie Calpotura Date Performed: 12/18/2017
Date Submitted: 01/03/2018
Instructor: Engr. Daryl D. David
1. Objective(s):
The laboratory aims to use Matlab for simulating communication system.

2.1 Understand the basic tools and concepts for simulating communication systems using MATLAB.
3. Discussion:
MATLAB is user-friendly, widely used software for numerical computations. MATLAB is vector-
oriented, that is, it mainly deals with vectors (or matrices). It is assumed that you have used MATLAB
before, and you can do simple operations, as well as create and run *.m files.
4. Equipment:

Multiplier
Phase Shifter
5. Procedure:
1. Write the matlab program for the illustrated communication system.
Figure 3.1

m-File:
2. Define the time interval
ts=0.00001;
t= -0.1:ts:0.1;
3. Define the functions m(t) and c(t)

m=exp(-100*abs(t));
4. Perform the multiplication

g=m.*c;
5. Perform full-wave rectification

y=abs(g);
6. Create the filter

cutoff=1000; [a b]=butter(5,2*cutoff*ts);
7. Get the output after the filter;

z=filter(a,b,y);
8. Plot the input and output on the same graph

figure (1)
plot(t,m,t,z);
legend('Input Signal','Output Signal')
xlabel ('time')
ylabel('amplitude')
title ('Case Study')
GRAPH:

9. Finding the FT of the signals

M=abs(fftshift(fft(m))); G=abs(fftshift(fft(g))); Y=abs(fftshift(fft(y))); Z=abs(fftshift(fft(z)));
10. Creating the vector for the frequency axis

f=[-length(t)/2:length(t)/2-1]/(length(t)*ts);
11. Plotting all FT on one sheet, in a 2x2 matrix format

figure (2)
subplot (221)
plot(f,M)
subplot(222)
plot(f,G)
subplot (223)
plot(f,Y)
subplot(224)
plot(f,Z)
GRAPH:
12. Change m(t) to 2+ sin(2π 1000t) and c(t) to cos(2π 104t) and the cutoff frequency of the filter to 2
kHz. Redo step 2 to 11.

GRAPH: Step number 8
GRAPH: Step number 11
Q1. Discuss the following commands used in every steps of the experiment.

6. Observation:
7. Interpretation:
8. Conclusion:

Noise
Course: Experiment No.:

Group No.: Section:
Group Members: Date Performed:
Date Submitted:
Instructor:
1. Objective(s):
1.1 The activity aims to show how noise affects the signal fidelity.
1.2 The activity aims to generate different signal and noise voltages.

2.1 Differentiate between noiseless and a signal affected by noise.
2.2 Explain the function of filter on a noisy signal.
2.3 Analyze the importance of signal to noise ratio.
3. Discussion:
Two types of channels are frequently required, namely low pass and band - pass.
A low - pass channel by definition should have a bandwidth extending from DC to some upper
frequency limit. Thus it would have the characteristics of a low - pass filter.
A speech channel is often referred to as a low pass channel, although it does not necessarily extend down
to DC. More commonly it is called a baseband channel.
A band - pass channel by definition should have a bandwidth covering a range of frequencies not
including DC. Thus it would have the characteristics of a band - pass filter.
Typically its bandwidth is often much less than an octave, but this restriction is not mandatory.
Such a channel has been called narrow band. Strictly an analog voice channel is a band - pass channel,
rather than low - pass, as suggested above, since it does not extend down to DC. So the distinction
between baseband and band - pass channels can be blurred on occasion. Designers of active circuits often
prefer band - pass channels, since there is no need to be concerned with the minimization of DC offsets.
Over simplification
The above description is an oversimplification of a practical system. It has concentrated all the
band limiting in the channel, and introduced no intentional pulse shaping. In practice the bandlimiting, and
pulse shaping, is distributed between filters in the transmitter and the receiver, and the channel itself. The
transmitter and receiver filters are designed, knowing the characteristics of the channel. The signal reaches
the detector having the desired characteristics.

Noise
Whole books have been written about the analysis, measurement, and optimization of signal-to-
noise ratio (SNR). SNR is usually quoted as a power ratio, expressed in decibels. But remember the
measuring instrument in this experiment is an rms voltmeter, not a power meter.
Although, in a measurement situation, it is the magnitude of the ratio S/N which is commonly
sought, it is more often the which is available. In other words, in a non-laboratory environment, if
the signal is present then so is the noise; the signal is not available alone. In this, and most other
laboratory environments, the noise is under our control, and can be removed if necessary. So that
rather than can be measured directly. For high SNRs there is little difference between the two
measures.
The noise channel model
A representative noisy, band limited channel model is shown in block diagram form in Figure 1 of
the following page. Band limitation is implemented by any appropriate filter. The noise is added before the
filter so that it becomes band limited by the same filter that band limits the signal. If this is not acceptable
then the adder can be moved to the output of the filter, or perhaps the noise can have its own band limiting
filter.
Figure 4.1: channel model block diagram
Controllable amounts of random noise, from the noise source, can be inserted into the channel
model, using the calibrated attenuator. This is non-signal-dependent noise. For low - pass channels low -
pass filters are used. For band - pass channels band - pass filters are used.
Signal dependent noise is typically introduced by channel non-linearity, and includes inter
modulation noise between different signals sharing the channel (cross talk). Unless expressly stated
otherwise, in TIMS experiments signal dependent noise is considered negligible. That is, the systems must
be operated under linear conditions.

Diagrammatic representation
In patching diagrams, if it is necessary to save space, the noisy channel will be represented by the
block illustrated in Figure 2 below.
Figure 4.2 : the macro CHANNEL MODEL module
Note it is illustrated as a channel model module. Please do not look for a physical TIMS module when
patching up a system with this macro module included. This macro module is modelled with five real TIMS
modules, namely:
1. An INPUT ADDER module.
2. A NOISE GENERATOR module.
3. A bandlimiting module. For example, it could be:
a. Any single filter module; such as a TUNEABLE LPF (for a baseband channel).
b. A BASEBAND CHANNEL FILTERS module, in which case it contains three filters, as well
as a direct through connection. Any of these four paths may be selected by a front panel
switch. Each path has a gain of unity. This module can be used in a baseband channel.
The filters all have the same slot bandwidth (40 dB at 4 kHz), but differing passband
widths and phase characteristics.
c. A 100 kHz CHANNEL FILTERS module (or any filter module), in which case it contains
two filters, as well as a direct through connection. Any of these three paths may be
selected by a front panel switch. Each path has a gain of unity. This module can be used in
a bandpass channel.
4. An OUTPUT ADDER module, not shown in Figure 1, to compensate for any accumulated DC
offsets, or to match the DECISION MAKER module threshold.
5. A source of DC, from the VARIABLE DC module. This is a fixed module, so does not require a slot
in the system frame.
Thus the CHANNEL MODEL is built according to the patching diagram illustrated in Figure 3 below, and
(noting item 5 above) requires four slots in a system unit.

Figure 4.3 : details of the macro CHANNEL MODEL module
Channel gain
Typically, in a TIMS model, the gain through the channel would be set to unity. This requires that
the upper gain control, 'G', of both ADDER modules, be set to unity. Both the BASEBAND CHANNEL
FILTER module and the 100 kHz CHANNEL FILTER module have fixed gains of unity. If the TUNEABLE
LPF is used, then its adjustable gain must also be set to about unity.However, in particular instances, these
gains may be set otherwise.
Noise level
The noise level is adjusted by both the lower gain control 'g' of the INPUT ADDER and the front
panel calibrated attenuator of the NOISE GENERATOR module. Typically the gain would be set to zero [g
fully anti-clockwise] until noise is required. Then the general noise level is set by g, and changes of precise
magnitude introduced by the calibrated attenuator.
Theory often suggests to us the means of making small improvements to SNR in a particular
system. Although small, they can be of value, especially when combined with other small improvements
implemented elsewhere. An improvement of 6 dB in received SNR can mean a doubling of the range for
reception from a satellite, for example
4. Equipment:

Noise Generator
Adder
Baseband Channel Filter

5. Procedure:
Filter amplitude response
A typical measurement arrangement is illustrated in Figure 4.4 below.
Figure 4.4 : measurement of filter amplitude response
In the arrangement of Figure 4:

 The audio oscillator provides the input to the filter, at the TIMS ANALOG REFERENCE LEVEL,
and over a frequency range suitable for the filter being measured.
 The BUFFER allows fine adjustment of the signal amplitude into the filter. It is always convenient to
make the measurement with a constant amplitude signal at the input to the device being measured.
The TIMS AUDIO OSCILLATOR output amplitude is reasonably constant as the frequency
changes, but should be monitored in this sort of measurement situation.
 The filter can be selected from the three in the module by the front panel switch (positions #2, #3,
and #4). Each has a gain in the passband of around unity. Remember there is a' straight
through , path -switch position # l
 The WIDEBAND TRUE RMS METER will measure the amplitude of the output voltage
 The FREQUENCY COUNTER will indicate the frequency of measurement
 The OSCILLOSCOPE will monitor the output waveform. With TIMS there is unlikely to be any
overloading of the filter if analog signals remain below the TIMS ANALOG REFERENCE LEVEL;
but it is always a good idea in a less controlled situation to keep a constant check that the analog
system is operating in a linear manner -not to big and not too small an input signal. This is not
immediately obvious by looking at the WIDEBAND TRUE RMS METER reading alone. Note that
the oscilloscope is externally triggered from the constant amplitude source of the input signal.
The measuring procedure is:
1. Decide upon a frequency range and the approximate frequency increments to be made over this
range. A preliminary sweep is useful. It could locate the corner frequency, and the frequency
increments you choose near the comer (where the amplitude-frequency change is fastest) could be
closer together.
2. Set the AUDIO OSCILLATOR frequency to the low end of the sweep range. Set the filter input
voltage to a convenient value using the BUFFER AMPLIFIER. A round figure is often chosen to
make subsequent calculations easier - say 1 volt rms. Note that the input voltage can be read;
without the need to change patching leads, by switching the front panel switch on the BASEBAND
CHANNEL FILTERS module to the straight-through condition -position # 1. Record the chosen

input voltage amplitude

______________________________________________________________________________
_______________________________________________________________
3. Switch back to the chosen filter, and record the output voltage amplitude and the frequency
4. Tune to the next frequency. Check that the input amplitude has remained constant; adjust, if
necessary, with the BUFFER AMPLIFIER. Record the output voltage amplitude and the
measurement frequency.
______________________________________________________________________________
______________________________________________________________________________
__________________________________________________________
5. Repeat the previous Task until the full frequency range has been covered.
6. Make a graph of your results. Choose your scales wisely. Compare with the theoretical response.
______________________________________________________________________________
______________________________________________________________________________
__________________________________________________________
Graph:
Signal to noise ratio
This next part of the experiment will introduce you to some of the problems and techniques of
signal-to-noise ratio measurements.
The maximum output amplitude available from the NOISE GENERATOR is about the TIMS
ANALOG REFERENCE LEVEL when measured over a wide bandwidth -that is, wide in the TIMS
environment, or say about 1 MHz. This means that, as soon as the noise is bandlimited, as it will be in this
experiment, the rms value will drop significantly* (To overcame the problem the noise could first be
bandlimited, then amplified).
You will measure both , (ie, SNR) and , and compare calculations of one from a
measurement of the other.

The uncalibrated gain control of the ADDER is used for the adjustment of noise level to give a
specific SNR. The TIMS NOISE GENERATOR module has a calibrated attenuator which allows the noise
level to be changed in small calibrated steps.
Within the test setup you will use the macro CHANNEL MODEL module already defined. It is
shown embedded in the test setup in Figure 5 below.
Figure 4.5 : measurement of signal-to-noise ratio
As in the filter response measurement, the oscilloscope is not essential, but certainly good
practice, in an analog environment. It is used to monitor waveforms, as a check that overload is not
occurring.
The oscilloscope display will also give you an appreciation of what signals look like with random
noise added.
7. Set up the arrangement of Figure 5 above. Use the channel model of Figure 3. In this experiment
use a BASEBAND CHANNEL FILTERS module (select, say, filter #3).
You are now going to set up independent levels of signal and noise, as recorded by the WIDEBAND
TRUE RMS METER, and then predict the meter reading when they are present together. After bandlimiting
there will be only a small rms noise voltage available, so this will be set up first.
8. Reduce to zero the amplitude of the sinusoidal signal into the channel, using the 'G ' gain control of
the INPUT ADDER.
9. Set the front panel attenuator of the NOISE GENERATOR to maximum output.
10. Adjust the gain control 'g' of the INPUT ADDER to maximum. Adjust the 'G' , control of the
OUTPUT ADDER for about 1 volt rms. Record the reading.
______________________________________________________________________________
__________________________________________________________________
The level of signal into the BASEBAND CHANNEL FILTERS module may exceed the TlMS
ANALOG REFERENCE LEVEL, and be close to overloading it - but we need as much noise out
as possible. If you suspect overloading, then reduce the noise 2 dB with the attenuator, and check
that the expected change is reflected by the rms meter reading. If not, use the INPUT ADDER to
reduce the level a little, and check again.

Before commencing the experiment proper have a look at the noise alone; first wideband, then
filtered.
11. Switch the BASEBAND CHANNEL FILTERS module to the straight-through connection - switch
position #1. Look at the noise on the oscilloscope. Graph the output.
Graph:
12. Switch the BASEBAND CHANNEL FILTERS module to any or all of the lowpass characteristics.
Look at the noise on the oscilloscope. Graph the output and compare the two.
Graph:
______________________________________________________________________________
______________________________________________________________________________
__________________________________________________________
13. Reduce to zero the amplitude of the noise into the channel by removing its patch cord from the
INPUT ADDER, thus not disturbing the ADDER adjustment.
14. Set the AUDIO OSCILLATOR to any convenient frequency within the passband of the channel.
Adjust the gain 'G' of the INPUT ADDER until the WIDEBAND TRUE RMS METER reads the
same value as it did earlier for the noise level.
15. Turn to your note book, and calculate what the WIDEBAND TRUE RMS METER will read when
the noise is reconnected.
______________________________________________________________________________
________________________________________________________________
16. Replace the noise patch cord into the INPUT ADDER. Record what the meter reads.
______________________________________________________________________________
________________________________________________________________
17. Calculate and record the signal-to-noise ratio in dB.
18. Measure the signal-plus-noise, then the noise alone, and calculate the SNR in dB. Compare with
the result of the previous Task.
______________________________________________________________________________
______________________________________________________________________________
______________________________________________________________________________
____________________________________________________
19. Increase the signal level, thus changing the SNR. Measure both , and , and
predict each from the measurement of the other. Repeat for different SNR for 2 different values of
your choice.
___________________________________________________________________________________
___________________________________________________________________________________
___________________________________________________________________________________
___________________________________________________________________________________
___________________________________________________________________________________
___________________________________________________________________________________
___
6. Observation:

7. Interpretation:
8. Conclusion:

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DSBSC Generation
Course: ECE 004 Experiment No.: 5

Group No.: Section: ECE42FA1
Group Members: Abustan Calderon Date Performed: 10/06/17
Date Submitted: 10/13/17
Instructor: Engr. Daryl D. David
1. Objective(s):
The laboratory exercise aims to generate a DSBSC signal.

2.1 Demonstrate the modulation and demodulation process of DSBSC.
3. Discussion:
Consider two sinusoids, or co sinusoids, cos ut and cos wt. a double sideband suppressed carrier
signal, od DSBSC is define as their product, namely:
DSBSC = E(cos ut. cos wt). . . . . . . . . . . . . . . . . .1
Generally, the context of this experiment is to understood that
w>>u. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .2
This will give to equation 3.

cos ut.cos wt = (Ecos (w-u)t + Ecos (w+u)t)/2 . . . . 3
Equation 3 shows that the product is represented by two new signals, the first one is the sum
frequencies (w+u), and the other one is the frequency difference (w – u).
4. Equipment:

Multiplier
Phase Shifter

5. Procedure:
1. Set up the arrangement in the figure below. From the equation Ecos wt, take the ouput of
the Master Signal 100 kHz cos (wt) to the input of the multiplier.
Figure 5.1
2. The Master Signal, take the output of the 2kHz message but this equation represents Esin
(ut), therefore, use the phase shifter to translate sin (ut) to cos (ut) to produce a DSBSC =
Ecos ut . cos wt.
GRAPH:

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6. Observation:
Based on the experiment conducted, the group have observed that a Double Sideband Suppressed
Carrier signal can be generated using the TIMS system. This DSBSC signal is just simply the product
of two co-sinusoidal signals having different frequencies. At first, the 2kHz message signal was
phase shifted at a degree of 180 (out of phase) in order to change the signal from sin (ut) to cos (ut).
Using the multiplier module, the 100kHz carrier signal and the 2kHz phase shifted signal we’re
multiplied in order to produce and generate the DSBSC signal. This process is called Modulation, by
which the 2kHz message signal was enveloped or carried by the 100 kHz carrier signal. Generally
speaking, the carrier must have higher frequency than the message itself in order to carry/envelope
the message signal.
7. Interpretation:
8. Conclusion:

Amplitude Modulation

Group No.: Section:
Date Submitted:
Instructor:
1. Objective(s):
The activity aims to generate an amplitude modulated signal with varying modulation index.

2.1 Explain the concept of amplitude modulation based on the output produce.
2.2 Interpret the modulation index using the output wave forms.
3. Discussion:
In the early days of wireless, communication was carried out by telegraphy, the radiated signal
being an interrupted radio wave. Later, the amplitude of this wave was varied in sympathy with a speech
message (rather than on/off by a telegraph key) and the message was recovered from the envelope of the
received signal. The radio wave was called a „carrier‟, since it was seen to carry the speech information
with it. The process and the signal was called amplitude modulation, or “AM‟ for short.
The amplitude modulated signal is defined as:
AM = E ( 1 = m.cos µt ) cosωt …1
= A ( 1 = m.cos µt ) . B cosωt …2
= [ low frequency term a9t0 ] x [ high frequency term c(t) ] …3
Here:
„E‟ is the Am signal amplitude from eqn. 910. For convenience eqn. (1) has been written into two
parts in eqn. (2). Where (A.B) + E.
„m‟ is a constant, which, as will soon see, define the „depth of modulation‟. Typically m<1. Depth of
modulation, expressed as a percentage, is 100.m. There is no inherent restriction upon the size of „m‟ in
eqn. (1). This point will be discussed later.
„µ‟ and „ω‟ angular frequencies in rad/s. where µ/(2π) is a radio, or relatively high, carrier
frequency.
Notice that the term a(t) in eqn. (3) contains both a DC component and an AC component. As will
be seen, it is the DC component which gives rise to the term at ω – the „carrier‟ – in the AM signal.
The AC term; m. cos µt‟ is generally thought of as the message, and is sometimes written as m(t).

But strictly speaking, to be compatible with other mathematical derivations, the whole of the low frequency
term a(t0 should be considered the message.
Thus:
A(t) = DC = m(t)
4. Equipment:

Audio Oscillator
Adder
Multiplier
5. Procedure:
1. First patch up according to Figure 1, but omit the input X and Y connections to the Multiplier.
Connect to the two oscilloscope channels, as shown.
Figure 6. 1
2. Use the FREQUENCY COUNTER to set the AUDIO OSCILLATOR to about 1 kHz.
3. Look at the message from the AUDIO OSCILLATOR. Adjust the oscilloscope to display two or
three periods of the sine wave in the top half of the screen. Now start adjustments by setting up
a(t ) , as defined by (4), and with m  1 .
4. Turn both g and G fully anti-clockwise. This removes both the DC and the AC parts of the
message from the output of the ADDER.
5. Turn the front panel control on the VARIABLE DC module almost fully anticlockwise (not

critical). This will provide an output voltage of about minus 2 volts. The ADDER will reverse its
polarity, and adjust its amplitude using the „g‟ gain control.
6. While watching the ADDER output, rotate the gain „g‟ of the ADDER clockwise to adjust the DC
term at the output of the ADDER to exactly 1 volt above zero. This is „A‟ volts.
7. While watching the ADDER output, rotate the ADDER gain control „G‟ clockwise. Superimposed
on the DC output from the ADDER will appear the message sine wave. Adjust the gain G until
the lower crests of the sine wave are EXACTLY coincident with zero. Graph the output.
Graph:
The sine wave will be centered exactly A volts above zero, and so its amplitude is A. Now the
DC and AC, each at the ADDER output, are of exactly the same amplitude A. Thus, you have
now modeled A(1  m cos t ) A with m  1 . This is connected to one input of the MULTIPLIER,
as required by (2).
8. Connect the output of the ADDER to input X of the MULTIPLIER. Make sure the MULTIPLIER is
switched to accept DC.
9. Connect a 100 kHz analog signal from the MASTER SIGNALS module to input Y of the
MULTIPLIER. Watch the output of the MULTIPLIER and graph the output/

Graph:
10. Measure the peak-to-peak amplitude of the AM signal, with m  1 , and confirm that this
magnitude is as predicted, knowing the signal levels into the MULTIPLIER, and its „k‟ factor.
Specify what you think „k‟ here is.__________________
11. Vary the ADDER gain G, and thus „m‟, and confirm that the envelope of the AM behaves as
expected, including for values of m > 1. Graph two different plots of the AM signal
corresponding to m > 1 and m < 1. Specify what you observed when varying „m‟ and try to
explain.
___________________________________________________________________________________
___________________________________________________________________________________
___________________________________________________________________________________
___________________________________________________________________________________
___________
Graph m >1

Graph m<1
TRAPEZOIDAL DISPLAY
12. Replace the Audio Oscillator output with a speech signal available at the Trunks Panel. How
easy is it to set the Adder gain G to occasionally reach, but never exceed, 100% amplitude?
Figure 6.2
Note: Steps to produce a Trapezoidal pattern in Oscilloscope:
1. Press Horizontal menu in the oscilloscope.

2. Press xy mode (F5)
13. Patch up the arrangement of Figure 2. Note that the oscilloscope will have to be switched to the
„X-Y‟ mode; the internal sweep circuits will have not required.

14. Obtain and graph the trapezoidal display for m<1, m=1 and m>1.
Graph m<1
Graph m = 1
Graph m<1

6. Observation:
7. Interpretation:
8. Conclusion:

SSB Generation

Group No.: Section:
Date Submitted:
Instructor:
1. Objective(s):
The laboratory exercise aims to understand the elimination method in suppressing sidebands

2.1Model the phasing method of SSB Generation
2.2 Estimate Sideband Suppression
2.3Measure PEP of SSB signals
3. Discussion:
The Filter Method
An SSB signal may be derived from this by the use of a suitable bandpass filter- commonly called,
in this application, an SSB sideband filter. The Filter method is the most common method of SSB
generation.
The Phasing Method
The phasing method of SSB generation requires an accurate phasing network, or quadrature
phase splitter (QPS). It is capable of acceptable performance in many applications. The QPS operates at
baseband, no matter what the carrier frequency (either intermediate or final), in contrast to the filter of the
filter method.
Weaver’s Method
In 1956 Weaver published a paper on what has become known either as „the third method‟, or
„Weaver‟s Method‟, of SSB generation.
The SSB signal

When, say, the lower sideband (LSB) is removed, by whatever method, then the upper sideband
(USB) remains.
This is a single frequency component at frequency .it is a (co)sine wave. Viewed on an

oscilloscope, with the time base set to a few periods of , it looks like any other sinewave.
The Envelope
The envelope has been defined as:

Envelope = | A (t) | …..... 6
= A/2(from eqn. 4) ...….... 7
Thus the envelope is a constant (i.e., a straight line) and the oscilloscope, correctly set up, will
show a rectangular band of color across the screen.
Generator Characteristics
A most important characteristic of any SSB generator is the amount of out-of-band energy it
produces, relative to the wanted output. In most cases this is determined by the degree to which the
unwanted sideband is suppressed. A ratio of wanted-to-unwanted output power of 40 dB was once
considered acceptable commercial performance; but current practice is likely to call for a suppression of 60
dB or more, which is not a trivial result to achieve.
A phasing generator
The phasing method of SSB generation is based on the addition of two DSBSC signal, so phased
that their upper sidebands (say) are identical in phase and amplitude, whilst their lower sidebands are
similar amplitude but opposite phase.
The two out-of-phase sidebands will cancel if added; alternatively the in-phase sidebands will
cancel if subtracted.
Degree of Modulation – PEP
The SSB generator, like DSBSC generator, has no „depth of modulation‟, as does for
example, and AM generator. Instead, the output of an SSB transmitter may be increased until some part of
the circuitry overloads giving rise to unwanted distortion components. In a good practical design it is the
output amplifier which should overload first. When operating just below the point of overload the transmitter
output amplifier is said to be producing its maximum peak output power – commonly referred to as the
„PEP‟ – an abbreviation for „Peak Envelope Power‟

4. Equipment:

2 Multipliers
Phase Shifter
Audio Oscillator
Quadrature Phase Splitter
5. Procedure:
1. Set up the arrangement of the figure below. The oscilloscope should be in X-Y mode, with equal
sensitivity in each channel. For the input signal source use an AUDIO OSCILLATOR module.
For correct QPS operation the display should be an approximate circle. We will not attempt to
measure phase error from this display.
Note: To configure XY mode, press Horizontal menu on the oscilloscope then press XY mode.
Figure 7.1
Figure 7.2

2. In the oscilloscope, press horizontal menu then press XY mode ( F5).
3. Vary the frequency of the AUDIO OSCILLATOR, and check that the approximate circle is
maintained over at least the speech range of frequencies.
4. Patch up a model of the phasing SSB generator, following the arrangement illustrated in the
figure below. Remember to set the on-board switch of the PHASE SHIFTER to the „HI‟ (100
kHz) range before plugging it in.
Figure 7.3
5. Set the AUDIO OSCILLATOR to about 1 kHz.
6. Switch the oscilloscope sweep to „auto‟ mode, and connect the output of the Audio Oscillator to
the external trigger. It is now synchronized to the message.
7. Display one or two periods of the message on the upper channel CH1-A of the oscilloscope for
reference purpose. Note that this signal is used for external triggering of the oscilloscope. This
will maintain a stationary envelope while balancing takes place. Make sure you appreciate the
convenience of this mode of triggering.
Separate DSBSC signals should already exist at the output of each MULTIPLIER. These need
to be of equal amplitudes at the output of the ADDER. You will set this up, at first approximately
and independently, then jointly and with precision, to achieve the required output result.
8. Check that out of each MULTIPLIER there is a DSBSC signal. Display the graph. Connect
multiplier 1 to Channel 1B and multiplier 2 to channel 2B.

GRAPH:
9. Return the connection from figure 2 and turn the ADDER gain „G‟ fully anti-clockwise. Adjust the
magnitude of the other DSBSC, „g‟, of Figure 2, viewed at the ADDER output on CH2-A, to
about 4 volts peak-to-peak. Line it up to be coincident with two convenient horizontal lines on
the oscilloscope graticule (say 4cm apart).
10. Remove the „g‟ input patch cord from the ADDER. Adjust the „G‟ input to give approximately 4
volts peak-to-peak at the ADDER output, using the same two graticule lines as for the previous
adjustment.
11. Return the „g‟ input patch to the ADDER. The two DSBSC are now appearing simultaneously at
the ADDER output. Display the output.
GRAPH:

12. Balance the SSB generator so as to minimize the envelope amplitude. During the process it
may be necessary to increase the oscilloscope sensitivity as appropriate, and to shift the display
vertically so that the envelope remains on the screen. Graph the output.
GRAPH:
13. When the best balance has been achieved, record results, using figure below as a guide.
Although you need the magnitudes P and Q, it is more accurate to measure
2P directly, which is the peak-to-peak of the SSB

Q indirectly by measuring (P-Q), which is the peak-to-peak of the envelope.
____________________________________________________________________________
____________________________________________________________________________
_______________________________________________________
As already stated, the TIMS QPS is not a precision device, and a sideband suppression of
better than 26dB is unlikely. You will not achieve a perfectly flat envelope. But its amplitude may
be small or comparable with respect to the noise floor of the TIMS system.
 Leakage of a component at carrier frequency (a fault of one or other MULTIPLIER)

 Incomplete cancellation of the unwanted sideband due to imperfections of the QPS.
 Distortion components generated by the MULTIPLIER modules.
 Twice that of the message, then the largest unwanted components is due to incomplete
cancellation of the unwanted sideband.
 The same as the message, then the largest unwanted component is at carrier
frequency („carrier leak‟)
If it is difficult to identify the shape of the envelope, then it is probably a combination of these
two; or just the inevitable system noise. An engineering estimate must then be made of the

wanted-to-unwanted power ratio (which could the nature of these residual signals.
14. If not already done so, use the FREQUENCY COUNTER to identify your sideband as either
upper (USSB) or lower (LSSB). Record also the exact frequency of the message sine wave
from the AUDIO OSCILLATOR. From your knowledge of carrier and message frequencies,
confirm your sideband is on one or other of the expected frequencies.
____________________________________________________________________________
____________________________________________________________________________
____________________________________________________________________________
________________________________________________
6. Observation:
7. Interpretation:
8. Conclusion:

Frequency Modulation

Group No.: Section:
Date Submitted:
Instructor:
1. Objective(s):
The activity aims to generate FM signals using TIMS modeling equipment

2.1 Generate an FM signal
2.2 Understand the modulator sensitivity and linear range of operations
2.3 Examine the factors affecting the shape of the spectrum and bandwidth of FM
3. Discussion:
A simple and direct method of generating an FM is b the use of a voltage controlled oscillator –
VCO. The frequency of such an oscillator can be varied by the magnitude of an input (control) voltage. The
block diagram of VCO-FM generator is shown in Figure 1(a). Figure 1(b) shows a snap shot of an FM
signal, together with the message from which it was derived. Note particularly that there are no amplitude
variations – the envelope of an FM waveform is a constant.
For the VCO to work as a frequency modulator, it has to manifest a linear operation between the
magnitude of the input signal and the output oscillation. Large signal amplitude may take the system out of
its linear range of operation. Therefore a careful design of the deviation sensitivity of the VCO is required to
ensure linear operation over the full range of input signal amplitudes.
Unlike Amplitude modulation, the bandwidth of FM signals is not determined by the message
bandwidth only, but also by message (maximum) amplitude and deviation sensitivity. The product of the
last two factors yields frequency deviation. The bandwidth of FM signal can be approximated by (Carson‟s
Rule):
Bandwidth of FM signal = 2 x (message bandwidth + frequency deviation
Figure 8.1: FM by VCO (a), and resulting output (b)

4. Equipment:

VCO
Audio Oscillator
5. Procedure:
This experiment has four parts. The first part studies the sensitivity and the range of linear
operation of the voltage controlled oscillator (VCO). In preparation for FM generation in the third
part, part II addresses designing the frequency deviation ratio for the modulator. Spectrum
analysis and bandwidth estimation will be the subject of the last part.
Sensitivity and Linearity of VCO
The output frequency of the VCO varies with the input voltage, Vin. The amount of variation
(Hz/volt) can be controlled by the deviation sensitivity (GAIN). Before generating an FM
waveform it is required to set the deviation sensitivity to a value that ensures linearity of the
VCO over the whole range of message amplitudes.
1. An information signal is generated by audio oscillator.
2. Plug in the VCO and make sure that the on-board switch SW2 is set to „VCO‟. Set it “Hi”.
3. Use the front panel „fo‟ control to set the output frequency (sinωt) close to 100 kHz. Measure the
output of the VCO to the analog counter.
4. Connect the VARIABLE DC voltage to the input (Vin)

of the VCO.
5. The deviation sensitivity can be set with the front

panel GAIN control. Set this to about 25% of is fully
clockwise rotation. See figure for approximate
setting.
6. Vary the VARIABLE DC from -2V to +2V in steps of 0.5 V and measure the output frequency.
Fill up the table.
7. With the variable DC on its minimum value, set the GAIN control (sensitivity) of the VCO to
about 60% (make sure that you
Do not overload the VCO, the LED should not light up). Redo step 5 and tabulate the result
also.

Table 8.1
Vin Output Frequency

(25%) (60%)
-2.0V
-1.5V
-1.0V
-0.5V
0V
+0.5V
+1.0V
+1.5V
+2.0V
8. Plot the output frequency versus the input voltage for each setting.

 Which of the above settings results in a more linear performance in the given range of
Vin?
_____________________________________________________________________
 Determine the linear range for the second case?

_____________________________________________________________________
 Using the table only, estimate the frequency of the VCO when the DC input is 1.75 V for
both settings? Which setting results in easier interpolation? Why?
_____________________________________________________________________
_____________________________________________________________________
_____________________________________________________________________
_____________________________________________________________________
Setting the Frequency Deviation
The frequency deviation is equal to the product of V in,max and Gain. Our objective is to design
the GAIN that yields frequency deviation of ±10 kHz, for an input signal of 4 volts peak-to-peak.
This can be done as follows:
9. Set a DC voltage of -2V as input to VCO.
10. Set the GAIN control fully anti-clockwise and the frequency to 100 kHz.
11. Advance the GAIN control until the frequency changes by 10 kHz.
12. Change the VARIABLE DC to +2V and confirm that the deviation is about 10 kHz in the other
direction. Record the measured frequency.
____________________________________________________________________________
FM Generation
13. Replace the DC voltage source with the output from an AUDIO OSCILLATOR. The frequency
deviation will now be about ±10 kHz, since the AUDIO OSCILLATOR output is about 2 volts
peak. Why the frequency counter is still at 100 kHz?
____________________________________________________________________________
____________________________________________________________________________
14. Observe the generated FM on the oscilloscope. Adjust the range and zoom-in to optimize the
view. Try 20μs/div. Graph the output.

Graph:
15. Vary the frequency of the AUDIO OSCILLATOR. Explain the change in the modulated signal.
____________________________________________________________________________
____________________________________________________________________________
____________________________________________________________________________
16. Vary the GAIN of VCO. Explain the change in the modulated signal
____________________________________________________________________________
____________________________________________________________________________
____________________________________________________________________________
Spectrum Analysis and Bandwidth Estimation
Many interesting observation can be made regarding the FM spectrum.
17. Fix the message frequency from the AUDIO OSCILLATOR to 2 kHz and the VCO gain to about
25%. Plot the spectrum, zooming the frequency range (40, 180 kHz).
Plot:

18. Vary the message frequency and describe the impact on the spectrum of the FM signal. Plot the
spectrum of the FM signal at the minimum and maximum frequencies of the AUDIO
OSCILLATOR.
Plot:
19. Reset the frequency of the message to 2 kHz, and vary the deviation ratio (by varying the gain
in the VCO). Describe the effect on the spectrum of the FM signal (make sure you do not
overload the VCO). Plot the spectrum at the minimum value and maximum GAIN setting (before
overload).
Plot:
20. Explain the obtained spectra in light of Carson‟s Rule for bandwidth estimation.
____________________________________________________________________________
____________________________________________________________________________
____________________________________________________________________________
____________________________________________________________________________

6. Observation:
7. Interpretation:
8. Conclusion:

Envelope Recovery

Group No.: Section:
Date Submitted:
Instructor:
1. Objective(s):
The exercise aims to understand the principle of modulation and demodulation using TIMS modeling
equipment
2.1Define the ideal envelope detector
2.2 Knows the limitation of the diode detector as an approximation to the ideal
3. Discussion:
Envelopes are found in the upper and lower sidebands of the AM wave, when you connect the tips
of the carrier. To get the envelope one way is to use a detector. The IDEAL ENVELOPE DETECTOR is a
circuit that takes the absolute value of its input, and then passes the result through a low - pass filter.
Figure 9.1: The ideal envelope recovery arrangement
The purpose of the lowpass filter is to separate the wanted from the unwanted components
generated by the absolute value operation. Another way is the IDEAL RECTIFIER, it‟s a circuit which takes
an absolute value is a fullwave rectifier (note that the operation of rectification is non-linear. The ideal
rectifier is a precision realization of a rectifier, using an operational amplifier and a diode in the negative
feedback arrangement.
4. Equipment:

Multiplier
Audio Oscillator
Adder
Utilities
Tunable LPF

5. Procedure:
1. Connection the illustration below.
Figure 9.2
2. Plug in the TUNEABLE LPF module. Set it to its widest bandwidth, which is about 12 kHz
(front panel toggle switch to WIDE, and TUNE control fully clockwise). Adjust its pass-band
gain to about unity. To do this you can use a test signal from the AUDIO OSCILLATOR, or
perhaps the 2 kHz message from the MASTER SIGNALS module.
3. Model the generator and connect its output to an ideal envelope detector as of figure 2. For
the low-pass filter use the TUNEABLE LPF module.
4. Set the frequency of the AUDIO OSCILLATOR to about 1 kHz. This is your message.
5. Adjust the sweep speed of the oscilloscope to display two periods of the message.
6. Adjust the generator to produce an AM signal, with a depth of modulation less than 100%.
This signal is not symmetrical about zero volts; neither excursion should exceed the 2 volt
peak level.
7. For the case m < 1 observe that the output from the filter (the ideal envelope detector
output) is the same shape as the envelope of the AM signal – a sine wave. Display the
original message and the output of the envelope recovery together, graph the output.

Note: For envelope recovery, toggle the switch of Channel 2 to B2 to see the envelope.
Graph:
8. Repeat the above step for m > 1. Graph the output. Explain what you observed.
____________________________________________________________________________
____________________________________________________________________________
____________________________________________________________________________
______________________________________________
Graph:
9. Vary the message frequency, observe the output of the envelope recovery and try to explain
it.
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10. Connect the signal, whose envelope you wish to recover, directly to the ANALOG INPUT of
the „DIODE + LPF‟ in the UTILITIES MODULE, and the envelope (or its approximation) can

be examined at the ANALOG OUTPUT. You should not add any additional low-pass
filtering, as the true „diode detector‟ uses only a single RC network for this purpose, which is
already included. Graph the plots corresponding to m > 1 and m < 1, respectively. Explain
any difference you observed as compared to the previous case using RECTIFIER + LPF.
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Graph m>1
Graph m<1
6. Observation:

7. Interpretation:
8. Conclusion:

Frequency Demodulation

Group No.: Section:
Date Submitted:
Instructor:
1. Objective(s):
The activity aims to understand the demodulation process using TIMS modeling equipment
2.1Implement the Phase Locked Loop for FM demodulation
2.2 Implement Frequency Discriminator for demodulating FM
3. Discussion:
The block diagram of a phase locked loop (PLL) is shown in Figure 1. The principle of operation is
simple. Suppose there is an unmodulated carrier at the input. If the VCO was tuned precisely to the
frequency of the incoming carrier, ω0, then the instantaneous output would be a DC voltage, of magnitude
depending on the phase difference between the output of the VCO and the incoming carrier. Now suppose
that the incoming carrier started to drift slowly in frequency. Depending upon which way it drifts, the output
voltage will vary accordingly. If the incoming carrier is frequency modulated by a message, the output of the
PLL will follow the message.
Figure 10.1: the PLL
FM can be demodulated as well by using a differentiator or a frequency discriminator. Frequency

discrimination can be achieved by applying the FM signal to the linear part (transition region) of a BPF as
depicted in Figure 2. The output of the discriminator is both FM and AM modulated. The message can be
recovered by applying the discriminator output to an envelope detector followed by LPF.

Figure10.2: Band Pass Filter as a Frequency
4. Equipment:

Multiplier
Audio Oscillator
Tunable LPF
VCO
5. Procedure:
FM Demodulation using PLL
21. Reconstruct the FM modulator as in experiment 6 (FM Mdulation). Let the message be 2 kHz
from the AUDIO OSCILLATOR, the carrier 100 kHz from VCO, and the modulator VCO GAIN
about 25%.
22. Model the PLL demodulator illustrated in Figure 1. For the filter use RC LPF provided in the
UTILITIES Module. In the MULTIPLIER Module set the toggle switch to AC. Draw the
corresponding module diagram.

Figure 10.3: PLL Demodulator
Note: For the filter, use RC LPF provided in the utilities module.
23. Set the VCO in the demodulator to 100 kHz. Set the GAIN control to mid-range position.
24. Connect the output of the modulator to the input of the demodulator.
25. The PLL may or may not lock on to the incoming FM signal. Tune the GAIN (and if necessary
the center frequency) of the PLL-VCO until you obtain lock. Examine the output of the PLL-VCO
and compare it with the original message through graph.
GRAPH:

Frequency Discriminator
26. Set the VCO to generate an FM signal with carrier frequency 85 kHz and GAIN around 25%.
27. Connect the FM signal to the BPF (You may use the 100 kHz CHANNEL FILTER MODULE, set
CHANNEL SELECT to 3 if available).
28. Perform envelope detection by connecting the filter output to the DIODE+LPF in the UTILITIES
module. Graph the output.
Note: Used Baseband Channel Filter to channel 3. Perform envelope detection by connecting
the filter output to the DIODE + LPF in the utilities module.
GRAPH:
6. Observation:

7. Interpretation:
8. Conclusion:

Amplitude Modulation Simulation

Group No.: Section:
Date Submitted:
Instructor:
1. Objective(s):
The activity aims to configure an AM wave using TIMS modeling equipment and generate signal in
Labview.

2.4 Model AM signal using Phase Shifter
2.5 Model and generate AM using labview
3. Discussion:
In experiment 4 entitled Amplitude Modulation, an amplitude modulated signal was defined as in eqn(1).
AM = E. (1 + m.cosμt).cosωt ........ 1
There are other methods of writing this equation; for example, by expansion, it becomes:
AM = E.m.cosμt.cosωt + E.cosωt ........ 2
= DSBSC + carrier ........ 3
The depth of modulation „m‟ is determined by the ratio of the DSBSC and carrier amplitudes, since, from
eqns. (2) and (3):
Ratio (DSBSC/carrier) = (E.m) / E = m ........ 4
The important practical detail here is the need to adjust the relative phase between the DSBSC and the
carrier. This is not shown explicitly in eqn. (2), but is made clear by rewriting this as:
AM = E.m.cosμt.cosωt + E.cos (ωt + α) ........ 5
Here α is the above mentioned phase, which, for AM, must be set to:
α = 0° ........ 6
Any attempt to model eqn. (2) by adding a DSBSC to a carrier cannot assume the correct relative phases
will be achieved automatically. It is eqn. (5) which will be achieved in the first instance, with the need for

adjustment of the phase angle α to zero.
A block diagram of an arrangement for modelling eqn. (3) is shown in Figure 1.
Figure 11.1: Block Diagram of AM Generator
4. Equipment:

Audio Oscillator
Adder
Multiplier
Phase Shifter
5. Procedure:
29. Model the AM generator using the block diagram describe by figure 1. Draw your AM model.

30. Use the FREQUENCY COUNTER to set the AUDIO OSCILLATOR to about 4 kHz.
31. An adequate method of phase adjustment, which requires only an oscilloscope, is to first set the
peak amplitude of the DSBSC and the carrier terms to equality. This means 100% amplitude
modulation, assuming the correct phase.
32. Adjust the phase shifter to obtain the desired AM waveform. The modulation index should be
one. Graph the output waveform.
Graph:
33. Describe how your model generates AM.
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6. Observation:
7. Interpretation:
8. Conclusion:

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Principles of Communications

TABLE OF CONTENTS ………………………..………………………………………………1

REGULATIONS AND SAFETY RULES ………………………………………………………. 2

TIMS OVERVIEW ………………………………………………………………………………. 4

LABORATORY EXERCISE NO 1 ACTIVE FILTER ……………………………………….…7

LABORATORY EXERCISE NO 2 MODELING EQUATION ………………………………..15

LABORATORY EXERCISE NO 3 COMMUNICATION SYSTEM USING MATLAB ….….21

LABORATORY EXERCISE NO 4 NOISE …………………………………………………. 26

LABORATORY EXERCISE NO 5 DSBSC GENERATION ………………………………….36

LABORATORY EXERCISE NO 6 AMPLITUDE MODULATION ……………….……….. 39

LABORATORY EXERCISE NO 7 SSB GENERATION ………………………………….. 46

LABORATORY EXERCISE NO 8 FREQUENCY MODULATION ………………………. 53

LABORATORY EXERCISE NO 9 ENVELOPE RECOVERY ………………………………. 60

LABORATORY EXERCISE NO 10 FREQUENCY DEMODULATION …………………. 65

LABORATORY EXERCISE NO 11 AMPLITUDE MODULATION SIMULATION …….. 70

College of Engineering and Architecture – Electronics Engineering Department 1

REGULATIONS AND SAFETY RULES

College of Engineering and Architecture – Electronics Engineering Department 2

College of Engineering and Architecture – Electronics Engineering Department 3

1. Signal Generation - oscillators, variable DC, etc

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All TIMS modules conform to the following conventions:

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It is important to note that:

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Course: ECE004 Experiment No.: 1

2. Intended Learning Outcomes (ILOs):

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1. Low – Pass Filter

Figure 1.2. Active Low – Pass Filter

If we set R1 = R2 and C1 = 2C2, the cutoff frequency is given by:

Figure 1.3. Frequency response of second order VCVS low-pass filter

College of Engineering and Architecture – Electronics Engineering Department 8

2. High – Pass Filter

Figure 1.4 - High – pass filter

For C1 = C2 and R2 = 2R1, the cutoff frequency is given by:

Figure 1.5 - High – pass filter

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3. Band – Pass Filter

1. Set the function Generator to 200 Hz with 2Vpp.

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Frequency Measured Gain (Vo/Vin)

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8. Remove all connections except for the supplied voltage.

Frequency Measured Gain (Vo/Vin)

10. Remove all connections except for the supplied voltages.

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Frequency Measured Gain (Vo/Vin)

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9. Assessment (Rubric for Laboratory Performance):

College of Engineering and Architecture – Electronics Engineering Department 14

Course: ECE-004 Experiment No.: 2

2. Intended Learning Outcomes (ILOs):

The equation you are going to model is:

Y(t) = V1sin(2πf1t) + V2sin(2πf2t + α) .....1

= V1(t) + V2 (t) .....2

Here y(t) is described as the sum of two sine waves.

Each is of the same frequency: f1 = f2 Hz .....3

Each is of the same amplitude: V1 = V2 volts .....4

and the 180° out of phase: α = 180 degrees .....5

then: y(t) = 0 .....6

College of Engineering and Architecture – Electronics Engineering Department 15

Block diagram to represent eqn. (1) is suggested in figure 1.

A suitable arrangement is illustrated in block diagram form in figure 2.