Communication Theory

Military Institute of Science and Technology
Department of Electrical Electronic & Communication Engineering

Communication Theory Lab EECE 310
Experiment No: 01
Name of the Experiment: AM Modulation by Transistor
1.
Objectives:
a. To understand the basic theory of amplitude modulation (AM).
b. To understand the waveform and frequency spectrum of AM modulator, also calculate the
percentage of modulation.
c. To design and implement the AM modulator by using transistor.
d. To understand the measurement and adjustment of AM modulator.
e. To understand the measurement and adjustment of AMmodulator.
2.
Theory: In amplitude modulation, we utilize the amplitude of audio signal to modulate the amplitude
of carrier signal, which means that the amplitude of carrier signal will be varied with amplitude of audio
signal. The waveform of AM modulation is shown in figure. 1.1and its block diagram is shown in fig 1.2. In
fig 1.2, we know that in order to generate the AM signal we just need to add a DC signal with the audio
signal, and then multiply the added signal with the carrier signal.
Figure 1.1 Signal waveform of amplitude modulation
Figure 1.2 Block diagram of AM modulator.

Let the audio signal be Am cos 2f m t and carrier signal be Ac cos 2f c t then the amplitude modulation
can be expressed as
x AM (t ) ADC Am cos 2f m t Ac cos 2f c t

ADC Ac 1 m cos 2f m t cos 2f c t ......
............................................. 1-1
-1-
Where
m Am / ADC .
ADC : DC signal magnitude

Am : Audio signal amplitude.
AC : Carrier signal amplitude.
Fm : Audio signal frequency.
FC : Carrier signal frequency.
m : Modulation index or Depth of modulation.
From equation 1-1, we notice that the variation of the magnitude ADC Ac 1 m cos 2f m t of the carrier
signal can be controlled by parameter m. This means that we can change the magnitude of the audio signal
Am or DC signal ADC to control the level or depth of the carrier signal. Therefore, parameter mis
known as the Modulation index.
Besides, we can also rewrite equation (1-1) as
x AM t
1
ADC AC m cos 2 f c f m t cos 2 f c f m t ADC AC cos 2f c t ..1.2
2
The first term represents double sideband signals; the second term represents carrier signal. From equation
(1-2), we can sketch the frequency spectrum of amplitude modulation as shown in figure 3-3. Since the audio
signal is hidden in the double sidebands and the carrier signal does not contain any message, therefore the
power is consumed in carrier during transmission of amplitude modulation signal. For this reason, the
transmission efficiency of AM modulation is lower than double sidebands suppressed carrier (DSB-SC)
modulation but its demodulation circuit is much more simple.
Figure 1-3 Frequency spectrum signal of amplitude modulation signal.

There is an important parameter m in equation (1-1) called modulation index or depth of modulation.
Normally it is represented in percentage, so we also call modulation percentage. Modulation index is an
important parameter in equation (1-1). The definition is as follow
m=
A
Audio signal amplitude
100% m 100%
DC signal magnitude
ADC
........................................................... 1-3
Generally, the magnitude of DC signal is not easy to measure; therefore we express the modulation index in
another form
E max E min
100% .......................................................................................................... 1-4
E max E min
-2-
Where E max & E min as shown in figure 3-1 are E max AC Am and E min AC Am
We know that at amplitude modulation, the audio signal is hidden in the double sidebands, so if the double
sideband signals are getting stronger, the transmission efficiency is getting better. From equation (1-2) , we
know that the double sideband signals are proportional to the modulation index, Thus the larger the
modulation index, the better the transmission efficiency.
Normally modulation index is smaller or equal to 1. If greater than 1, we call it over modulation, as shown in
fig 1-4. Fig 1-4 shows the waveforms of the over modulation. In figure 1-4, we can see that the variation of
carrier signal is no longer a sinusoidal wave. It is rather a distorted sinusoidal wave, therefore, this kind of
AM signal is unable to demodulated and recover to the original by using the envelop detection in next
chapter.
Figure 1-4 Waveforms of the over modulation

As we know that the AM modulator can be implemented by using a multiplier. However, in electronics
circuit, the multiplier is constructed by the nonlinear characteristics of active component. Therefore, in this
chapter, we will discuss the design of AM modulator by using a single transistor and balanced modulator.
Transistor AM Modulator :
The circuit diagram of transistor AM modulator is shown in fig 1-5. In fig 1-5, the audio signal (
Am cos 2f m t ) will pass through a transformer and send into the base of the transistor. The carrier signal
Ac cos 2f c t also passes through and transformer and sends into the emitter of the transistor. These two
signals will from a small about of small signal voltage difference at the base and emitter of the transistor. The
small signal voltage difference is
Vbe Vb C e Am cos(2f m t ) Ac cos(2f c t ) ..................................... 1-5
Then at the collector of the transistor, this voltage difference will produce a small signal collector current as
I c I s e Vbe / VT .............................................................................................................................. 1-6
Expand equation (3-6) by Taylors expansion, then we get
I c aVbe bVbe2 .............................................................................................................. 1-7
Figure 1-5 Circuit diagram of transistor AM modulator.
-3-
In equation (1-7), we notice that after the audio signal and the carrier signal input into the base and collector
2
2
of the transistor, we can obtain cos (2f m t ) , cos ( 2f c t ) and cos(2f m t ) cos( 2f c t ) Signals at the
collector. Then we utilize the filter to obtain the modulated AM signal cos(2f m t ) cos(2f c t ) . In figure
1-5, the inductor L1, capacitor C3, resister R3 comprise a high pass filter, which is used to obtain the
modulated AM signal. Capacitor C1 is the coupling capacitor. Capacitor C2 and C3 are the bypass capacitors.
Resistors R1 and R2 are the bias resistors. Variable resistor VR 1 is used to change the operation point of the
transistor and it also used to control the magnitude of the carrier, which inputs into the collector of the
transistor. Therefore, it can adjust the output signal waveform of the modulator.
3.
Equipment Needed
ETEK module of AM modulator
Signal source (Function generator)
Oscilloscope.
4.
Procedure :
a) Refer to the circuit diagram in figure 1-5 or figure ACS3-1 on ETEK ACS 3000-02 module.
b) At audio signal input port (Audio I/P), input 100 mV amplitude, 1 kHz sine wave frequency; at
carrier signal input port (carrier I/P), input 300 mV amplitude, 500 kHz sine wave frequency.
c) By using oscilloscope, observe of output signal waveforms of AM output port (AM O/P). Adjust
VR1 so that the modulated AM signal is maximum without distortion. Then record the measured
results in table 3-2.
d) By using oscilloscope, observe of output signal waveforms of the base TP1 and collector TP3 of
he transistor. Then record the measured results table 3-2.
e) By using oscilloscope, observe on output signal waveforms of the mixing (TP2) of the transistor.
Then record the measured results in table 3-2.
f) By using spectrum analyzer observe on the frequency spectrum of AM O/P and TP2. Then record
the measured results in table 3-2.
g) Substitute the measured results into equation (3-4), find the modulation percentage and record in
table 3-2.
h) According to the input signals in table 3-2, repeat step 4 to 7 and record the measured results in
table 3-2.
i) According to the input signals in table 3-3, repeat step 2 to 7 and record the measured results in
table 3- 3.
Table 3-2 Observe on the variation of amplitude modulation by changing the amplitude of audio signal. (f m =
1KHz, fc = 500 KHz, Vc = 300 mV)
Output signal ports
Audio signal amplitudes

100 m V
150 mV
Emax = _________
Emin = _________
m = _________%
Emax = _________
Emin = _________
m = _________%
AM O/P
TP1
TP3
TP2
AM O/P
Output signal
spectrums
TP2
Output signal
spectrums
Modulation
Index
-4-
Table 3-3 Observe on the variation of amplitude modulation by changing the frequency of audio signal. (V m
= 100 mV, fc = 500 KHz, Vc = 300 mV)
Output signal ports
Audio signal amplitudes

100 Hz
400 Hz
Emax = _________
Emin = _________
m = _________%
Emax = _________
Emin = _________
m = _________%
AM O/P
TP1
TP3
TP2
AM O/P
Output signal
spectrums
TP2
Output signal
spectrums
Modulation
Index
5.
Problem discussion
a.
b.
c.
Explain the objectives of the transistor Q1 in figure 1-5.

Explain the objectives of the inductor L1, Capacitor C3 and Resistor R3 in fig 3-5.
When modulation index, m 50% and 110%, what are the ratio of E max & E min ?

-5-

Experiment No: 2
Name of the Experiment: AM Demodulation by Diode detector
1.
Objectives:
To understand the theory of amplitude demodulation.
To design and implement the diode detection amplitude demodulator.
To design and implement the product detection amplitude demodulator.
To understand the measurement and adjustment of AM demodulator.
2.
Theory:
From chapter 3, we know that amplitude modulation signal utilize the amplitude of audio signal to modulate
high frequency carrier signal. Therefore, when we receive the amplitude modulation signal, we need to
restore the audio signal. Fig 4-1 is the theory diagram of amplitude modulation. Normally detector can be
classified as synchronous detector and asynchronous detector. We will discuss these two types of detectors in
this experiment.
Diode detector for Amplitude Demodulation:
Since amplitude modulation signal utilize the amplitude of audio signal to modulate high frequency carrier
signal, which means the variation of carrier signal amplitude is followed by the change of audio signal
amplitude. Hence the objective of amplitude demodulator is to take out the variation envelop detection from
modulation AM signal. Fig 2-2 is the block diagram of diode detector. This circuit is typical asynchronous
detector. It rectifies the modulated AM signal and obtains a positive half wave signal. After that the signal
will pass through a low pass filter and obtain an envelop detection. The n get rid of the DC signal, the aidio
will be recovered. If the input signal of the diode detector is the over modulated AM signal, as shown in fig
2-2. The n we are unable to recover the distorted signal to the audio signal by the diode detector. As for the
over modulated AM signal, we need to use the product detector to demodulated this kind of signal, which
will be discussed in next section.
Figure 2-1 Theory diagram of amplitude demodulator.
Figure 2-2 Block diagram of diode detector.
-6-
Figure 2-3 Circuit diagram of diode detector.

Fig 2-3 is the circuit diagram of diode detector, in which resistors R1, R2, R3, R4, U1 and U2 form two
groups of inverting amplifiers to amplify the input signal, the amplified rate is 10 times of the original signal;
Diode D1 is the rectifier diode which can make the amplitude modulation signal become a positive half wave
signal. Capacitor C1, C2 and Resistor R5, R6 comprise a low-pass filter to remove the envelop detection
signal of audio signal which includes the DC level; then finally the objective of C3 is to block the DC level
and we can obtain a pure audio signal at output port.
3.
Equipment Needed:
a.
b.
c.
4.
ETEK ACS 3000-02 module.

Oscilloscope
Procedure:
Refer to the circuit diagram in figure 3-7 or figure ACS3-2 on ETEK ACS 3000-02 module.
Let J1 be short circuit and J2 be open circuit to produce the modulated AM signal as the
signal source in this experiment.
At audio signal input port (Audio I/P), input 600 mV amplitude, 3 kHz sine wave frequency;
at carrier signal input port (carrier I/P), input 300 mV amplitude, 300 kHz sine wave
frequency.
Adjust VR1 so that the modulation index of the AM signals is maximum adjust VR2 so that the
signal at AM O/P1 is 250 mVp-p.
Connect the output signal of the AM modulator (AM O/P1) to the input port (AM I/P) of
diode detector in figure 4-3 or figure ACS4-1 on ETEK ACS-3000-02 module.
By using oscilloscope and switching to DC channel, observe on the first stage (TP1) and
second stage (TP2) amplified signal waveforms. Then record the measured results in table 41.
By using oscilloscope, observe on the output signal waveforms of the rectifier (TP3). Then
record the measured results in table 2-1.
By using oscilloscope, observe on the output signal waveforms of the low-pass filter (TP4)
and the demodulated AM out [put port (Audio O/P). Then record the measured results in table
4-1.
According to the input signals in table 2-1, repeat step 4 to step 7 and record the measured
According to the input signals in table 2-2, repeat step 3 to 7 and record the measured results
in table 2-2
a.
b.
c.
d.
e.
f.
g.
h.
i.
Table 2-1 Observe on the variation of amplitude modulation by changing the amplitude of audio
signal.(fm=2kHz, fc=300kHz,Vc=300mV)
Output Signal Ports
Audio Signal Amplitudes

600 mV
300mV
AM I/P
TP1
TP2
TP3
TP4
Audio O/P
-7-
Table 2.1
Table 2-2 Observe on the variation of amplitude modulation by changing the frequency of audio
signal.(Vm=600mV, fc=300kHz,Vc=300mV)
Output Signal Ports
Audio Signal Frequencies

3kHz
6 kHz
AM I/P
TP1
TP2
TP3
TP4
Audio O/P
Table 2.2
Problem discussion
a.
b.
c.
d.
e.
f.
g.
By using diagram, explain the reasons why the diode detector is unable to recover the over
modulated AM signal to the original audio signal.
Refer to fig 4-3, explain the results if we connect the output of AM modulator to the diode
detector without two stage amplifier, what will the results be.
Refer to fig 4-3, describe the function of low- pass filter in diode detector. And also explain
what kinds of components comprise the low-pass filter.
Refer to figure 4-5, explain the results if the carrier signal and modulated AM signal are
asynchronous.
Refer to figure 4-5, explain the objectives of VR1 and VR2.
Refer to figure 4-5, explain the objectives of C7, C9 and C8.
Refer to figure 4-5, explain the objectives of VR3 and R5.
-8-

Experiment No: 3
Name of the Experiment: FM Modulation
1.
Objectives:
j) To understand the characteristics of Varactor diodes.
b.
To understand the operation theory of voltage controlled oscillator (VCO).
c.
To design and implement the frequency modulator by using the voltage controlled oscillator.
d.
To design and implement the frequency modulator by using MC4046.
e.
To design and implement the modulator by using LM 566.
2. Theory:
The operation theory of FM Modulation: In frequency modulation (FM), we utilize the amplitude
of audio signal to modulate the frequency of carrier signal. The transmitted high and low frequency signals
will follow the received audio signal, which has different frequency that keeps on changing. The frequency
modulation can be expressed as
X FM t Ac cos t Ac cos 2f c t 2f fx d ..................... 3-1

if x Am cos 2f m ,
Then
f f
x FM t Ac cos 2f c t m sin 2f m t
fm
.................................. 3-2
Ac cos 2f c t sin 2f m t
where
t : Instantaneous modulated frequency.
f c : Carrier frequency
f m : Modulating frequency or audio signal frequency.
: Modulation index, Am f f m .
f
: Frequency deviation.
Frequency deviation of FM x FM t is shown as below
1 d
1 d
(t )
2f c t sin 2f m t
2 dt
2 dt
f c f m cos 2f m t f c Am . f . cos 2f m t ............................... 3-3
From equation (3-3), we know that when the amplitude of modulating signal changes, the Frequency of FM
will change too, and it uses the center point of carrier Frequency to achieve Frequency deviation. From
Carsons role, the bandwidth of modulated signal can be expressed as
A .f
BW 2 2. f m 2 m 2 . f m 2 Am f 2 f m
fm
If the FM signal is the largest amplitude and largest Frequency (i.e.

bandwidth of FM can be simplified as
BW 2 f W
Varactor diode
-9-
Am 1 and f m W )then the
Varactor diode is also called tuning diode. Varactor diode is a diode, which its capacitance can be varied by
adding a reverse bias voltage to pn junction. When reverse bias voltage increases, the depletion region
becomes wide, this will cause the capacitance value decreases, nevertheless when reverse bias voltage
decreases, the depletion region will be reduced, this will cause the capacitances value increases. Varactor
diode also can be varied from the amplitude of AC signal. If an AC signal is added to a Varactor diode, the
variation of capacitance of Varactor diode will follow the amplitude of modulating signal.
Figure 3-1 is the analog diagram of capacitance of varactor diode. When a varactor diode without bias, the
concentration will be different from minor carriers at pn junction. Then these carriers will diffuse and
become depletion region. The P type depletion region carries electron positive ions, then the n type depletion
region carries negative ions. We can use parallel plate capacitor to represent the depletion region.
Figure 3-1 Analog diagram of capacitance of varactor diode.

The transition capacitance pn junction of the plates can be expressed as
A
C
............................................ 3-4
d
where
11.8 0 (dielectric constant of Silicon)
0 8.85 10 12
A: The PN junction area

d: Depletion width
When reverse bias voltage increases, the width of depletion region d will increase but the cross section area
A remains, therefore the capacitance would be reduced . On the other hand the capacitance value will
increase when reverse bias voltage decreases.
Varactor diode can be equivalent to a capacitor series a resistor (RS) and an inductor (LS) as shown in figure
3-2. From figure 3-2 Cj is the junction capacitor of semiconductor which only exist in pn juncton. RS is the
sum of bulk resistor and contact resistor of semiconductor material, which is related to the quality of varactor
diode (generally below a few ohm). L S is the equivalent inductor of bounding wire and semiconductor
material.
- 10 -
Tuning ratio, TR is the ratio of capacitance value under two different biases for varactor diode. The
expression is shown as follow.
Figure 3-2 Equivalent circuit diagram varactor diode.

TR
CV 2
CV 1
........................................................... 3.5
Where
TR : Tuning ratio.
CV1 : The capaciutance value of varactor diode at V1.
CV2 : The capacitance value of varactor diode at V2.
From this experiment, the characteristics of the varactor diode 1SV55 is shown as below
C 3V 42 pF (the capacitance of varactor diode at bias 3V)
TR = 2.65 (3V~30V).
Fig 7-3 block diagram of MC4046.

Figure 3-3 Implementation of FM Modulator by using PLL MC4046
MC 4046 is the phase locked loop (PLL) integrated circuit. Figure 3-3 is the internal structure diagram of
MC 4046. Pin 1, pin 10 and pin 15 are in N.C mode. Pin 5 is the input of INH, which is situated in low
voltage level. The VCO oscillation frequency of MC 4046 is determined by the input voltage at pin 9, the
capacitances at pin 6 and pin 7, the resistances at pin 11 and pin 12.
Figure 3-4 is the circuit diagram of FM modulator by using MC4046. By adjusting the variable resistor VR 1
(DC level), we can control the output frequency at pin 4, which is the frequency f 0 ; Capacitor C2, Resistor R6
and R7 determine the oscillation frequency f0; Capacitor C2 and Resistor R6 determine the maximum
frequency of f0; capacitor C2 and resistor R7 determine the minimum frequency of f0, i.e. the modulation
bandwidth.
- 11 -
Figure 3-4 Circuit diagram of MC 4046 FM modulator.

Implementation of FM modulator by using VCO LM566
LM 566 is voltage controlled oscillator integrated circuit Figure 3-5 is the internal structure diagram of
LM566 . Figure 3-6 shows the circuit diagram of FM modulator by using LM566. We let SW 1 be opened
circuit, and the circuit is a voltage controlled oscillator. The output signal frequency is controlled by C 3, VR1
and audio signal input terminal voltage. C 2 is used to eliminate paracitic oscillation. If C 3 and VR1 remain a
constant, then the output signal frequency and the voltage difference between pin 8 and pin 5 (V 8-V5) is
proportional. In another words, when input signal voltage (V 5) increase, the voltage difference (V8-V5)
between pin 8 and pin 5 will decrease, the output signal frequency will decrease as well. But, when input
signal voltage (V5) decrease, the frequency of output signal will increase. Another factor that affects the
output signal frequency is VR1 C3 value, the output signal frequency and VR 1 C3 is inverse
proportionally. When the VR1 C3 value is getting larger, the output signal frequency is getting lower. But
when the VR1 C3 value is getting the smaller then the output signal frequency is getting higher. From figure
3-6, when we short circuit SW1, then R1 and R2 provide a DC bias voltage as the DC level of input audio
signal. The centre frequency (f0) can be adjustable by using VR1. If audio signal input terminal is inputted
with an AC signal, the VCO output signal frequency will follow the change of the input audio signal voltage,
which the FM signal is deviated.
Figure 3-5: Internal structure diagram of LM566
- 12 -
Figure 7-5 Internal Structure diagram of LM566.
Figure 3-6 Circuit diagram of LM566 FM modulator

3.
a.
b.
c.
4.
Equipment Needed
ETEK ACS 3000-04 module
Oscilloscope
Procedure for MC 4046 FM Modulator
a. Refer to the circuit diagram in figure 3-4 or figure ACS7-1 on ETEK ACS 3000-04 module.
b.
By using oscilloscope, observe of output signal waveforms of modulated FM signal. Adjust variable
resister VR1 so that output signal is 20 kHz square wave. Then record the measured results in table 31.
c. At the audio signal input port (Audio I/P), input 300 mV amplitude and 1 KHz sine wave
frequency. By using oscilloscope, observe on the output signal waveforms of FM O/P, then record the
measured results in table 3-2.
d. According to the input signals in table 4-2, repeat step 3 to 7 and record the measured results in
table 3-2.
Table 3-1 Measured results of MC4046.
FM O/P
Table 3.1
Table 3-2 Measured results of MC4046 FM modulator
(f = 1 KHz, Vm = 300 mV)
FM O/P
Table 3.2
5.
Procedure for LM566 FM Modulator:

a.
b.
Refer to the circuit diagram in figure 3-6 or figure ACS7-2 on ETEK ACS-3000-04 module.
Let J1 be short circuit, i.e. the circuit is the FM modulator. J3 be short circuit and J2 be open
circuit, i.e. the selected capacitor is C4 = 10 nF. Adjust variable resistor VR1 so that the
- 13 -
c.
d.
frequency at the modulated FM output port (FM O/P) is 20 kHz square wave. Then record the
measured results in table 3-3
At the audio signal input port (Audio I/P), input 300 mV amplitude and 1 kHz sine wave
frequency. By using oscilloscope, observe on the output signal waveforms of FM O/P, then
record the measured results in table 3-4.
According to the input signals in table 3-4, repeat step 3 and record the measured results in
table 3-4.
Table 3-3 Measured results of LM566.
(f = 1 KHz, Vm = 300 mV)
FM O/P
Table 3.3
Table 3-4 Measured results of LM566 FM modulator
(f = 1 KHz, Vm = 300 mV)
FM O/P
Table 3.4
6.
Problem discussion
a.
b.
c.
Describe the operation theory of FM modulation.

Explain the implementation of FM modulator by using MC4046.
Explain the implementation of FM modulator by using LM566.

Experiment No: 4
- 14 -
Name of the Experiment: FM Demodulation

1.
Objectives:
a. To understand the operation theory of phase locked loop.
b. To understand the basic characteristics of MC4046 phase locked loop.
c. To understand the basic characteristics of LM565 phase locked loop.
d. To design and implement the FM demodulator by using MC4046.
e. To design and implement the FM demodulator by using LM565.
3.
Theory:
Frequency demodulator is also called frequency discriminator, which can convert the variation of
frequency to the variation of linear voltage. Normally we use FM to AM conversion circuit, balanced
discriminator circuit, phase shift discriminator circuit and PLL synthesizer for the FM demodulator. In this
chapter, we will introduce the phase locked loop frequency demodulator FM to AM conversion
discriminator.
The Operation Theory Of Phase Locked Loop
Phase locked loop or PLL is a feedback circuit. In the feedback loop, the feedback signal will lock the output
signal frequency and phase with the same frequency and phase of the input signal. So, for wireless
communication if the frequency of the carrier signal deviation during transmission, then the PLL in the
receiver will operate andlock the carrier signal. In this experiment, there are two types of using PLL, and the
first is demodulator, which is used fior demodulation by following the variation of phase and frequency. The
second is the carrier frequency tracking which is used to track the changes of the frequency of the carrier
signal and synchronize the oscillation.
Normally, phase locked loop can be divided into 3 sections, there are
1.
Phase detector (PD)
2.
Low-pass filter (LPF)
3.
Voltage controlled oscillator (VCO)
From figure 4-1, the function of phase detector is to receive input signal and VCO signal, then the two
signals are compared by phase detector and provided an output signal, which is a pulse signal. After that this
signal is then sent to a low pass filter to remove the unwanted signal and left the DC voltage.
Fig 4-1 Blocked diagram of Phase locked loop.
This DC voltage can be used to con troll the output signal frequency of VCO. Fig 4-1 the Blocked diagram
of Phase locked loop, Where
- 15 -
Kd = The gain of phase detector (Volts/Radian)

Ka = The gain of amplifier (Volts/Volts)
Ko = The gain of VCO (kHz/Volts)
KL = Kd Ka K0 = The gain of closed loop (kHz/Radian)
We use a simple circuit to explain the basic concept of phase detector. From figure 4-2 (a) shows the phase
difference between two input signal is the smallest, So the output signal pulse width is the narrowest. Then
From figure 4-2 (b) shows the phase difference between two input signal is larger than fig. 4-2 (a), So the
O/P signal pulse width is wider than figure 4-2 (a). Figure 8-2 (c) shows the phase difference between two
input signal is the largest and therefore the output signal pulse width is the widest. If this three output signals
pass through the low pass filter to remove the AC signal, then the magnitude of DC voltage in figure 4-2 is as
follow: 1 Figure 4-2 (c) has the highest DC voltage , 2. figure 4-2 (b) is the second higher, and 3. fig. 4-2 (a)
is the lowest. The relation of DC voltage and the phase difference of A, B input signals is shown in fig 4-2
(d).
Fig 4-2 Theory of phase detector

From fig 4-3, assume that the free running frequency of a VCO is set to 1 kHz (assume the bias voltage is 2
V). If inputting a signal A is below 1 kHz and a signal B is higher than 1 kHz. From figure 8-3, we found
that, when input signal A frequency lower than the free running frequency of VCO, then the output of low
pass filter will receive a lower voltage level (assume is 1 Volt), this lower voltage will adjust the oscillation
frequency of VCO, so that the oscillation frequency will decrease until the frequency of output signal of
VCO and the frequency of signal A equal to each other. When input signal B frequency is higher than the
basic frequency of VCO, the output terminal of low pass filter will receive a higher voltage (assume is 3 V),
so that the oscillation frequency of VCO will increase until the frequency of output signal of VCO and the
frequency of signal B equal to each other. Normally the time needed for VCO locked frequency is very short.
The above mentioned discussion is only the description of the concept, however, practically; the circuit of
phase detector is quit difficult and complicated.
- 16 -
Fig 4-3 Theory of locked frequency

The Basic Characteristics of PLL LM565
(i) Free running frequency
Figure 4-4 is a LM 565 Phase locked loop circuit diagram, from figure 4-4, when input terminal does not
input any signal, the output signal frequency of VCO is called free-running frequency. Where C2 is timing
capacitor, VR1 is timing variable resistor, the free-running frequency (fo) of LM565 is decided by C2 and
VR1.
Free-running frequency : f 0
1
3.7VR1 C 2
Closed loop gain : K L K d K a K o
33.6 f 0
Vc
.............................................
4-1
................................................... 4-2
Where Vc = Total voltage supply = Vcc Vcc 5 5 10V
Figure 4-4 LM565 phase locked loop.
(2) Locked Range

When phase locked loop circuit is at al ready locked situation, assume the input signal frequency
fi
slowly move away from fo, when f i reaches at a certain frequency, the PLL will leave the locked situation.
- 17 -
At this moment, the maximum frequency difference for frequency f i and f 0 is called locked range (refer
to fig 4-5). The locked range of LM565 is
fL
8 f0
VC
....................................................... 4-3
(3) Captured range

At the beginning, LPP is not locked situation, and then let the input signal frequency fi slowly move close to
fo, when fi, reaches at a certain frequency, PLL will be at already-locked situation. Then at this moment, the
frequency difference between fi and fo is called Captured range (refer to fig 4-5). LM 565 Captured range is
fc
2 f L
3.6 10 3 C 2
................................................... (8-4)
Figure 4-5 Lock range and capture range diagram

Implementation of FM demodulator by Using LM 565PLL
Figure 4-4 is the circuit diagram of LM 565 Phase locked loop., we can use this circuit as a FM demodulator.
When the input signal frequency increases, then the output signal voltage decreases. However, when the
input signal frequency decreases, the output signal voltage will increase, therefore, we can utilize the
relationship between the voltage of PLL and frequency to design the FM demodulator.
LM565 phase detector and VCO are designed in the IC package, this VCO and LM566 are the same. The
free running frequency f 0 of VCO is decided by the external C2 and VR1. The low-pass filter is comprised
by the internal resistor R3 at pin 7 and external capacitor C3. The objective of capacitor C4, which is
connected between pins 7 and 8 is to reduce the paracitic oscillation.
Figure 4-6 block diagram of FM to AM frequency discriminator.

Basic characteristics of PLL MC 4046
1. Free running frequency
Figure 4-6 is a MC4046 phase locked loop circuit diagram. From figure 4-6, when input terminal does not
input any signal, the output signal frequency of VCO is called free running frequency. Where C 2 is timing
- 18 -
capacitor, VR1 is timing variable resistor, the free-running frequency ( f 0 ) of MC4046 is decided by C 2 and
VR1.
Free running frequency : f 0
1
VR1C 2
............................................ 4-5
(2) Locked Range

When phase- locked loop circuit is at already locked situation, assume the input signal frequency (f i) slowly
move away from fo, when fi reaches at a certain frequency, the PLL will leave the locked situation. At this
moment, the maximum frequency deference for frequency fi and fo is called locked- range (refer to figure 45). The locked-range of MC4046 is
2fL = fmax fmin............................................................................( 4-6)
(3) Captured Range
At the beginning, PLL is at not locked situation, and then let the input signal frequency f i slowly move close
to fo, when fi reaches at a certain frequency, PLL will be at already-locked situation. Then at this moment, the
frequency difference between fi and fo is called captured range (refer to Figure 4-5). The captured range of
MC4046 is
2 fc
2 f L
..................................(4-7)
R1 R2 C1
Figure 4-6 Circuit diagram of MC4046 PLL.

Implementation of FM demodulator by Using MC 4046 PLL
Figure 4-6 is the circuit diagram of MC4046 phase-locked loop, which its functions are similar to LM565,
we can use this circuit as a FM demodulator. When the input signal frequency increases, then the output
signal voltage decrease. However, when the input signal frequency decreases, the output signal voltage will
increase, therefore, we can utilize the relationship between the voltage of PLL and frequency to design the
FM demodulator.
As a result of the demodulated audio signal consists of noise signal, therefore, we utilize the low- pass filter
in figure 4-7 to remove all the unwanted signal. Capacitors C 1, C2, resistors R1, R2, R3 , R4 and A741
comprise an active low-pass filter . This structure is a voltage controlled voltage source (VCVS) low- pass
filter. The expression of the gain is
Av 1
R4
............................... (4-8)
R1
Cutoff frequency is
fo
1
2
R2 R3 C1C 2
................................................ (4-9)
- 19 -
If R2 = R3 = R and C1 = C2 = C, then
fo
1
.................................................................... (4-10)
2RC
Figure Circuit diagram of second order active low pass filter.

3.
Equipments Needed
ETEK module LM565 demodulator
ETEK module MC4046 demodulator
SIgnal source (Function generator)
Oscilloscope
4.
Procedure understanding basic characteristics of LM565
a.
b.
Let J2 be short circuit and J3 be open circuit, i.e. C 2 = 100 nF. Let J1 be open circuit, i.e. SW 1
be open circuit.
c.
Adjust the variable resistor VR1, then measure the maximum (f0h) and minimum (fol) free
running frequencies (refer to figure 4-5) at the VCO output port (VCO O/P). Then record the
d.
Adjust the variable resistor VR1 so that the free-running frequency of VCO O/P (fo) is 2 KHz.
e.
Let J1 short circuit, and at the input port, input 0.25V amplitude and 2 KHz square wave
frequency.
f.
By using oscilloscope, observe on the demodulated output port (Audio O/P). Slightly increase
the input signal frequency unit the output signal frequency of Audio O/P is unable to lock
input signal. Then record the signal frequency fLH at this moment in table 4-1.
g.
Readjust the input signal frequency to the free running frequency (f o) of PLL. Then decrease
the input signal frequency until the output signal frequency of Audio O/P is unable to lock
input signal. Then record the input signal frequency fL1 at this moment in table 4-1.
h.
By using equation fL = (fLh fL1)/2, then calculate the locked range.
i.
Increase the input signal frequency so that the output signal frequency of Audio O/P is unable
to lock the in put signal. Then slightly decrease the input signal frequency until the audio O/P
- 20 -
locks the input signal. Then observe on the input signal frequency f Ch and record the measured
j.
Decrease the input signal frequency so that the output signal frequency of Audio O/P is unable
to lock the input signal. Then slightly increase the input signal frequency until the Audio O/P
locks the input signal. Then observe on the input signal frequency f C1 and record the
k.
By using equation fC = (fCh fC1)/2, then calculate the captured range.
l.
Let J1 be open circuit, J3 be short circuit and J2 be open circuit, which means that C 2 changes
to C5, i.e. 100nF changes to 10 nF, then repeat step 3.
m.
Adjust the variable resistor VR1, so that the free running frequency (f o) of the VCO O/P is 20
KHz. Let J1 be short circuit and at the input terminal, input 0.25 V amplitude and 20 Khz
frequency square wave frequency, then repeat step 6 to step 11.
Table 4-1 Measured results of the basic characteristics of LM565 PLL.
C2
fo
100
nF
KHz
Free running
frequency range
Locked range fL
foh
fLh
fL1
fCh
Hz
Hz
Hz
Hz
fol
Hz
20
10
KHz
nF
fL = _________Hz
Hz
Hz
Hz
Captured range fC
fC1
Hz
fL =_______-Hz
Hz
Hz
Hz
fL = _________Hz
fL =_______-Hz
Procedure understanding voltage and frequency conversion of LM565

a.
b.
Let J2 be short circuit and J3 be open circuit, i.e. C 2 = 100 nF. Let J1 be open circuit and
adjust the variable resistor VR1 so that the free running frequency (fo) of VCO O/P is 2 KHz.
c.
Let J1 be open circuit, i. e. SW1 be open circuit.
d.
At the demodulated FM input port (FM I/P), input 0.25 amplitude and 2 KHz square wave
frequency. Then measure the voltage of Audio O/P and record the measured results in table 8-2.
e.
Change the input signal frequencies to 0.5 KHz, 1KHz, 1.5KHz, 2KHz, 2.5KHz, 3KHz, 3.5
KHz. Then measure the voltage of Audio O/P and record the measured results in table 8-2.
f.
Sketch the characteristics diagram with voltage versus frequency in figure 8-8.
g.
Let J3 be short circuit and J2 be open circuit, which means that C 2 changes to C5, i.e. 100 nF
changes to 10nF.
h.
Let J1 be open circuit and adjust the variable resistor VR1, so that the free running frequency
(fo) of the VCO O/P is 20 KHz.
i.
Let J1 be open circuit, i .e. SW1 be open circuit.

- 21 -
j.
At the FM I/P, input 0.25 V amplitude and 20 KHz square wave frequency. Then measure the
voltage of Audio O/P and record the measured results in table 4-3.
k.
Change the input signal frequencies to 16.5 KHz, 17.5KHz, 18.5KHz, 20.5KHz, 21.5KHz,
22.5KHz, 23.5 KHz. Then measure the voltage of Audio O/P and record the measured results in table 4-3
l.
Sketch the characteristics diagram with voltage versus frequency in figure 4-9.
Table 4-2 Measured results of the voltage and frequency conversion characteristics of LM565 PLL. (V m =
0.25V, fo = 2KHz, C2 = 100 nF)
Input signal frequencies (KHz)
0.5
1.0
1.5
2.0
2.5
3.0
3.5
Output voltage (V)
Figure 4-8 Characteristics curve of voltage versus frequency.

Table 4-3 Measured results of the voltage and frequency conversion characteristics of LM565 PLL. (V m =
0.25V, fo = 20KHz, C2 = 10 nF)
Input
16.5
signal
frequencies
17.5
18.5
20
(KHz)
Output
voltage
(V)
- 22 -
21.5
22.5
23.5
Figure 4-9 Characteristics curve of voltage versus frequency.

6.
Procedure for LM 565 FM demodulator
1. Refer to the circuit diagram in figure 4-6 or figure ACS7-2 on ETEK ACS 3000-04 module to
produce the demodulated FM signal as the signal source. Let J1 be short circuit, i.e. the circuit is
the FM modulator. J3 be short circuit and J2 be open circuit, i.e. the selected capacitor is C4 = 10
nF. Adjust variable resister VR1 so that the frequency at the modulated FM output port (FM O/P)
is 20 kHz square wave.
b. Refer to the circuit diagram in figure 4-4 or figure ACS8-1 on ETEK ASC-300-04 module. Let J3
be short circuit, J1 and J2 be open circuit, i.e. C 5 = 10 F. Adjust the variable resistor VR1, so that
the free running frequency (fo) of the VCO O/P is 20 kHz.
c. Connect the output port (FM O/P) of the VCO LM566 to the input port (FM I/P) of the PLL
LM565.
d. At the audio input port (Audio I/P) of the VCO LM566, input 250 mv amplitude and 1 kHz sine
wave frequency. By using oscilloscope, observe on the output signal waveforms of the
demodulated FM signal (Audio O/P) at PLL LM565. Then record teh measured results in table 44.
e. According to the input signals in table 4-4, repeat step 4 and record the measured results in table
4-4.
f.
table 4-5
Table 4-4 Measured results of input and output signal waveforms of PLL frequency demodulator. (Vm = 250
mV, f0 = 20 KHz)
Audio signal frequencies
1 KHz
2 KHz
FM I/P
Audio O/P
3 KHz
mV, f0 = 20 KHz)
1 KHz
2 KHz
FM I/P
Audio O/P
3 KHz
7. Procedure for MC 4046 FM Demodulator

1. Refer to the circuit diagram in figure 3-5 or figure ACS7-1 on ETEK ACS 3000-04 module to
produce the demodulated FM signal as the signal source. Adjust variable resister VR1 so that the
frequency at the modulated FM output port (FM O/P) is 20 kHz square wave.
2. Refer to the circuit diagram in figure 4-6 or figure ACS8-2 on ETEK ACS 3000-04 module.
Adjust the free running frequency (fo) of the VCO output port (TP1) be 20 kHz.
3. Connect the output port (FM O/P) of the VCO MC 4046 to the input port (FM I/P) of the PLL MC
4046.
4. At the audio input port (Audio I/P) of the VCO MC4046, input 250 mV amplitude and 1 kHz sine
wave frequency. By using oscilloscope, observe on the output signal waveforms of the
- 23 -
demodulated FM signal (Audio O/P) at PLL MC4046. Then record the measured results in table 46
5. According to the input signal in table 4-6, repeat step 4 and record the measured results in table 4-6.
6. According to the input signal in table 4-7, repeat step 4 and record the measured results in table 4-7.
Table 4-6 Measured results of the input and output signal waveforms of FM to AM conversion frequency
demodulator. (Vm = 250 mV, f0 = 20 KHz)
FM I/P
TP2
LPF IN
Audio O/P
1 KHz
2 KHz
mV, f0 = 20 KHz)
1 KHz
2 KHz
FM I/P
Audio O/P
3 KHz
8.
Problem discussion
a. From the measured results of the basic characteristics experiment of LM 565 PLL, when the input
signal frequency moves away from the frequency locked range, what is the oscillation frequency of
the VCO?
b. For LM565 PLL, compare the locked range and the captured range.
c. In figure 4-4, what are the functions to capacitor C 3? If let C3 change from 0.1 F, what are the
changes of the pin 7 of LM565?
d. In the LM565 frequency demodulator experiment, if the output signal passes through the first order
low-pass filter, the is the output signal flatter than the pervious one? Try to design the low-pass filter.
e. How to use the PL circuit and the logic circuit to comprise a doubler frequency circuit?

Communication Theory Lab, EECE 310
Experiment No: 5
Name of the experiment: DSB-SC and SSB Modulator.
1.
Objectives:
- 24 -
a.
b.
c.
d.
2.
To understand the operation theory of double sideband suppressed carrier (DSB-SC)

modulator and single sideband (SSB) modulator.
To understand the waveforms and frequency spectrum of DSB-SC and SSB modulators.
To design and implement the DSB-SC and SSB modulators.
To understand the measurement and adjustment of DSB-SC and SSB modulators.
Theory:
The operation theory of DSB-SC and SSB Modulator
Figure 5-1 shows the waveforms of the amplitude modulation (AM). Let the audio signal be Am cos 2f m t
and carrier signal be Ac cos 2f c t , then the amplitude modulation can be expressed as

X AM ADC Am Cos 2f m f Ac cos 2f c

......................................................... 5.1
ADC Ac 1 m cos 2f m t cos 2f c t
Figure 5-1 Signal waveform of amplitude modulation.

where
m Am / ADC .
ADC : DC signal magnitude.
Am : Audio signal amplitude.

Ac : Carrier signal amplitude.
fm : Audio signal frequency.
fc : Carrier signal frequency
m : Modulation index or depth of modulation.
We can rewrite equation (5-1) as
X AM (t )
1
ADC AC m cos 2 f c f m t cos 2 f c f m t ADC AC cos 2f c t
2
....... 5-2
The first term represents the double sideband signals; the second term represents the carrier signal. From
equation (5-2), we can sketch the frequency spectrum of amplitude modulation as shown in figure 5-2 (a).
Since the AM signal is hidden in the double sidebands and the carrier signal does not contain any signal,
therefore the power is consumed in carrier during transmission of amplitude modulation signal. The double
sideband suppressed carrier (DSB- SC) modulation means the term ADC Ac cos 2f c t equals to zero,
therefore, it can suppress the carrier signal and only left the double sideband. We can use the DSB-SC
modulation to obtain the SSB modulation, We utilize two DSB-SC modulators and let the phase difference
- 25 -
between the two audio signals and carrier signals be 90 degree, i.e. (DSB-SC) Q and (DSB-SC)I, as shown in
equation (5-3) and (5-4).
( DSB SC ) P cos 2 f c f m t cos 2 f c f m t.................(5 3)
( DSB SC ) Q cos 2 f c f m t cos 2 f c f m t.................(5 4)
Equations (5-3) and (5-4) show that both (DSB-SC)Q and (DSB-SC)I signals connect to the adder, the we
can obtain USSB or LSSB signal at the output port.
X LSSB DSB SC I ( DSb SC ) Q
Cos 2 ( f c f m )t ...........................(5-5)
X LSSB DSB SC I ( DSb SC ) Q
Cos 2 ( f c f m )t ...........................(5-6)
Figure 5-2 (a) is the frequency spectrum of AM signal. We can see that the frequency spectrum
consists of three kinds of signals, which are f c-fm, fc and fc+fm. The output voltage of fc is higher than the
other two9 signals, therefore, the carrier signal does not contain any signal, and the power is consumed in
carrier during transmission of amplitude modulation signal. Figure 5-2 (b) is the frequency spectrum of
DSB-SC signals which are fc-fm and fc+fm. These two kinds of signals consists of the transmission signal,
therefore, by using this type of modulation, the power will not consume in the carrier. Besides, as a result of
the audio signal is hidden in the double sideband, so, the stronger the double sideband signal, the
transmission efficiency will be better.
From equation (5-2), we notice that the larger the modulation index, the better the transmission efficiency.
Generally, the modulation index is smaller or equal to 1. If the modulation index is greater than 1, we call
this situation as over modulation. Figure 5-2 (c) and figure 5-2 (d) are the frequency spectrum of SSB signal.
We can see that the frequency spectrum consists of either fc-fm signal or fc+fm signal. Therefore, during
transmission, the power consumption of SSB modulation is less than DSB-DC modulation. From the abovementioned discussion, we know that the sequence of power consumption of the three different types of
modulation is AM> DSB SC>SSB.
- 26 -
Fig 5-2 Different frequency spectrums of AM modulation.
- 27 -
Figure 5-3 Block diagram of DSB-SC modulation.
Fig. 5-3 Block Diagram of DSB-SC modulation

Implementation if DSB-SC Modulator
DSB-SC modulation is a kind of AM modulation, therefore, we can utilize the structure of AM modulator to
implement the DSB-SC modulator. Figure 5-3 is the block diagram of DSB-SC modulator. We utilize
balanced modulator MC1496 to design the DSB-SC modulated signal. Figure 5-4 is the internal circui9t
diagram of MC 1496, where D1, R1, R2, R3, Q7 and Q8 comprise an electric source, which can supply DC bias
current for Q5 and Q6. Q5 and Q6 comprise a differential combination to drive the dual differential amplifiers
constructed by Q1, Q2, Q3 and Q4. Pin 1 and 4 are the inputs of audio signal; Pin 8 and 10 are the inputs of
carrier signal. The resistor between pins 2 and 3 controls the gain of the balanced modulator; the resistor of
pin 5 determines the magitude of bias current for amplifier.
Figure 5-4 Internal circuit diagram of MC1496.

Figure 5-5 is the circuit diagram of AM modulator. We can see that the carrier signal and audio signal belong
to single ended input. The carrier signal is inputted from pin 10 and the audio signal is inputted from pin 1.
Therefore R8 determine the gain of the whole circuit and R9 determine the magnitude of bias current. If we
adjust the variable resistor VR1 or change the input amplitude of audio signal, then we can control the
percentage modulation of amplitude modulation, which means we can adjust the output become the DSB-SC
modulation. By adjusting variable resistor VR2, we can control the magnitude of the output amplitude, which
is also the gain.
- 28 -
Figure 5-5 Circuit diagram of DSB-SC modulation by utilizing MC1496.

Implementation of SSB Modulator
From equations (5-5) and (5-6), we know that the SSB modulator is the combination of two DSB-SC
modulators. Figure 5-6 is the block diagram of SSB modulator, where the phase difference of each audio
signal and carrier signal of the two DSB-SC modulators is 90 degree (i,e. 90 degree phase difference
between TP1 andTP2, and 90 degree phase difference between TP3 and TP4). In figure 5-6, the block if the
quadrature phase shift and the phase shift represent the phase shifter. The circuit diagram of phase shifter is
shown in figure 5-7. By adjusting the variable resistor, we can control the phase difference between the input
and output phase. The circuits of balanced modulator 1 and balanced modulator 2 are similar to the circuit
diagram in figure 5-5.
Then the output terminals of the two balanced modulators, which are TP5 and TP6 will be added by the
linearity adder, then we can obtain the modulated SSB signal. The circuit diagram of the linearity adder is
shown in figure 5-8. We utilize oscilloscope or spectrum analyzer to observe TP5 and adjust VR 1 of the
balanced modulator 1, so that the output is the modulated DSB-SC signal. By adjusting the variable resistor
BR2, we can control the gain of the DSB-SC modulator so that the output amplitude is maximum without
distortion. Similarly, we utilize oscilloscope or spectrum analyzer to observe TP6 and adjust VR 1 of the
balanced modulator 2, so that the output is the modulated DSB-SC signal. By adjusting the variable resistor
VR2, we can c0ontrol the gain of the DSB-SC modulator so that the output amplitude is maximum without
distortion. Finally, we use the spectrum analyzer to observe the output signal terminal is whether the
modulated SSB signal. If the frequency spectrum is not true, we can adjust the variable resistor of the
quadrature phase shift.
Figure 5-6 Block diagram of SSB modulator.
- 29 -
Figure 5-7 Circuit diagram of phase shifter.
Figure 5-8 Circuit diagram of linearity adder.

3.
Equipments needed
a.
b.
c.
d.
4.
Oscilloscope
ETEK module.
Spectrum analyzer
Procedure DSB-SC modulator

a.
b.
c.
d.
e.
f.
g.
h.
i.
j.
k.
To implement a DSB-SC modulator as shown in figure 5-5 or refer to figure ACS5-1 on

ETEK ACS-3000-03 module.
At the audio signal input port (Audio I/P), input a 300 mV amplitude and 1 KHz sine wave
frequency. Next at the carrier signal input port (Carrier I/P) input a 300 mV amplitude and
100KHz sine wave frequency.
By using oscilloscope, observe on both the audio signal output ports TP1 and TP2 at the same
time. Next adjust variable resistor QPS so that the phase difference between TP1 and TP2
is 900. Then record the measured results in table 5-1. By using oscilloscope, observe on both
the carrier signal output portsTP3 and TP4 at the same time. Next adjust variable resistor
Phase adjust so that the phase difference between TP3 and TP4 is 90 0. Then record the
By using oscilloscope, observe on the output signal waveforms of DSB-SC Q modulation
output port (TP5). Next adjust variable resistor VR1(gain adjustment) so that the output
amplitude is maximum without distortion, and also adjust variable resistor VR3 (modulation
index adjustment) so that the center level of upper peak and lower peak are 0V or the
modulation index is 100%. Finally, record the measured results in table 5-2.
Change the oscilloscope to spectrum analyzer, observe on the output signal waveforms of
TP5 and record the measured results in table 5-2.
By using oscilloscope, observe on the output signal waveforms of DSB-SC 1 modulation
output port (TP6). Next adjust variable resistor VR2 (gain adjustment) so that the output
amplitude is maximum without distortion, and also adjust variable resistor VR4 (modulation
index adjustment) so that the center level of upper peak and lower peak are 0 V or the
Change the oscilloscope to spectrum analyzer, observe on the output signal waveforms of
TP6 and record the measured results in table 5-3.
table 5-4.
According to the input signals in table 5-4, repeat steps 4 and 5, then record the measured
According to the input signals in table 5-4, repeat steps 6 and 7, then observe on TP6 and the
DSB-SC1 output port (DSB-SC O/P). Finally record the measured results in table 5-6.
table 5-7.
- 30 -
l.
According to the input signals in table 5-7, repeat steps 4 and 5, then record the measured
m.
According to the input signals in table 5-7, repeat steps 6 and 7, then observe on TP6 and the
DSB-SC1 output port (DSB-SC O/P). Finally, record the measured results in table 5-9.
Table 5-1 Measured results of phase adjustment.
(Audio I/P VP = 300 mV, f = 1 KHz; Carrier I/P VP = 300 mV, f = 100 KHz)
TP1
and
TP2
TP3
and
TP4
Table 5-2 Measured results of modulated DSB-SC signal (TP5).
Oscilloscope
Spectrum
analyzer
Oscilloscope
Spectrum
analyzer
TP1
and
TP2
TP3
and
TP4
Oscilloscope
Spectrum
analyzer
Oscilloscope
- 31 -
Spectrum
analyzer
TP1
and
TP2
TP3
and
TP4
Oscilloscope
Spectrum
analyzer
Oscilloscope
Spectrum
analyzer
5.
Procedure - SSB Modulator:

a.
b.
c.
d.
e.
f.
g.
h.
i.
To implement a SSB modulator as shown in figure 5-6 or refer to figure ACS5-1 on ETEK
ACS-3000-03 module.
At the audio signal input port (Audio I/P), input a 300 mV amplitude and 1 KHz sine wave
frequency. Next at the carrier signal input port (Carrier input), input a 300mV amplitude and
200 KHz sine wave frequency.
By using oscilloscope, observe on both the audio signal output ports TP1 and TP2 at the same
time. Next adjust variable resistor QPS so that the phase difference between TP1 and TP2 is
900. Then record the measured results in table 5-10. By using oscilloscope, observe on both the
carrier signal output ports TP3 and TP4 at the same time. Next adjust variable resistor Phase
adjust so that the phase difference between TP3 and TP4 is 900. Then record the measured
output port (TP5). Next adjust variable resistor VR1(gain adjustment) so that the output
amplitude is maximum without distortion, and also adjust variable resistor VR 3 (modulation
index adjustment) so that the centre level of upper peak and lower peak are 0 V or the
Change the oscilloscope to spectrum analyzer, observe on the output signal waveforms of TP5
and record the measured results in table 5-11.
By using oscilloscope, observe on the output signal waveforms of DSB-SC 1 modulation
amplitude is maximum without distortion, and also adjust variable resistor VR 4 (modulation
index adjustment) so that the center level of upper peak and lower peak are 0 V or the
Change the oscilloscope to spectrum analyzer, observe on the output signal waveforms of TP6
and record the measured results in table 5-12.
By using oscilloscope, observe on the output signal waveforms of SSB modulation output
port (SSB O/P), then record the measured results in table 5-13.
By using spectrum analyzer, observe on the output signal waveforms of SSB modulation
output port (SSB O/P), then record the measured results in table 5-13.
- 32 -

TP1
and
TP2
TP3
and
TP4
Oscilloscope
Spectrum
analyzer

Oscilloscope
Spectrum
analyzer
Table 5-13 Measured results of SSB modulation signal (SSB O/P).
Oscilloscope
Spectrum
analyzer
6.
Problem discussion
a.
b.
c.
d.
Explain the definitions of DSB-SC modulation and SSB modulation.

Explain the reasons that why the audio signal and the carrier signal need phase shifter to
produce the orthogonal signal.
Explain the advantages and disadvantages of DSB-SC modulation and SSB modulation.
Explain the output signal waveform of SSB O/P, if the phase difference of DSB-SC Q and
DSB-SC1 is same. (Refer to the measured results from oscilloscope and spectrum analyzer in
table 5-13).

Communication Theory EECE-310
Experiment No: 6
Name of the experiment: DSB-SC and SSB Demodulators.
1.
Objectives:
- 33 -
2.
a.
To understand the operation theory of double sideband suppressed carrier (DSB-SC)

demodulator and single sideband (SSB) demodulator.
b.
To design and implement the DSB-SC and SSB demodulators.
c.
To measure and adjust the DSB-SC and SSB demodulators.
Theory:
The Operation Theory of DSB-SC and SSB Demodulator:
When the modulated signal is recovered to the original audio signal, the procedure is known as
demodulation. In this chapter, we will discuss the operation theory of DSB SC and SSB demodulator.
Assume that a DSB SC signal is X DSSC t m t cos 2f C t , where m(t) represents the audio signal or
the low frequency signal. If this signal is multiplied by 2 cos 2f c t , then we get
y D kX DSSC t 2 cos 2f c t
= m t cos 2f c t 2 cos 2f c t .................................................... 6-1

= 0.5m t cos 4f c t 1
By using Fourier Transform on equation (6-1), we can rewrite the expression as
YD 0.5M f 0.5 f 2 f c f 2 f c f
.................. 6-2
= 0.5M f 0.25 M f 2 f c M f 2 f c
When YD f pass though a low-pass filter, which its frequency bandwidth equals or greater than the
frequency bandwidth of m(t), but smaller than 2 f c , then the only term left in equation (6-2) is
X D f 0.5M f ....................................... 6-3
By using Fourier Transform on equation , we get

X D t 0.5m t ................................................... 6-4
From equations, we know that the synchronous demodulator in figure can recover the m(t) signal from the
DSB SC signal.
Figure 6-1 Block diagram of synchronous demodulator.
On the other hand, we consider the demodulator 0(t) between the carrier signals of the demodulator and
modulator, then this situation will cause the signal distortion and the demodulator is unable to recover the
original audio signal.
y D t kx DSSC t 2 cos 2f c t t
= km t cos 2f c t 2 cos 2f c t t
................................................... (6-5)
= 0.5m t cos 4f c t t cos t
- 34 -
When y D f pass through a low-pass filter, which its frequency bandwidth equals or greater than the
frequency bandwidth of m(t), but smaller than 2 f c , then we get
X D t 0.5m t cos t ............................................................ (6-6)

In equation if the phase difference t is a constant, then it will cause attenuation on the amplitude.
However, if the phase difference t is a time domain function, then the signal will critically distort and
unable to recover to the original audio signal.
As for the SSB signal, it can be divided into upper SSB signal and lower SSB signal. The expressions are
as follow
X USSB t DSB p DSBQ
= 2 cos 2 f c f m t ............................................................ 6-7

or
X LSSB t DSB P DSBQ
= 2 cos 2 f c f m t .................................................. 6-8
where,
DSB P k cos 2f m t cos 2fLc t
DSBQ k sin 2f m t sin 2fLc t
f c : frequency of carrier signal.
f m : frequency of audio signal.
k : gain of the multiplier of mixer.

If equations (6-7) and (6-8) is multiplied 2 cos 2f c t , then we get
y DU t kxUSSB t 2 cos 2f C t
0.5 2 cos 2 f C f m t 2 cos 2f C t ................................................ 6-9
cos 2f m t cos 2 2 f C f m t
or
y DL t kxLSSB t 2 cos 2f C t
0.5 2 cos 2 f C f m t 2 cos 2f C t ................................................ 6-10
cos 2f m t cos 2 2 f C f m t
When y DU (t ) or y DL (t ) pass through a low pass filter, which its frequency bandwidth equals or greater
than the frequency bandwidth of m(t), but smaller than 2 f c , then we get
x D (t ) cos(2f m t )
.......................................................... (6-11)
From equations (6-7) to (6-11), we know that the synchronous demodulator in figure 6-2 can recover the
m(t) signal from the SSB signal.
On the other hand, if we consider the phase difference (t ) between the carrier signals of the demodulator
and modulator, then this situation will cause the signal distortion and the demodulator is unable to recover
the original audio signal.
- 35 -
y DU t kxUSSB t 2 cos 2f C t t
0.5 2 cos 2 f C f m t 2 cos 2f C t t
cos 2f mt (t ) cos 2 2 f C f m t (t )
.......................... 6-12
or
y DL t kx LSSB t 2 cos 2f C t t
0.5 2 cos 2 f C f m t 2 cos 2f C t t
....................... 6-13
cos 2f m t (t ) cos 2 2 f C f m t (t )
Therefore, when y DU (t ) pass through the low-pass filter, then we get
Figure 6-2 Block diagram of synchronous demodulator.

x D (t ) cos( 2f m t (t ))
cos 2f m t cos (t ) sin 2f m t sin (t )
............................ 6-14
Or when y DU (t ) pass through the low pass filter, then we get

x D (t ) cos(2f m t (t ))
cos 2f m t cos (t ) sin 2f m t sin (t )
....................... 6-15
From equations 6-14 and 6-15, we know that if the phase difference between the carrier signals of the
demodulator and modulator equals to each other, then x D (t ) cos(2f m t ) . This situation indicates that the
audio signal can be recovered. If the phase difference is not zero, then we noticed that the demodulated
signal will distort and unable to recover to the original audio signal.
6-2 Implementation of DSB-SC and SBSC Demodulator:
From the above mention discussion, we utilize the balanced modulator to implement the DSB-SC and SSB
synchronous detectors, Assume that x AM (t ) be the modulated DSB-SC and SSB signal, xc (t ) be the
carrier signal, then
x AM (t ) ADC 1 m cos(2f m t ) Ac cos( 2f c t ) ............................... 6-16
xc (t ) Ac cos( 2f c t ) ...................................................... 6-17
When these two signal input into two differential ports of balanced modulator, then the output signal of the
balanced modulator is as follow
- 36 -
xout (t ) kxc (t ) x AM (t )
kADC Ac 1 m cos 2f m t cos 2 2f c t

2
kA A
kA A
kA A
DC c DC c m cos 2f m t DC c 1 m cos 2f m t cos 2 2f c t
2
2
2
.............. .... 6-18
Where k represents the gain of the balanced modulator. The first term is the DC signal, second term is the
audio signal and third term is the second harmonic of modulated AM signal. If we can take out the second
term from xout (t ) , then we can obtain the exact demodulated DSB-SC and SSB signals or audio signal.
Figure 6-3 is the circuit diagram of synchronous product detector. Variable resistor VR1 controls the
input magnitude of carrier signal; variable resistor VR2 controls the input magnitude of modulated AM
signal; then the output signal of MC1496 is located at pin 12. Capacitors C7, C9 and resistor R9 comprise a
low pass filter which can remove the unwanted third term of equation 6-18, i.e. second harmonic of
modulated AM signal. Since the active low pass filter provides gain, so, the objective of the low pass filter is
to prevent attenuation on the output signal due to the RC circuit. The DC signal, which is the first term of
equation 6-18, can be blocked by C10. Therefore the signal that we obtain at output port will be
2
xout (t )
kADC Ac
m cos 2f m t
2
................................................... 6-19
Equation 6-19 represents the audio signal or in others words the original modulated AM signal can be taken
out via product detector.
Figure 6-3 Circuit diagram of synchronous product detector.
3.
Equipments Needed
a.
b.
c.
4.
ETEK ACS-3000-03 module

Oscilloscope
Procedure DSB-SC Demodulator:

a.
To implement a DSB-SC modulator as shown in figure 5-5 or refer to figure ACS5-1on ETEK
ACS-3000-03 module to produce the modulated DSB-SC signal source.
- 37 -
b.
To implement a product detector of DSB-SC demodulator as shown in figure 6-3 or refer to

figure ACS6-1 on ETEK ACS-3000-03 module. Then let J1 be short circuit and J2 be open
circuit.
c.
At the audio signal input port (Audio I/P) in figure ACS5-1, input a 300 mV amplitude and
1KHz sine wave frequency. Then at the carrier signal input port (Carrier I/P) in figure ACS51, input a 300mV amplitude and 200 KHz sine wave frequency.
d.
By using oscilloscope, observe on the both the audio signal output ports TP1 and TP2 in
figure ACS5-1 at the same time. Next adjust variable resistor QPS so that the phase
difference between TP1 and TP2 is 90 0. Then by using oscilloscope, observe on both the
carrier signal output ports TP3
and TP4 in figure 5-1 at the same time. Next adjust
variable resistor Phase adjust so that the phase difference between TP3 and TP4 is 900.
e.

amplitude of the carrier signal is maximum without distortion, and also adjust variable resistor
VR3 (modulation index adjustment) so that the canter level of upper peak and lower peak are
0V or the modulation index is 100%. By using oscilloscope again, observe on the output
signal waveforms of DSB-SC1 modulation output port (TP6). Next adjust variable resistor
VR2 (gain adjustment) so that the output amplitude of the carrier signal is maximum without
distortion, and also adjust variable resistor VR4 (modulation index adjustment) so that the
center level of upper peak and lower peak are 0V or the modulation index is 100%.
f.
Connect the modulated DSB-SC1 signal (DSB-SC1 O/P) in figure ACS5 1to the input
terminal (DSB-SC/SSB I/P) of the product detector in figure ACS6-1. At the same time, input
the same carrier signal in figure ACS5-1 to the carrier signal input port (Carrier I/P) in figure
ACS6-1.
g.
By using oscilloscope, observe on the output signal waveforms of the product detector (Audio
O/P) in figure ACS6-1. Next adjust variable resistors VR 1 and VR2, so that the output
amplitude is maximum without distortion. Finally, record the output signal waveforms of the
product detector TP1, TP2 and the demodulated signal (Audio O/P) in table 6-1.
h.
Let J1 be open circuit and J2 be short circuit. Then repeat step 7 and record the measured
Table 6-1 Measured results of DSB-SC demodulator.
(J1 be short circuit, J2 be open circuit)
TP1
TP2
Audio
O/P
Table 6-2 Measured results of DSB-SC demodulator.
(J1 be open circuit, J2 be short circuit)
TP1
TP2
Audio
O/P
5.
Procedure SSB Demodulator

a.
To implement a SSB modulator as shown in figure 5-6 or refer to figure ACS5-1 on ETEK
ACS-3000-03 module to produce the modulated SSB signal source.
- 38 -
b.
c.
d.
e.
f.
g.
h.
To implement a product detector of SSB demodulator as shown in figure 6-3 or refer to figure
ACS6-1 on ETEK ACS-3000-03 module. Then let J1 be short circuit and J2 be open circuit.
At the audio signal input port (Audio I/P) in figure ACS5-1, input a 300 mV amplitude and 2
KHz sine wave frequency. Then at the carrier signal input port (Carrier I/P) in figure ACS5-1,
input a 300 mV amplitude and 200 KHz sine wave frequency.
By using oscilloscope, observe on both the audio signal output ports TP1 and TP2 in figure
ACS5-1 at the same time. Next adjust variable resistor QPS so that the phase difference
between TP1 and TP2 is 900. Then by using oscilloscope, observe on both the carrier signal
output ports TP3 and TP4 in figure 5-1 at the same time. Next adjust variable resistor Phase
adjust so that the phase difference between TP3 and TP4 is 900.
By using oscilloscope, observe on the output signal wave forms of DSB-SCQ modulation
output port (TP5). Next adjust variable resistor VR 1 (Gain adjustment) so that the output
amplitude of the carrier signal is maximum without distortion, and also adjust variable resistor
VR3 (modulation index adjustment) so that the center level of upper peak and lower peak are
0 V or the modulation index is100%. By using oscilloscope again, observe on the output
signal wave forms of DSB-SC 1 modulation output port (TP6). Next adjust variable resistor
VR2 (gain adjustment) so that the output amplitude of the carrier signal is maximum without
distortion, and also adjust variable resistor VR4 (modulation index adjustment) so that the
center level of upper peak and lower peak are 0 V or the modulation index is 100%.
Connect the modulated SSB Signal (SSB O/P) in figure ACS5-1 to the input terminal (DSBSC/SSB I/P) of the product detector in figure ACS6-1. At the same time input the same carrier
signal in figure ACS5-1 to the carrier signal input port (carrier I/P) in figureACS6-1.
By using oscilloscope, observe on the output signal waveforms of the product detector (Audio
O/P) in figure ACS6-1. Next adjust variable resistors VR 1 and VR2, so that the output
amplitude is maximum amplitude is maximum without distortion. Finally, record the output
signal waveforms of the product detector TP1, TP2 and the de4modulated signal (Audio O/P)
in table 6-3.
Let J1 be open circuit and J2 be short circuit. Then repeat step 7 and record the measured the
Table 6-3 Measured results of SSB demodulator.
(J1 be short circuit, J2 be open circuit)
TP1
TP2
Audio
O/P
Table 6-4 Measured results of SSB demodulator.
(J1 be open circuit, J2 be short circuit)
TP1
TP2
Audio
O/P
6.
Problem Discussion:
a.
b.
c.
Explain the demodulation of the DSB-SC and SSB signals.

If the phase difference of the carrier signal in the synchronized demodulator and the carrier
signal in the modulator is different, then explain what will the results be.
Explain the functions of low-pass filter in the synchronized demodulator. And also explain
what will the output waveform be, if the low-pass filter is neglected.
MILITARY INSTITUTE OF SCIENCE AND TECHNOLOGY (MIST)

DEPARTMENT OF ELECTRICAL AND ELECTRONIC ENGINEERING,
MIRPUR CANTONMENT, DHAKA -1216, BANGLADESH.
- 39 -
COURSE NO.: EECE 310 Communication Theory Lab

Experiment No. 7 Delta Modulation and Demodulation
Objectives:
The main objectives of this experiment are
1. To study signal digitization using Delta Modulator and reconstruction using Delta Demodulator.
2. To observe the effect of Slope overload problem and its remedy.
Equipment needed:
1.
2.
3.
4.
5.
MODICOM 4 Board
IC Power 60 unit
Set of connection leads
Multi meter
Oscilloscope
Theory:
The three voice digitization techniques are:
Pulse Code Modulation (PCM)

Adaptive Pulse Code Modulation (ADPCM)
Delta Modulation (DM)
The operation of the delta modulator is as follows: The input signal is applied to a comparator, whose output
will be high if the non inverting input is greater than the inverting input, and low in the opposite case. The
output of this comparator is applied to a bi stable type D, which at the same time, will provide us an output in
each clock cycle that will be the data output.
This data output is also applied to the input of the level changer, whose mission is to convert the high level
into a continuous voltage of -4 volts, and the low level into +4 volts. These outputs are inserted into the input
of an inverter integrator that generates an increasing slope if its input is of -4 volts, and a decreasing one in
the other case. The output of the integrator is applied to the inverting input of the comparator, closing the
loop of the circuit.
To better understand the operation of the modulator, consider the figure given below.
- 40 -
Suppose the time instant t=0. In that moment the analogical signal input is more positive than the output of
the integrator, this originates a high level at the output of the comparator, which at the same time generates a
high level at the output of the bistable, which will be synchronized with clock signal. In other words, the
modulator is transmitting a bit 1.
The high level, at the same time, is the input of the level changer, which will give a continuous signal of -4 V
at its output, which will be converted into an increasing slope at the output of the inverter integrator, which
will be the inverting input of the comparator at the time t=1.
In t=1, the signal originates from the inverter is more positive than the analogical input signal. This will give
rise to a low level at the output of the comparator. This output will also be present at the output of the
bistable. In this case the following bit transmitted will be 0.
This low level, on being applied to the level changer, cause an opposite effect to that previously described,
giving rise this time to a decreasing slope that will be applied to the comparator. The process continues for
the rest of the time instant as can be seen in figure given in the previous page.
In this figure we can also visualize the effect that the overload error would produce. The signal generated by
the integrator has a constant increasing and decreasing slope, this implies that if the analogical signal varied
very quickly, the modulator would not be able to follow it, giving rise to an overload error.
The delta demodulator receives data flow, each of which will pass in each clock cycle through the bistable D.
From the output of the bistable, each bit will enter a circuit level changer identical to that of the modulator,
that is to say, it will generate a continuous value of -4 V if the bit that arrives is a 1, and a value of +4 V if the
bit that arrives is a 0.
The output of the level changer is applied to an inverter integrator, also identical to that of the modulator,
which will produce a signal equal to that of the modulator but with different amplitude and a certain delay.
The output of the integrator is then filtered through a low-pass filter that will give us the original signal as
output.
Procedure:
1. Adjust the transmitter and receiver level changer.
Transmitter Level Changer Setup Procedure:
Connect the transmitter clock output tp2 to the clock input of the D-type Bi stable tp174.
Connect the voltage comparator output tp8 to the data input on the D-type Bi stable tp16.
On the voltage comparator connect the inverting input tp6 to 0v and the non-inverting input tp7 to
the Bipolar output of the level changer tp19.
Use your oscilloscope to monitor the level changer output at tp19.
Carefully turn the level adjust preset until the monitored waveform is symmetrical about the 0v
level.
- 41 -
Receiver Level Changer Setup Procedure:
Connect the receiver clock output tp3 to the clock input of the D-type Bi stable tp32.
Connect the voltage comparator output tp8 to the data input on the D-type Bi stable tp31.
Connect the voltage comparator output tp8 to the level changer bipolar output tp34.
Use your oscilloscope to monitor the level changer output at tp34.
Carefully adjust the level adjust preset until the monitored waveform is symmetrical about the 0-v
level.
2. Carry out the assembly specified in figure given in next page.

3. Set the switches A and B of the clock generation block in the position 00, which is equivalent to the
frequency of 32 KHz.
4. Set the gain switch of the integrator blocks in the left position. In the same manner, set the switches A
and B of both integrators in the position.
A=0, B=0.
5. Set the potentiometer of the signals of 250 Hz, 500Hz, 1KHz and 2KHz so that their output will be
maximum.
6. Set the console switch, or that of the power supply being used, in the position ON.
7. ADJUSTMENT OF THE MODULATOR
It should be adjusted using a continuous level of 0V, therefore, adjust the DC potentiometer until it gives
a level of 0V.
Connect this signal of 0 V to the positive input of the comparator.
Using an oscilloscope, visualize the existing signal in the output of the transmitter integrator. This must
be a triangular signal centered on 0V. In the event of not having a stable trace in the oscilloscope, move
the potentiometer located in the level changer of the transmitter. The amplitude of this triangular signal
must be of about 0.5 volts peak to peak. Once this has been done, the transmitter is ready.
ADJUSTMENT OF THE DEMODULATOR
Visualize the signal at the output of the receiver integrator. This signal must be the same as that of the
transmitter and ideally centered on 0V. If this is not so, try to adjust it using the potentiometer of the level
changer of the receiver and of the integrator itself. This last is located in the upper part of the board. Once
this has been done, the demodulator is adjusted.
1. Once the adjustment has been done, disconnect the continuous signal from the positive points of the
comparator and in their place we will connect the sinusoidal signal of 250 Hz.
2. Visualize the signal at the output of the transmitter integrator. Note how the integrator follows the
analogical signal. For a stable trace in the oscilloscope, it may be necessary to readjust the
potentiometer of the level changer of the transmitter.
3. Visualize the signal at P19 and observe that, in fact a data flow is being transmitted.
4. Visualize together the signal at the output of both integrators. Note that these signals are very similar
and that of the receiver present a slight delay.
5. See the signal at the output of the receive filter. Note that the recovered signal mat be slightly fuzzy.
This fuzziness is due to a quantification error produced by the size of the integrator step. Increasing
the frequency of the clock can reduce the error.
6. Change the frequency of the clock using the switches A and B of the clock generation block as
indicated on the board itself and observe the previous signals. See how the fuzziness has reduced and
the transmission has improved.
7. Reset the switches A and B of the clock generation block in the position A=0 and B=0.
8. Remove the signal of 250 Hz from the positive input of the comparator and in its place, put the signal
of 2 KHZ.
9. Visualize the signal at the output of the integrator. You will observe that we have triangular signals.
This means that the integrator is not following the analogical signal; therefore an overload error has
been produced.
10. Increase the clock frequency of the system and observe that the error still exists. Leave the system in
the clock frequency of 256 KHZ and the switches A and B in the positions A=1, B=1.
11. In a delta modulator-demodulator system there are three possible ways to solve the overload error:
Reduction of the input signal frequency
Reduction of the input signal amplitude
Increase in the integrator gain
- 42 -
12. The first case has already been studied. With the signal of 250 Hz there were no overload problems,
while with that of 2 KHz this error always appeared. Now, study the other two possibilities.
13. Reduce the amplitude of the signal of 2 KHz while visualizing the output of the transmitter integrator.
Observe how, when the amplitude is reduced, the integrator is already capable of following the input
signal.
14. The other way is to increase the integrator gain. For this, set the switches A and B of both integrator
blocks in the position A=1 and B=1. Note how the overload error has disappeared. The gain of the
integrator operates in the following way: there are four positions according to the position of A and
B(B being the less significant bit). Each time we pass from one position to the next gain is doubled.
15. Test the assembly with the two remaining signals and draw conclusions.
Report:
1. Draw and submit all the wave shapes, you observe at different test points, with your report. Discuss
the nature of the wave shapes. (Wave shapes should be drawn on graph papers)
2. What are the advantages of DM over PCM, DPCM and ADPCM?
3. Brief discuss slope overload noise and granular noise occurred in DM.
4. DM is basically a 1 bit DPCM-explain this statement.
1. How can the S/N (signal to quantization noise ratio) performance of DM system be improved?
Reference:
1. MODICOM 4 manual
2. Modern Digital and Analog Communication Systems ------by B. P. Lathi
3. Communication System----- by Haykin
4. Modern Communication Systems Principles and Applications -----by Leon W. Couch II
Figure: Detail Connection of Delta Modulation and Demodulation
- 43 -

Experiment No.8 T.D.M Modulation and Demodulation
OBJECTIVE:
The objective of this experiment is to gain practical introduction to time division multiplexing (TDM)
through the following topic areas.
Introduction to time division multiplexing (TDM)

The TDM transmitter
Synchronization
The Demultiplexer and Demodulator
The complete PAM-TDM system
Equipments needed:
1.
2.
3.
4.
5.
MODICOM 2 board.
IC Power 60 unit
Set of connection leads.
Multimeter.
Oscilloscope.
Theory:
The sampling theorem provides the basis for transmitting the information contained in a band limited signal
m(t) as a sequence of samples of m(t) taken uniformly at a rate that is usually slightly higher than the
Nyquist rate. An important feature of the sampling process is the conservation of time. That is, the
transmission of the sample engages the communication channel for only a fraction of the sampling interval
on a periodic basis, and in this way some of time interval between adjacent samples is cleared for use by
other independent message sources on time shared basis. We, thereby, obtain a Time Division Multiplex
(TDM) system, which enables the joint utilization of a communication channel by a plurality of independent
message sources without mutual interference among them.
The information can be sent in the amplitude, duration or position of pulses,
This will give rise to:
1. A PAM signal (Pulse Amplitude Modulation).
2. A PWM signal (Pulse Width Modulation).
3. A PPM signal (Pulse Position Modulation).
The Time Division Multiplexing (TDM) technique, in general, consists of the simultaneous sending of
several signals through only one medium of transmission. What is actually done is to intercalate different
signal samples within a single sampling interval (the time that elapses between two consecutive samples of
the same signal), and to transmit the set as a single signal through one medium of transmission. In order to
recuperate all the signals properly, the receiver must separate the different samples that reach it in the correct
order and direct each one of them to its assigned output channel for its recovery.
So there must be correct synchronization between the commutator (an electronic switch, in the transmitter
side) and the decommutator (an electronic switch, in the receiver side). This synchronization is essential for a
satisfactory operation of the system. See the figure below:
- 44 -
Procedure:
We can achieve complete PAM-TDM system by using three modes of connection.
Mode 1: Three links between Transmitter and Receiver:
A simple method of TDM transmission is to provide three connections between the sending and receiving
end. One channel for message, another for clock and another for frame synchronization.
Connections:
Tx output to Rx input
Tx CH0 to Rx CH0
Tx Clock to Rx Clock
See figure 1 for connections.
Mode 2: Two links between Transmitter and Receiver:
Message signal and frame synchronizing signal is transmitted and clock is generated at receiver by PLL.
Connections:
Tx CH 0 to PLL I/P
Sync of PLL to RX CH 0
CLK of PLL to Rx Clock
Mode 3: One link between Transmitter and Receiver:
TDM can be achieved using only one link and so achieve significant transmission medium savings.
Connections:
Sync Level to CH 0
Sync of PLL to Rx CH0
CLK of PLL to Rx Clock
Put duty cycle ON position by rotating the knob. Keep at around 20%. You can observe the sampling
clock signal at P7, P8, P9 and P10. Observe the differences of the clock signals at these points.
Check that all switched faults are in OFF position.
Operate the TDM by Mode 1 connection. Figure 1 gives the basic Mode 1 connection diagram.
Observe the wave shapes at TX output at P20 and before and after low pass filtering at P42 and P43
respectively for CH0. Observe P44 and P45 for CH1, P46 and P47 for CH2, P48 and P49 for CH3.
Repeat the above steps using Mode 2 connection. Figure 2 gives the basic connection diagram. To
change the amplitude of the input signal very the knob associated with the input signal clockwise.
Repeat the above steps using Mode 3 connection. Figure 3 gives the basic connection diagram.
Report:
1.
2.
3.
4.
5.
Show the wave shapes, observed in the experiment, in graph papers with corresponding pin numbers.
What do you mean by TDM modulation and demodulation?
Why synchronization is necessary in the receiver circuit?
What is the major advantage of TDM over FDM?
Describe the operation of PLL. What is the function of divided by 4 counter and delay in the PLL
of this Trainer?
Reference:
1. MODICOM 2 Manual
2. Modern Digital and Analog Communication Systems
3. Communication Systems --- Haykin
- 45 -
--- B.P. Lathi
Figure 1: Connection of Mode 1 TDM
Figure 2: Connection of Mode 3 TDM
- 46 -

.
Experiment No. 9 ASK Modulation and Demodulation
Objectives:
The main objectives of this experiment are:
To know the technique of digital amplitude modulation.
To study the characteristics of a signal modulated in amplitude.
Required Equipments:
1.
2.
3.
4.
5.
MODICOM 5 Board
IC Power 60 unit
Multimeter
Oscilloscope
Theory:
In transmission by radio frequency, the data cannot be transmitted directly. In such cases, a high frequency
carrier signal must be used. The simplest way to modulate a carrier with a data is to change the carrier signal
amplitude each instance the data changes. The technique is called amplitude Shift Keying (ASK).
The simplest form of carrying out this modulation is to make the transmitter emit the carrier signal as long as
the bit to be transmitted is a 1 and to suppress it totally when this is a 0, as can be seen in the following
figure.
To generate an ASK signal, the transmitter uses a balanced modulator. This device simply multiplies the two
signals at its inputs. The output voltage at any moment is the product of two input voltages. One of the
inputs will be the carrier signal and the other is the data. See the following figure:
In the receiver the circuitry required to demodulate a signal of this type is minimal. The simplest method is
to rectify the ASK signal and then to filter it. The signal is then passed through a voltage comparator and thus
the output data is obtained. See the following figure:
Modulator signal (Data): Adjusting the MODICOM 3/1 with the DC1, we should obtain the data:
D6D0
1 0 1 0 1 0 1
The signal we shall use as modulator is that corresponding to the NRZ (L) codification of the said data.
PART A: ASK MODULATION
- 47 -
Procedure:
1. Carry out the assembly indicated in the following figure.
2. Visualize the digital modulator in the oscilloscope. Draw the signal and indicate the following
characteristics: Vp-p, voltage level associated with logic 1 and logic 0.
3. Visualize the carrier in the oscilloscope. Draw the signal and indicate the following
characteristics: Vp-p, frequency.
4. Vary the position of GAIN adjustment fully clockwise and the CARRIER OFFSET and the
MODULATOR OFFSET adjustments at the center point and draw the modulated signal.
5. Repeat the step adjusting CARRIER OFFSET fully clockwise and the other two at the center
point and draw the modulated signal.
6. Again repeat the step adjusting MODULATOR OFFSET fully clockwise and the other two at the
center point and draw the modulator signal.
7. Using the three adjustments, try to obtain a modulated signal that is as close as possible to the
ideal signal. Draw the modulated signal.
8. Indicate the following characteristics of the modulated signal: Amplitude and Frequency of the
signal corresponding to logic 1 and logic 0 (A1, f1 and A0, f0).
PART B: ASK DEMODULATION
Procedure:
1. Draw the signal at the following points:
At the input of the MODICOM 5/2.
At the output of the rectifier.
At the output of the low pass filter.
At the output of the DATA SQUARING circuit.
2.
Check that the data generated in the MODICOM 3/1 are recovered in
MODICOM 3/2. For this, set the SYNC switches to ON.
Report:
1. Submit all the wave shapes you observed at different test points, with your report. Discuss the
nature of the wave shapes at different output levels. (Wave shapes should be plotted on graph
paper.)
2. What do you mean by ASK modulation of a carrier with unipolar and polar binary data?
3. Briefly discuss the coherent and non-coherent detection of the ASK modulated signal. Which
method is advantageous when considering the noise effect?
4. When it is required to obtain the lowest probability of bit error when the input ASK signal is
corrupted by Additive White Gaussian Noise (AWGN), what should be performed?
5. What is the function of a Phase Locked Loop (PLL) circuit in product (coherent) detection?
References:
1.
2.
3.
4.
MODICOM 5 manual.
Modern Digital and Analog Communication Systems. -- B. P. Lathi
Communication Systems. -- Simon Haykin
Modern Communication Systems- Principles and Applications. -- Leon W. Couch II
- 48 -

COURSE NO.: EECE 310 COMMUNICATION THEORY LAB
Experiment No. 10 FSK Modulation and Demodulation
Objectives:
The main objectives of this experiment are:
1. To know the technique of digital frequency modulation.
2. To study the characteristics of a signal modulated in frequency in the transmitter and
its demodulation in the receiver.
Required Equipments:
MODICOM 5 Board
IC Power 60 unit
Multi meter
Oscilloscope
Theory:
In FSK, the output of the transmitter changes continually from one frequency to another each time there is a
level change in the data signal modulator. If the higher frequency is used to represent the data 1, then the
lower frequency for the data 0.
The generation of an FSK signal in the transmitter can be done by two ASK generators. One of them
generates the ASK signal with the carrier of the highest frequency and the data to transmit as modulator, and
in the other, ASK signal is formed with the smaller frequency carrier and, an inverted signal of the data to
transmit as modulator.
The block diagram for generating an FSK signal is represented in the next page:
In the receiver, the FSK signal is decoded by means of a PLL detector. The PLL detects the frequency
Changes of the FSK signal and provides an output voltage proportional to the frequency of the input signal.
The block diagram required to demodulate an FSK signal in the receiver is represented in the following
figure: Modulator Signal (Data):
Adjusting the MODICOM with the DC1, we should obtain the data:
D6..D0
1 0 1 0 1 0 1
The signal we shall use as modulator is that corresponding to the NRZ (L) codification of the said
data.
- 49 -
PART A: FSK MODULATION

Procedure:
1. Carry out the assembly indicated in the following figure.
2. Using the adjustments GAIN, CARRIER OFFSET and MODULATION OFFSET of the modulators
1 and 2 to generate a signal modulated in FSK of amplitude 2 volts (p-p) that is as close as possible to
the ideal. Draw the modulated signal.
PART B: FSK DEMODULATION
1. Draw the signal at the following points:
At the input of the MODICOM 5/2.
AT the output of the FSK demodulator (PLL).
At the output of the low pass filter.
AT the output of the DATA SQUARING circuit.
2. Carry out the necessary assembly between MODICOM 5/2 and MODICOM 3/2 to recover the digital
data signal.
3. Check that the data generated in the MODICOM3/1 are recovered in MODICOM 3/2. For this, set
the SYNC CODE GENERATOR switch of the MODICOM 3/1 to ON.
Report:
1. Submit all the wave shapes you observed at different test points, with your report. Discuss the nature
of the wave shapes at different output levels. (Wave shapes should be plotted on graph paper)
2. What do you mean by FSK modulation and demodulation?
3. Briefly discuss the coherent and non-coherent detection of the FSK modulated signal. Which method
is advantageous when considering the noise effect?
4. When it is to be required to obtain the lowest probability of bit error when AWGN corrupts the
received FSK signal, what should be performed?
5. What is the function of a Low Pass Filter (LPF) on coherent detection?
References:
1.
2.
3.
4.
MODICOM 5 manual.
Modern Digital and Analog Communication Systems. -- B. P. Lathi
Communication Systems. -- Simon Haykin
Modern Communication Systems- Principles and Applications. -- Leon W. Couch II
- 50 -

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Military Institute of Science and Technology

Department of Electrical Electronic & Communication Engineering

Figure 1.1 Signal waveform of amplitude modulation

Figure 1.2 Block diagram of AM modulator.

x AM (t ) ADC Am cos 2f m t Ac cos 2f c t

ADC : DC signal magnitude

Figure 1-3 Frequency spectrum signal of amplitude modulation signal.

Figure 1-4 Waveforms of the over modulation

Figure 1-5 Circuit diagram of transistor AM modulator.

Audio signal amplitudes

Audio signal amplitudes

Explain the objectives of the transistor Q1 in figure 1-5.

Military Institute of Science and Technology

Department of Electrical Electronic & Communication Engineering

Figure 2-1 Theory diagram of amplitude demodulator.

Figure 2-2 Block diagram of diode detector.

Figure 2-3 Circuit diagram of diode detector.

ETEK ACS 3000-02 module.

Audio Signal Amplitudes

Audio Signal Frequencies

Military Institute of Science and Technology

X FM t Ac cos t Ac cos 2f c t 2f fx d ..................... 3-1

Frequency deviation of FM x FM t is shown as below

f c f m cos 2f m t f c Am . f . cos 2f m t ............................... 3-3

If the FM signal is the largest amplitude and largest Frequency (i.e.

Am 1 and f m W )then the

Figure 3-1 Analog diagram of capacitance of varactor diode.

A: The PN junction area

Figure 3-2 Equivalent circuit diagram varactor diode.

Fig 7-3 block diagram of MC4046.

Figure 3-4 Circuit diagram of MC 4046 FM modulator.

Figure 3-5: Internal structure diagram of LM566

Figure 7-5 Internal Structure diagram of LM566.

Figure 3-6 Circuit diagram of LM566 FM modulator

Procedure for LM566 FM Modulator:

Describe the operation theory of FM modulation.

Military Institute of Science and Technology

Name of the Experiment: FM Demodulation

Phase detector (PD)

Low-pass filter (LPF)

Voltage controlled oscillator (VCO)

Fig 4-1 Blocked diagram of Phase locked loop.

Kd = The gain of phase detector (Volts/Radian)

Fig 4-2 Theory of phase detector

Fig 4-3 Theory of locked frequency

Closed loop gain : K L K d K a K o

Where Vc = Total voltage supply = Vcc Vcc 5 5 10V

Figure 4-4 LM565 phase locked loop.

(2) Locked Range

(3) Captured range

Figure 4-5 Lock range and capture range diagram

Figure 4-6 block diagram of FM to AM frequency discriminator.

(2) Locked Range

Figure 4-6 Circuit diagram of MC4046 PLL.

Figure Circuit diagram of second order active low pass filter.

Procedure understanding basic characteristics of LM565

By using equation fL = (fLh fL1)/2, then calculate the locked range.

By using equation fC = (fCh fC1)/2, then calculate the captured range.

Table 4-1 Measured results of the basic characteristics of LM565 PLL.