C O N T E N T S

ED-2950-1

INTRODUCTION ················································································································ 5

ASSIGNMENTS

(1) The Signal Source ······························································································· 9

(2) Tuned Circuits ···································································································· 23
(3) The Crystal ········································································································· 53
(4) The Amplifier ······································································································ 69
(5) Filters ···················································································································· 89
(6) Tuned Circuits ································································································· 107
(7) Simple Amplitude Modulator ······································································· 123
(8) Detection and Demodulation ······································································· 143
(9) Super-Heterodyne Radio ·············································································· 167
(10) Frequency Modulation ················································································· 181
(11) FM Detection ·································································································· 201

APPLICATION NOTE

[1] Oscillator ··········································································································· 229

[2] Effect of bandwidth on Transmission of Speech ······························· 235
[3] Display Method of Examining AM Waveforms ··································· 239
[4] Amplitude Stabilised Oscillator ································································· 242
[5] Demonstration of Automatic Frequency Control ································ 244

APPENDIX

(1) The Decibel ······································································································· 250

(2) Relation of Q to Bandwidth ········································································ 254
(3) Amplitude Modulation ···················································································· 265
INTRODUCTION

Why Communications?

Ever since man started to think he has wanted to communicate. The first
messages were quite simple; information about the presence of food; warning
of impending attack by hostile men or animals; words of affection to his mate.
At first he used his voice. Then he found that messages could be written
down. Secret codes were developed and information passed on by noting the
position of objects. The range of the human voice was limited by the power of
the lungs but written information could be sent over long distances and
eventually our postal systems grew from this. This was not instaneous; early
man lit bonfires on hilltops but these beacons could only convey one message
- yes or no. (The American Indians had a refinement of this system - smoke
signals!) Telegraph systems were developed and semaphore was one of the
codes used.

Modulation

All methods of communication convey information by MODULATING one or

more properties of a CARRIER. For example, when you use your voice to
communicate, the carrier is the intervening air and you use your vocal cords to
modulate this carrier by causing variations in pressure to make the sounds we
call speech. It is very important to notice that an unmodulated carrier conveys
no information. Try exhaling without using your vocal cords and see if you can
communicate with anyone!
This principle is the foundation of communications and consequently we will be
spending most of the time on practicals looking at the basic modulation. Once
you have mastered the principles it is then easy to see how they can be used
in any specific application.

Electrical Systems

Electrical signals can be modulated very easily and since they can travel
almost instantaneously over long distances, it is only natural that they are
employed in communication systems. Historically the signal was modulated
simply by switching it on and off. Morse codes was used in which letters were

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represented by varying the arrangement of on and off times. This is called an
interrupted wave system and is sometimes still used to transmit digital
information.
However to convey continuos signals such as voice or music, a continuos wave
system is required. If we consider a sinewave signal as a carrier
signal it can be represented by the equation v=Vsin(  t + ∅ ). Thus we
could vary either V,  or ∅ , giving voltage amplitude, frecquency and phase
modulation respectively (remember that the unmodulated carrier will convey
no information). We will study amplitude and frequency modulation in detail.
Another advantage of electrical signals is the ease with which they can be
amplified to compensate for losses in the equipment or the transmission
channel and thus we shall also study amplification.

Transmission Channels

Communication does not only mean radio. The subject is sometimes separated
into radio and line transmission but there are many more divisions. However,
as mentioned, modulation is common to them all. They differ only in the choice
of carrier medium. Radio channels use high frequency signals which propagate
easily.
Line transmission can use a wide range of carrier frequencies but the cables
and associated equipment must be suitably designed. Frequency dependent
circuits are use to select the correct frequency or range of frequencies of
interest and therefore examples of these will be covered in the first
assignments (tuned circuits, crystals, active and passive filters). The carrier
frequency chosen for this kit is 465kHz and the reasons for this are given in
Assignment 1.

Coverage

The paragraphs above should tell you why we place emphasis on the three
topics into which the kit is divided:

1. Signal Processing (amplifier + frequency dependent circuits)

2. Amplitude Modulation + Detection
3. Frequency Modulation + Detection

You are referred to earlier kits in the ED-2900 series range for coverage of

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components and circuits, basic transistor theory, and power supplies.
The following topics are deliberately not covered, for cost or technical
reasons:

Electromechanical Telephony
Radiation and propogation
Aerials
Multiplex systems
Pulse Modulation
Information Theory (cybernetics)

The ED-2900 series System

ED-2900 series consists of a powered student station and a number of plug-in

kits. Each comprises a fully documented assignment manual, together with the
necessary hardware to teach the basic principles of various technologies.
Further information on the first three kits covering Basic Electricity, Basic
Electronics and Transducers can be supplied on request. Information on the
various power supplies is contained in the ED-2900P operation and
maintenance manual. The full circuits and other information relevant to this
ED-2950 are contained in the Instructor's Manual. A check list of the hardware
and total auxiliary equipment requirement for ED-2950 follows.

ED-2950 List of Equipment

Hardware

Module Function
A Signal source and attenuator
B Amplifier
C Diode detecter and operational amplifier
D Balanced modulator
E Active filter
F Passive filter
G Diode ring bridge
H Tuned circuit (two units)
J Crystal tuning circuit
K Aerial

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Auxiliary Equipment Requirement

The following is the minimum specification of all the auxiliary equipment

required to perform all the assignments.

Oscilloscope

Bandwidth d.c 20 MHz, Timebase down to at least 500ms/cm.

Tow channels, calibrated, Timebase output or external X facility.
Long persistance (P7) screen or storage facilities.
External trigger facility.

Digital Frequency Counter

Maximum frequency to be measured 1 MHz, Input sensitivity to 10mV

Function Generator

Frequency range 0.1Hz-100kHz.

Waveforms - sine and square (triangle is also an advantage).
Output Voltage up to 10V pk-pk.

DC Voltmeter - digital is ideal

High input impdance, Max.voltage to be measured 15V d.c.

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ASSIGNMENT - 1
- THE SIGNAL SOURCE

Object
To introduce the signal source module.

Equipment required
● Power Console ED-2900P
Unit 0-4/0-20 volts d.c, variable, metered.
Unit +15 volts d.c.

● Oscilloscope, double beam 5MHz, calibrated.

● Digital Frequency Counter, 1MHz.
● Module Type 2950A

Approximate time required

One hour

Prerequisites
Introduction

Theory Assumed
Very basic knowledge of electricity and mathematics.

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Preliminary Procedure

1. Attach the console ED-2900P.

2. Connect the power supply unit(s) to the mains supply line.

DO NOT SWITCH ON YET .

3. Turn the variable d.c control knob to minimum.

4. If you are using any of modules 2950A, B, C, D or E, plug them into the
sockets on the console in alphabetical order. If any change is required, it
will be indicated in the assignment. The act of plugging in the modules
connects them to the power supply in the correct manner. All sockets are
identical.

5. This completes the preliminary procedure.

Experimental Procedure

You will have read in the introduction to this Manual that in order to
communicate information some form of modulated carrier must be used. This
carrier is often a high frequency alternating signal (a radio wave for example).

In order to carry out future assignments we need to have a suitable signal as

a module in our kit. The purpose of this assignment is, quite simply to
introduce this signal source module to you, so that you will be able to use it
in future assignments.

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 Fig 1.1

The layout of the module is shown in Fig 1.1. The heart of the module is
an integrated circuit voltage-controlled oscillator,type LM566. This procedures
both triangular and square waveforms although we only use the square-wave
output. You will see later that this is adequate for our practical work. The
frequency of operation is determined by the combination of a capacitor, chosen
to give the required frequency range, and a voltage V, which provides v.c
tuning. Two frequency ranges can be selected. By grounding socket D on the
module, an additional capacitor is connected in circuit to provide the other
range. The tuning voltage V is the sum of three signals A, B and C. Signals A
and B are external inputs with different sensitivities, each having a calibrated
potentiometer and C provides a manual frequency control. The square-wave
output of the integrated circuit is connected through an attenuator to the
output sockets on the right of the module. Let us now connect up the module
and see how it works.

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Practical 1.1
Calibration of Voltage-controlled Oscillator

Make sure that sockets A, B and D (see Fig 1.2) are not connected, and turn
the variable attenuator to zero (fully clockwise). Connect your oscilloscope to
the ground and OdB output sockets to the module. Set the timebase of the
oscilloscope to 1 s/cm and the Y amplifier to 0.5 volt/cm and switch on the
power supply. Adjust the oscilloscope trigger controls to provide a stable
display and then turn the manual frequency control C.

Fig 1.2

Q1.1 What do you see?

Q1.2 Estimate from the timebase of your oscilloscope the frequency range
available with the manual frequency control C between the settings 1
and 5.

At the lower end of this control you will notice that the circuit stops
oscillating. That is why we measure the lowest frequency available with the
control on setting 1. Now connect socket D to ground, reset your oscilloscope
timebase to 100 s/division and turn the manual frequency control C again.

Q1.3 What do you see now?

Q1.4 Estimate from the timebase of your oscilloscope the frequency range

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 available with the manual frequency control C between the settings 1
and 5.

Now connect the digital frequency counter to the module output and measure
these frequencies more accurately. In fact, measure the frequency at each
scale setting for both ranges, grounding socket D for the low range. Record
your readings in a table as in Fig 1.3. Note the frequency corresponding to the
dot, on the high range only.

Scale Frequency (kHz)

setting Low range High range

Fig 1.3

Exercise 1.1

Plot graphs in the same sheet of linear graph paper, of frequency against scale
setting for both ranges.

Q 1.5 What shapes are the graphs?

Q 1.6 Given that the scale on the manual frequency C corresponds to the
control voltage V, can you deduce from your graphs any
approximate relationship between the control voltage and the output
frequency?

The graph for the low range should be a straight line and the frequency in
kHz should correspond to the control voltage V. The graph for the high range

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is not linear. This is a function of the integrated circuit, but it is no
disadvantage since we have our calibration curve and we will always measure
the frequency with a digital frequency counter as we need it. The most used
frequency on the high range is 465kHz and the dot is a guide to the position
on the scale which gives this frequency.

Two external inputs A and B are provided, to enable the frequency to be

controlled by an external voltage. Both have different sensitivities, controlled
by potentiometers. Let us investigate briefily their operation.

Practical 1.2
Use of external frequency control inputs

Ground socket D (see Fig 1.4) so that you are on the lower frequency range.
With the frequency meter and oscilloscope still connected to the output of the
module, adjust the manual frequency control C until the frequency is 5kHz.

Fig 1.4

Make sure that the variable d.c control on the Console ED-2900P is set to
zero, and select the 0-4-volt range. Connect the Var. d.c socket on your
module to input A and set the A sensitivity to 1. Slowly increase the variable
d.c output on the Console up to +4 volts and notes the output frequency.

Q 1.7 Has the frequency increased or decreased and to what value?

Now adjust the A input sensitivity control and note the frequencies at a scale

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setting of 0.5.
Q 1.8 What is this frequency? Does a greater setting on the sensitivity
control affect the frequency to a greater or lesser extent?

Q 1.9 Given that the marking on controls A and B multipliers of input voltage
(+4 volts in this case), can you see any relationship between the
settings and the measured frequency?

In practical 1.1 the frequency of oscillation was varied by control C. This

control C is an internal potentiometer applying a voltage through the summing
junction to the VCF input of the IC. When also using controls A and B, the
VCF value is the difference between the value of C and that of the external
input voltage multiplied by the setting of the sensitivity controls. This
relationship is shown in Fig 1.4.

With +4 volts still on input A, reset the sensitivity control to 1 and disconnect
socket D so that you are on the high range. Note the output frequency from
your digital frequency counter.

Q 1.10 How does this compare with the frequency for scale setting 1 (5-4) as
determined from your calibration graph?

Now return the variable d.c control to zero and connect the variable d.c input
instead to the B input socket. Set the B input sensitivity control to 0.1.
Observe the frequency counter as you slowly increase the voltage to +4 volts.

Q 1.11 What is the frequency with +4 volts on input B (x 0.1)?

Q 1.12 Has the frequency increased or decreased and by a greater or lesser

amount than when you used socket A ? How does it compare with
that from your calibration curve for a control voltage of (scale setting
external input) i.e [5-(4 0.1)] ?

You will investigate these inputs more fully in future assignments.

For the moment, disconnect all inputs.

Now let us investigate the attenuator on the module output. You will note that
there is a variable attenuator calibrated from 0 to 20dB and a set sockets
marked 0,-6,-12,-18,-24,-30 and -36dB.

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An advantage of expressing the total gain or loss of several cascaded stages
in dB, is that the resulting values for the stages can then be summed.
Appendix 1 gives a full explanation of the decibel, showing how we express
the input and output Power and Voltage ratios in dB.

 
Gain or Loss in dB =      
 

The usefulness of this method will be seen as we progress through the

assignments.

Practical 1.3 Calibrating the attenuator

With your oscilloscope still connected to the 0dB output of the signal source
module, make sure that the variable attenuator is also set the 0dB position
(fully clockwise). Also ensure that socket A, B and D are disconnected. The
setting of the manual tuning control C should have no effect on your results,
so set it onto the dot. Adjust the oscilloscope for a stable display and measure
the peak-to-peak voltage from the oscilloscope screen. Record your reading in
a table as in Fig 1.5. Turn the variable attenuator to -3dB and again measure
the peak-to-peak voltage from the oscilloscope screen and record in your
table.

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 Attenuation Output volts    
      
dB pk-pk   
0 1.0 0 0
-3
-6
-9
-12
-15
-18
-21
-24
-27
-30
-33
-36
Fig 1.5

Q 1.13 How can you use the combination of the fixed attenuator sockets and
the variable attenuator to fill in all the rows of the table?

By setting to a fixed value and then setting the variable attenuator knob to
the difference in value needed, all the attenuator amounts required can be
obtained.

Now transfer your lead to the -6dB position and set the variable attenuator to
-6dB. Record the peak-to-peak voltage from your oscilloscope.

Q 1.14 How does this compare with the reading in the table for the -12dB
position?

Q 1.15 Does this confirm the theory that dB figures may be simply added
together when used in combination?

Always remember that 0dB here means zero attenuation and not zero output!

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Exercise 1.2

Taking the voltage at the 0dB position as the reference voltage Vo, calcurate
the ratio of all the other voltages V to this, i.e Vo V.
 
Then calculate    and 20   .
 

Insert your figures in the remaining columns of the table. The first line will
have a ratio of one, and hence logarithms of zero, therefore it has already
been filled in for you. Remember we are dealing with voltage ratios.

Q 1.16 Do the figures in the last column of your table compare with the dB
figures quoted in the first column? If not, can you explain the
discrepancy?

Since the dB is a logarithmic unit, the easiest way to represent this graphically
is to use logarithmic graph paper.

Plot a graph of attenuator setting (dB) against output voltage (pk-pk open
circuit) as measured from your oscilloscope. Use three-cycle logarithmic graph
paper with axes as shown in Fig 1.6. If you have no such paper, you can use
the logarithms of the voltages as in your table, but you should try to get some
now as it will be required for other experiments. A blank sheet is included at
the end of this manual.

Keep this table and graph carefully: you will need them in future assignments.

Fig 1.6

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Practical considerations & applications

This assignment has been largely one of familiarization with the signal source
module, so that you will be able to use it in future assignments. We use the
term 'signal source' rather than 'oscillator' since we are using an integrated
circuit as our function generator. 'Oscillator' may be loosely used to describe
any circuit producing regular oscillation of almost any shape, but it is important
that you should appreciate that there are many types of oscillator. Some of
these will be mentioned in Assignment 4.
However the method used in our module provides a satisfactory signal source
for our experiments. Several important points emerge from this assignment.

a) The output is a square wave

b) The frequency can be varied from about 200kHz - 1000kHz on the high
range and from about 1kHz - 5kHz on the lower range.

c) The frequency can be controlled by an externally applied voltage.

Positive voltages reduce the frequency from the scale setting. The two
inputs have different sensitivities.

d) the output attenuator is calibrated in dB.

Let us consider these points in order.

Square-wave Output

By an advanced piece of mathematics known as Fourier Analysis, ANY complex

repetitive waveform can be expressed as the sum of a series of sinusoidal
waveforms. It is then possible to see how a circuit treats each component
separately, then add the results together to see the effect on the complex
wave as a whole. If some components are attenuated or phase-shifted
differently to others, the total signal suffer distortion.

Our square wave contains components at odd multipules of the fundamental

frequency, e.g third, fifth, seventh, ninth, etc. These are called Harmonics.
Their relative amplitudes decrease in the same order. This can be written
mathematically as

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 Square wave amplitude

   
=  (sin  t +  sin 3 t +  sin 5 t +  sin 7 t + ▪▪▪▪)
   

Where V is the peak-to-peak voltage of the square wave and is the

fundamental frequency.

The addition of these waveforms is shown in Fig 1.7. Notice how the inclusion
of more terms improves the wave shape. A large number of terms must be
included for the shape to be nearly perfect. The sqaure waves produced by
the intergrated circuit in our module are not perfect and may consequently
contain some harmonics not indicated above. This represents distrotion.
However we will often use this waveform in conjunction with circuits which do
not respond to the harmonic frequencies and our signal will then appear more
sinusoidal. In any case it should be clear which is the fundamental component
of frequency. This circuit was chosen because it is simple and inexpensive to
build and as discussed, is the adequate for our experiments.

Frequency

You plotted calibration curves of frequency against manual control C setting.

This setting corresponds to the control voltage V. Now V consists of three
voltages summed together in an operational amplifier: the manual frequency
control C setting, minus input A, minus input B. The non-liner relationship on
the high range does not matter as we have a calibration curve.

The two external inputs A and B and as the minus sign indicates, a positive
voltage at these inputs, reduces the frequency from the manual control setting
by an amount depending upon the applied voltage and the sensitivity setting.
The relationships are marked on the module. You will see how these inputs
can be used as you progress through the assignments. One use will be to
sweep the frequency to provide frequency domain displays.

The high frequency range of 200-1000kHz was chosen for ED-2900 series in
preference to a higher RF frequency because of the nature of the system.
Special care is necessary at RF and the long unscreened interconnecting leads
and associated stray capacitances of the ED-2900 series system could have
upset the results. It is of course still a good idea to keep lead lengths to a
minimum and use screened instrument leads where possible, even within this

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frequency range. You should also keep your hands away from the modules
when taking readings.

Such frequencies are used for line transmission and for long-wave ratio.
The use of the low range will be made clear in later assignments.

Varying the signal source frequency is of course a method of modulation in its

own right and this will be fully covered in Assignment 6.

Attenuator

You have investigated the output attenuator with its variable control and set of
sockets calibrated in fixed 6dB steps. Note that 0dB here means zero
attenuation and not zero output. While the dB is essentially a power ratio, it
can also be applied to voltage and current ratios. Any discrepancy you found
from the indicated values in your practical work should only have been at low
levels where the environmental noise became apparent.

You will make use of this calibrated attenuator in later assignments to measure
the response of various circuits directly in dB. You will also see that it may
also be sometimes advantageous to use a logarithmic scale for frequency also.

Applications

As previously mentioned, carrier signals which are to be modulated for

transmission are often alternating signals. Our module provides a suitable
source of such signals for our experiments. The variable frequency will also be
used to determine the frequency response of circuits falling within its range.
As explained, the use of a square wave is no serious disadvantage.

There are of course other types of oscillator which could be used. Some will
be mentioned in later assignments, but you should also refer to your textbook.

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Square waveform showing fundamental and harmonics

Fig 1.7

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ASSIGNMENT - 2
- TUNED CIRCUITS

Object
To investigate the characteristics of passive tuned circuits containing
inductance and capacitance.

Equipment required
● Power Console ED-2900P
0.4 volts d.c variable, metered.
+15 volts d.c
-15 volts d.c
● Oscilloscope, double beam, 5MHz, calibrated.
● Digital Frequency Counter, 1MHz
● Modules Type 2950A and 2950E
● Tuned Circuit Assemblies Type 2950H: 2e

Approximate Time Required

Three hours

Prerequisite Assignments:
No.1

Theory Assumed
Simple introductions to reasonance and transformers.

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Preliminary Procedure

1. Attach the console ED-2900P.

2. Connect the power supply unit(s) to the mains supply line.
DO NOT SWITCH ON YET.
3. Turn the variable d.c control knob to minimum.
4. If you are using any of modules 2950A, B, C, D or E, plug them into the
sockets on the Console in alphabetical order.
If any change is required, it will be indicated in the assignment. The act of
plugging in the modules connects them to the power supply in the correct
manner. All sockets are identical.
5. This completes the preliminary procedure.

Experimental Procedure

You will already have studied how the reactance of inductors and capacitors
varies with frequency according to the formulae

XL = 

XC = 


These variations are shown in Fig 2.1.

If an inductor and capacitor are co-

nnected together, important things
happen in the region of the frequency
indicated as fO, where the reactances
are numerically equal but opposite in
sign, of phase,(reactance is a phasor
quantity). Fig 2.1

These topics were fully covered in ED-2900 series.

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Practical 2.1

Examine one of the two identical tuned module 2950H assemblies and its
circuit diagram shown in Fig 2.2

Note that there are two coils wound together on the former. They act as a
transformer and have a turns ratio of just over 7, as indicated on the diagram.
You will see the reasons for this as we progress. Both coils are centre tapped.
The primary is of low impedance and the secondary of high impedance. Across
the secondary coil are connected a fixed capacitor of 800pF in parallel with a
variable capacitor of (0-820)pF to enable the circuit to be turned.
Let us find out what this means.

Fig 2.2 (a) Fig 2.2(b)

Connect up as in Fig 2.3

On the module 2950A

Set the manual frequency control to position 1.
Set the variable attenuator control to 0dB.
Ensure that the A, B and D inputs are disconnected.
On the Module 2950H:
Set the variable capacitor to mid-position.

On the oscilloscope:

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 Set the timebase to 1 s/cm and the amplifier to 1V/cm

Adjust the controls for a stable display.

Slowly adjust the manual tuning control C on the signal source module from
end to end.

Fig 2.3

Q 2.1 Is the output from the circuit the same at all frequencies?

Find the setting at which the output is a maximum.

Q 2.2 We know that our signal source delivers a square wave. Why do you
think a sine wave is now displayed?

Q 2.3 Estimate from the scale setting and your calibration graph from
Assignment 1 the approximate frequency for this maximum output?

Check this against the oscilloscope timebase and the approximate calibaration
on the tuned circuit variable capacitor.
Adjust the variable capacitor.

Q 2.4 What happens to the output?

Q 2.5 From the approximate calibration on the variable capacitor, estimate
frequency range over which the circuit gives an output greater than
about 20% of its peak value.

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 At maximum output we say that the circuit is TUNED

Since the impedances of the components vary with frequency, we would expect
the output voltage to vary with frequency. Let us see if we can find the form
of this variation.

Practical 2.2

With your tuned circuit still connected as in practical 2.1.

On the module 2950A

Connect a lead from the Var.d.c output to the B input.

Set sensitivity control B to 0.1 maximum.
Set the variable attenuator control to -0dB.

On the oscilloscope:
Set the timebase to 1 s/cm and the amplifier to 1V/cm

On the console:
Set the Var.d.c to the +4V range
On the digital frequency counter.
Connect its inputs to the -6dB and ground outputs on the 2950A Module

Adjust the manual frequency control C on the 2950A module until the
frequency reads exactly 465kHz. This will give us a reference frequency in the
centre of the capacitor's range, about which to start. You will see the
relevance of this frequency in later assignments. With your oscilloscope still
set to 1V/cm, adjust the tuning capacitor until the output is a maximum. Read
the peak-to-peak output voltage from your oscilloscope.

Now slowly increase the variable d.c comtrol voltage to its maximum on the
0-4 volt range.
You remember that from assignment 1 this lowers the frequency.

Q 2.6 What happens to the output voltage as displayed on our oscilloscope?

- 27 -
Increase the sensitivity of your oscilloscope to 5mV/cm and note the shape of
the waveform.

Measure its peak-to-peak value as best you can and record the frequency
from the digital frequency counter.

Q 2.7 Why does the waveshape deteriorate at frequencies away from

the maximum?

Exercise 2.1

Calculate the ratio between the voltage at this frequency and the voltage at
465kHz, in dB, by the formula

   
ratio =    
   

This gives a measure of the 'magnification effect' of this circuit. You should
find you have a figure somewhere between 20 and 30dB corresponding to a
voltage magnification of between about 10 and 30. This magnification effect is
very important and we will return to it when you have plotted a graph of the
exact form of the variation. We are going to use the calibrated attenuator on
the signal source module to plot the frequency response of this tuned circuit,
directly in dB. To do this we must put in the maximum voltage when we have
minimum output and measure how mych we have to attenuate the input to give
the same output at other frequencies. In other words, we will use the
oscilloscope only as a voltage indicator, the measurements being taken directly
from the attenuator. The tuning capacitor must NOT be adjusted during this
practical.

Starting from the point where you left off, with the variable d.c voltage at +4
volts, corresponding to a low output at a low frequency, adjust the signal
source output attenuator in 3dB steps and, by resetting the variable d.c control,
record the frequency at which the peak-to-peak waveform displayed on the
screen is the same as that just measured. You may find it helpful to adjust the
position of the starting point slightly so that your trace fills a whole number of
divisions on your oscilloscope screen.

Record your readings in a table as in Fig 2.4. When the d.c control is at

- 28 -
minimum, you should find you have returned to the frequency of maximum
output. This may not correspond exactly with one of the 3dB steps, so record
the exact attenuator setting which gives the correct value.

Attenuator Frequency
setting (dB) kHz
0
3
6
9
12

Maximum

12
9
6
3
0
Fig 2.4

Q 2.8 Does this setting of the variable attenuator correspond to the value you
calculated in exercise 2.1?

In order to find out what happens at frequencies above that for maximum
output, we must set the oscillator frequency higher so that it can be reduced
by the application of our positive control voltage. Without touching the
attenuator setting turn the d.c control to maximum (+4 volts) and reset the
manual frequency control c until the frequency reads exactly 465kHz. Continue
your readings in 3dB steps, recording in table 2.4, until the variable d.c control
is at minimum. If you have not quite returned to the 0dB attenuator setting,
increase manual control C slightly for the last reading.

- 29 -
Exercise 2.2 (a)

Plot a graph of attenuator setting in dB against frequency. This corresponds to

the output of the circuit against frequency and is called the response curve of
the tuned circuit.

You have in effect plotted the voltage appearing across the secondary
terminals of the circuit of Fig 2.2, arising from an a.c input applied via the
mutual inductance between the primary and secondary.

At low frequencies, the reactance of the inductor is low compared with that of
the capacitor, consequently a larger current flows through the inductor, and the
voltage across the whole circuit is low. In the exteme, at zero frequency, the
inductor appears a short-circuit and no voltage can appear across it. At high
frequencies, the voltage is again very small since the reactance of the
capacitor is low. Again, in the extreme, at infinite frequency, the capacitor
appears a short-circuit and no voltage can appear across it.

In the frequency region between these two we have loosely called low and
high (the exact location depends on the values of L and C) the ratio of the
currents in the inductor and capacitor changes, thus the voltage across the
circuit changers and at the value where the reactances are numerically equal,
the currents in inductor and capacitor are equal and in opposite phase and tend
to cancel out. This phenomenon is known as RESONANCE.

Q 2.9 What is the resonant frequency from your graph?

The frequency at which resonance occurs is denoted by fo and can be found

by putting

  
 
           
     

This can also be expressed as


       



- 30 -
Exercise 2.2 (b)

Using the maximum and minimum values of capacitance and the value of the
inductance indicated on the circuit diagram Fig 2.2, calculate the minimum and
maximum resonant frequencies available with the tuning capacitor in the tuned
circuit by application of the above formula. Use it in reverse to calculate the
value of capacitance the resonates with 100 H at 465kHz.

Theoretically, at resonance, current could flow backwards and forwards

between the inductance and capacitance, no external current need be supplied
and the whole of the applied voltage would appear across the circuit. The
impedance would be infinite (open circuit). Obviously this state of affairs cannot
exist in practice.

Q 2.10 What limits the current?

It should be obvious that any practical inductor has a resistance due to the
wire from which it is wound. This is shown as r in Fig 2.5 (a). Note that r
cannot be separated from L. This causes losses in the circuit and means that
even at resonance some current must be supplied from the external circuit (in
our case via mutual inductance from the primary) to make up these losses.
This reduces the amount of voltage magnification available and gives the circuit
a finite maximum impedance at resonance. Let us further explore this.

Fig 2.5 (a) Fig 2.5 (b)

The 'magnification effect' you have noted on your graph is the ratio of the
magnitudes of the voltage appearing across the tuned circuit at resonance and
at frequencies away from resonance. Our experiment did not take us far

- 31 -
enough away from the resonant frequency to demonstrate the full effect of
this. At resonance, a large circulating current flows in the tuned circuit
consisting of IC through the capacitor and ILr through the inductor. These
currents are almost equal and opposite, as shown in the phasor diagram Fig
2.5 (b). the resultant is IX the current that needs to be supplied from the
external circuit to make up the losses in the resistance r.

The ratio of IX to IC or ILr at resonance is called the QUALITY FACTOR or Q

of the circuit. The higher the Q, the shaper the peak of the curve.

Q 2.11 Will a coil with a high resistance give the circuit a high or a low
value?

You should also be able to see from the phasor diagram Fig 2.5(b) that at
resonance IX is very nearly in phase with the applied voltage E. This means
that the impedance of the tuned circuit at resonance is resistive and of high
value since IX is small. Let us see if we can find an expression for this.

Fig 2.6(a) Fig 2.6(b)

Fig 2.6 sows a simplified view of our tuned circuit. Referring to Fig 2.6(b), the
circuit impedance is given by

     ∙     ∙  
 =  +  or Z =  = 
         

where ZS is the series impedance around the circuit ZLr, ZC.

At resonance, the two reatances cancel out and ZS = r.

- 32 -
Since r is assumed to be small, ZLr is approximately equal to XL and likewise
Zc=XC. This circuit is called a low loss circuit. We shall see later what
happens if this assumption is not justified.


   R = 
∴ Zresonance = so 




Since at resonance      and


XC = 
  

We have denoted this by R, since it is resistive. It is sometimes called the

dynamic resistance of the circuit. This also shows that the impedance at
resonace depends only on the component values and is independent of
frequency.

We can now derive an expression for Q.

   
We have defined Q as the ratio  = 
    

   ∘   

=
  ∩ 

Substituting the value we have just found for Zresonance

  
Q =  ÷  =   
    

 
Q =    Q =  
∴  or 

These results imply that if we vary the resonant frequency fo by changing the
value of the capacitance Q will vary. We will return to this.

- 33 -
 
A similar result is given if we define Q as 
 

In this case Q =  ÷  
 

 
Q =  Q = 
∴    or  

You have learnt that the higher the Q the sharper the peak of the curve. It
can be shown that Q is related to the resonant frequency fo and the bandwidth
by the formula


Q =
∆

The bandwidth 2 is taken as the frequency difference between the points on


the resonance curve where the voltage (or current) has dropped to  the

value at the peak. This corresponds closely to the -3dB points which are

usually used in practice. This value of  is chosen because it leads

toconvenient mathematical relationships, one of which, as we have seen, is that
it is voltage or current at the half power points. These half power points are
sometimes referred to as the upper half power point and lower half power
point as in Fig 2.7.

Fig 2.7

- 34 -
It follows that the band or range of frequencies passed by a tuned circuit with
a high Q (sharper peaked curve) is less than that for one with Q. This effect
is known as SELECTIVITY.

You can see that a low resistance coil will give a high value of Q. Also notice
that these results imply that the total circuit impedance or dynamic resistance
at resonance is Q times the reactance of either branch and you remember it is
purely resistive

i.e R = Q oL = 
 

It is difficult to measure Q by measuring the currents and difficult to calculate

since the value or r is difficult to measure. For r is not simply the d.c
resistance of the inductor since at the frequency we are using the current
tends to travel along the surface of the wires. This is known as the skin effect
and, together with other effects, increases the effective high frequency
resistance compared with the d.c resistance. Let us see therefore if we can
find a way of obtaining Q from your previous measurements.

Exercise 2.3

Mark on your graph the resonance and -3dB points and calculate the
bandwidth by subtracting the frequencies at these points.

Exercise 2.4

Calculate the Q value by dividing the resonant frequency by the bandwidth.

Express this in dB.

Q 2.12 How does this compare with your previously found value for Q?
Can you account for any discrepancy?

Q 2.13 Why is this not a very accurate computation of Q?

We will show you a way to get a more accurate result by looking at only the
peak of the curve with finer control of the signal source frequency.

- 35 -
Practical 2.3

This uses the same set up as practical 2.2, so if you have not done it already,
connect your tuned circuit to the signal source module as in Fig 2.3. In
addition, connect a wire between the var.d.c socket and input B on the signal
source module. Turn the variable d.c control and sensitivity control B both to
zero. Set the variable output attenuator to -3dB.

With your digital frequency counter connected to the -6dB output, adjust the
manual tuning control until the frequency is about 475kHz. Then turn the
variable d.c control to +4 volts and slowly increase the input B sensitivity
control until the frequency is about 455kHz. This now gives us a fine control
over about 10kHz of frequency either side of 465kHz. Adjust the variable d.c
until the frequency displayed is exactly 465.0 kHz, then tune the tuned circuit
to resonance by observing the output voltage on your oscilloscope. Note the
value of the peak-to-peak output voltage. Now turn the variable attenuator
back to the zero position and increase the variable d.c control until the output
voltage returns to the value you have just measured.

This frequency f1 corresponds to the lower half power point. Now reduce the
variable d.c until the output voltage increases through resonance and returns
once more to the -3dB value. This frequency f2 corresponds to the upper half
power point. Record these frequencies accurately from the digital frequency
counter in a table as in Fig 2.8.

Fig 2.8

You should recognize this as the same method you used when plotting the
response curve in practical 2.2, except that we now have finer control over
the frequency and we are only looking for the half power points (-3dB).

Exercise 2.5

Calculate the bandwidth between the half power points by subtraction f1 from
f2.
Calculate Q by dividing fO by the bandwidth. Express Q in dB.

- 36 -
Q 2.14 How does this compare with your previously found values for Q?
Can you account for any discrepancy?

Before we proceed, let us explain a method for automatically displaying on

your oscilloscope the resonance response curve you have just plotted in
Exercise 2.2. This method will also be widely used in future assignments.

You remember that the signal source is voltage controlled. If you take the
timebase output from your oscilloscope and use it as the control voltage for
the signal source, the frequency will vary as the timebase moves the trace
across the screen, as shown in Fig 2.9 (a) and (b). Connecting the signal
source to the circuit under test and its output to the oscilloscope will cause
the varying response to be displayed as in Fig 2.9 (c).

The oscilloscope timebase output should be d.c coupled. Its actual voltage
varies between oscilloscopes but with the sensitivity controls A and B and
the manual tuning control C, most oscilloscopes with timebase outputs up to 50
volts should be able to be accommodated. If your oscilloscope has no
timebase output, or proves to be unsuitable, you can use an auxiliary function
generator with a triangular wave output in the frequency range of about
1-10Hz, connected also to the X amplifier of your oscilloscope (sine outputs
can also be used but do not provide as clear a display).

- 37 -
 Fig 2.9

Practical 2.4

Leave your tuned circuit set up as in practicals 2.2 and 2.3 (see Fig 2.3) but:

On the module 2950A

Disconnect the connection from the variable d.c to input B.
Set the variable attenuator control 0dB.
Connect the timebase output of the oscilloscope to socket A.
Set the A sensitivity to zero.
Reset the manual control C to give a frequency counter reading of about
465kHz.

On the module 2950H:

Observing the tuned circuit output, tune it to resonance.

- 38 -
On the oscilloscope:
D.C couple the instrument.
Set the timebase to 10ms/cm and the amplifier to 1V/cm.

On the module 2950A

Slowly increase the A sensitivity control and the display should change
shape.
Adjust the manual tuning frequency control C until the peak occurs in
the centre of the screen. The display should be as in Fig 2.9(c). If
the A input is too sensitive, try connecting your oscilloscope timebase
to input B.
These settings depend upon the magnitude of your oscilloscope timebase
output, whether it is positive or negative going and whether it starts
from zero.

Q 2.15 What is the effect of manual frequency control C?

Q 2.16 What is the effect of the sensitivity control A or B?

You may find these two controls to be interdependent but try and set C until
the peak is at the centre of the screen. You may see some smaller peaks
appearing at the sides of the display. This is the tuned circuit responding to
the harmonics in the output of the signal source at other frequencies and they
can be ignored for the moment. Adjust the sensitivity control until these are
off the screen.

If you have a positive-going timebase, low frequency will be to the right of

your screen, the opposite way to your graph. If you have a nagative-going
timebase, low frequency will be on the left.

Temporarily increase the timebase speed and note how the display becomes
unsatisfactory. We could obtain an even better display if we could remove the
negatine half of this waveform and display only the upper envelope.

Q 2.17 What components could be used to do this?

A diode and capacitor are available on the detector module 2950E. Connect
them to the output of your circuit as shown in Fig 2.10. Do not include
resistor R at this stage. If you increase the Y gain to 0.5 volt/division,

- 39 -
 Fig 2.10

You should get a display as in Fig 2.9(d). Remember this method well. Not
only will we use it in later assignments to display frequency responses, but it
is also the basis of one method of demodulation. This will be fully covered in
Assignment 8. Adjust the controls on the signal source module until the peak is
in the centre of your oscilloscope screen.

Move the tuning capacitor in the tuned circuit.

Q 2.18 What happens to the displayed peak?

Q 2.19 Does it stay the same height?

Why do you think this is?

Q 2.20 Does it stay the same shape, i.e does Q alter?

As you change the value of the tuning capacitor, you change the capacitive
reactance, and consequently the voltage developed across it varies. This
accounts for the variation in height of the peak as you move it across the
screen. However, you should have noticed that as the height increased, the
bandwidth increased also, i.e the shape of the curve altered. This means that
the Q value did not vary much.
 
You remember one formula for Q was Q =  so if we vary o by

changing capacitor, you would expect Q to change. In practice, the
resistance r changes. You remember we said that you must consider r as the
high frequency resistance of the coil, not the d.c resistance, and its changes
with frequency often cancel out the variation of Q caused by the frequency
variation. If you want to demonstrate this, if you have time, you can easily
measure Q at frequencies of 400kHz and 500 kHz using the method of
practical 2.3. You will find that they are about the same value.

- 40 -
The value of Q depends upon the resistance of the inductor r. But there are
other resistances present in the circuit. These will lower the effective Q of the
circuit. This is called DAMPING. Such loads can be represented by a parallel
resistor R, Shown dotted in Fig 2.10

Q 2.21 What happens to the Q value?

Q 2.22 Does it shift the resonant frequency?

You should be in no doubt as to the reduction in Q value. Theory shows that

the frequency will shift slightly, but this may be very difficult to detect.

In fact we have assumed if this assignment that

resonance occurs at the

frequency where the reactances are equal i.e  o =   . However
 
resonance may also be defined as the point where the voltage is in phase with
the current, or where the impedance is a maximum. In a parallel circuit these
three conditions occur quite separately, as theory shows, but the difference
between the three frequencies is very small for Q values greater than about
10, which will be the case in our assignments.

It is therefore important, to achieve high values of Q, that the damping effect

of any loading resistor must be as small as possible. In a parallel tuned circuit
this is achieved by making the input and output impedances as high as
possible.

This is one of the advantages of using the mutual coupling between the
primary and secondary coils to provide the excitation to our tuned circuit. As
well as stepping up the voltage available, the transformer also reduces the
loading effect of the 600 source impedance of the signal source. If you have
studied transformers, you will know that it provides an impedance
transformation proportional to the square of the turns ratio. Our turns ratio is
4:29, so 600 impedance of the primary appears across the secondary

as about 600 x 
 

or 31.5k .

However this is still sufficient to damp the Q from that theoretically obtainable.
You will have noticed that even 100k provides some damping. We will see in
Assignment 4 how to use the tuned circuit in conjunction with the amplifier to
provide even higher values of Q. When damping is unavoidable, e.g when a

- 41 -
load must be connected, the resultant lower Q is sometimes called the
effective Q of the circuit, Qeff.

But a high value of Q is not always what is required. Although a high Q

provides excellent selectivity over a small bandwidth, if a larger bandwidth is
required the Q must be artifically reduced. One way to do this is to provide a
damping resistance. The reduction in output can be compensated for by
subsequent amplification. Another method of modifying the effective Q is by
coupling coil together.

Coupled coils are used where it is necessary to transfer signals from one
circuit to another without any direct connection between the two. This may be
necessary if, for example, the two circuits are at different d.c potentials. It can
also be used to deliberately exploit some of the characteristics of such circuits.
We have used a coupled circuit where only the secondary is tuned and seen
that this behaves as an ordinary tuned circuit. The advantage here is the
impedance transformation provided by the transformer. Let us now see what
happens when both halves of the coupled circuits are tuned.

Practical 2.5

With your swept frequency response still displayed as in practical 2.4, with no
dampng resistor, centre the tuned circuit on 465kHz and widen out the display
using the controls on the signal source module 2950A as you learnt in
Practical 2.4. Centralize the display such that the skirts of the response curve
are near the sides of the screen.

Take the other tuned circuit (that you have not used so far) and set its
resonant frequency to about 465kHz using the approximate calibration of the
tuning capacitor. Slowly bring it closer to the tuned circuit whose response is
displayed so that the coils are in line. You now have the set-up indicated in
Fig 2.11.

Q 2.23 What happens to the displayed response?

We will call the first tuned circuit the primary, the second the secondary.
Observe what happens when you mistune the primary or the secondary. Check
that both are tuned to the same resonant frequency by selecting a position
when they are about 40mm apart and the display is just beginning to show two

- 42 -
peaks, and adjust them so that the peaks are symmetrical.

Fig 2.11

Exercise 2.6 (a)

Sketch the characteristics for several different values of coupling. Label these
sketches 'primary current'.
Now connect the leads from the oscilloscope through the diode and capacitor
to the secondary terminals of the second tuned circuit. With the coils together
you should get a similar display to that you have just sketched. Observe what
happens when you move the circuits apart so that there is no coupling and
then back together again. Note the effect of mistuning either the primary or
the secondary and then check that they are both tuned to the same resonant
frequency by adjusting for a symmetrical display at the point where two peaks
are just displayed.

Exercise 2.6 (b)

Sketch the characteristics for several different values of coupling. Label these
sketches 'secondary output'.

The mathematics of coupled tuned circuits is too complex to be included in

this book but let us offer you an explanation of the effects you have just
sketched.

- 43 -
We are using two separate tuned circuits. The one fed from the signal source
we are calling our primary tuned circuit; the one we bring up to it we are
calling our secondary tuned circuit. Although these tuned circuits are
transformers in themselves, with their own primaries and secondaries, the
coupling here is fixed and you need not consider this. Consider only that
varying the physical distance between the primary and secondary tuned circuits
varies the amount of mutual coupling between them. When they are far apart
they are said to be loose coupled, when they are close together, the coupling
is said to be tight or close. The circuits are no longer independent of one
another. The primary current sets up a magnetic field. This induces a voltage
in the secondary, causing a current to flow and build up another magnetic field.
This in turn induces a voltage in the primary and so modifies the conditions in
that circuit. The magnitudes of these currents and the effects involved depend
upon the tuning of the two circuits. We will confine the discussion to the case
where both circuits have been separately tuned to the same resonant frequency
fO. You have seen in your practical what happens if one is off tune.

When the coupling is very loose, the response of each circuit is almost
unaffected by the presence of the other. The primary curve is thus unaltered
and no voltage is induced in the secondary. With a little more coupling the
secondary begins to show its own resonant curve. On the slopes of this curve
as the secondary is approaching its own resonance, not only is the induced
voltage from the primary increasing, but the secondary impedance is also
changing. these two effects occur together and the resultant secondary curve
is very sharp, i.e it shows excellent selectivity. Further increase in coupling is
accompanied by a continuation of this trend until a particular value of coupling
is reached - critical coupling.

At this point the current induced in the secondary at resonance is producing a

magnetic field which is beginning to affect the primary. This causes the peak
of the primary curve to flatten slightly. This again is reflected in a flattening
of the secondary curve. This can be thought of as an increase in the
resistance coupled into the primary and it can be shown that critical coupling
occurs when the coupled resistance equals the primary resistance. This means
that the conditions for maximum power transfer are satisfied. (Remember that a
tuned circuit is resistive at resonace).

Further increasing the coupling increases the resistance coupled into the
primary at resonance and consequently the values of both primary and
secondary currents at resonance fall.

- 44 -
Slightly above resonance, both primary and secondary impedances increase and
in fact become inductive. However the coupled impedance into the primary
contains a capacitively reactive component which tends to counterbalance the
change in primary impedance. This 'pulls' the primary into another state of
resonance with consequent increase in the current. Because of this, the
secondary output rises to a new peak approximately equal to the optimum
value when under critically coupled conditions. A similar situation occurs
slightly below resonance, except that the phases of the impednaces reverse.
Another state of primary resonance occurs equally displaced about the orignal
resonance.

Your sketched curves should correspond to the above descriptions. As we

stated, the mathematics is beyond the scope of this manual. Further
consideration of these results is given in the next section.

Practical considerations & applications

The frequency selective characteristics of impedance networks are of great

importance in all branches of communications. In this assignment you have
studied the phenomenon of resonance that results from an alternating voltage
of a predetermined frequency being applied to a parallel combination of
capacitance and inductance (and resistance).

The text of the practicals described in detail how resonance occurs and how
the output is a maximum at a frequency called the resonant frequency. This is
determined by the component values and you should have found that the range
of frequencies you calculated in exercise 2.2(b), with the range of component
values available, was between 395 and 562kHz. This range of frequencies
corresponds with the markings on the circuit tuning capacitor. Resonance can
thus be induced either by varying the component values L or C or by varying
the frequency. All three methods have been used, as you have seen. Care must
be taken when winding inductors so that the turns do not introduce extra
capacitance.
The size of the maximum output available at resonance depends upon the
circuit resistance, the smaller this is the steeper the gradient of the
characteristic curve and the more selective the circuit is said to be. This is
the concept of Q, which again was discussed in the text of the practicals. If
the resistance could be reduced to zero, Q would become infinity and the
output of resonance would be infinitely great. In practice, the inductor has

- 45 -
some resistance due to the wire from which it is wound. The loss resistance
of the capacitor is usually small enough to be ignored. It must be remembered
that at high frequencies the resistance is the r.f resistance, which varies with
frequency and is several times greater than the d.c resistance.

We have been working with a parallel tuned circuit with the resistance in the
inductive arm, as shown in Fig 2.5. Because the currents through the
inductance and capacitance are thus never exactly of opposite phase, the three
different conditions for resonance as indicated in practical 2.4 occur at slightly
different frequencies. It can be shown that for Q values greater than about 10,
the difference between these frequencies is negligible. However there are
occasions when the circuit Q may be so small that these effects must be taken
into account in calculations. Examples occur in high power transmitters and
industrial r.f heating coils, where the damping effect of the load dramatically
reduces the Q. You have seen the effect of damping when you connected a
load resistor across the circuit. The output impedance of the signal source also
damps the Q, so for good selectivity a parallel tuned circuit should be fed from
a high impedance supply. In this assignment the damping is noticeable, so any
calculations of dynamic impedance, etc will not give a true value. You will
learn in Assignment 4 how to use a tuned circuit in conjunction with an
amplifier in order to overcome this. Thus you can see that tuned circuit design
is a compromise between output, impedance and selectivity.

Remember also that the circulating current at resonance is Q times the supply
current and, especially in power amplifiers and transmitters, the size of the
conductors may have to be large to carry this. THis high current implies a
high value of stored energy in the inductor and capacitor and it is quite
common to speak of this circuit as a tank circuit as it behaves as an energy
reservoir.

Similar effects happen in a series tuned circuit, but it can be shown that with a
fixed voltage applied, the current is a maximum, corresponding to a minimum
impedance. Again the Q factor is determined by the series resistance, so it
follows that for good selectivity, a series tuned circuit should be fed from a
generator of low output impedance. The three separate conditions for
resonance in our parallel circuit occur at the same frequency as a series tuned
circuit. Taking our output as the voltage across the capacitor, this voltage will
reach its maximum slightly to one side of resonance, the amount being small
for high Q circuits. Thus a series circuit is sometimes called a voltage
resonant or acceptor circuit and a parallel one a current resonant or rejector

- 46 -
circuit. Parallel resonance is sometimes known as antiresosnance. These
differences are summarized in Fig 2.12. You will have met some of these
before in your earlier work.

Fig 2.12

The impedance frequency response of a parallel tuned circuit is the same as

the current frequency response curve of a series tuned circuit. It can also be
shown that for high Q values, the same equations apply. These factors simplify
matters because it is frequently unnecessary to consider whether a circuit is of
the series or parallel type. We can draw a UNIVERSAL RESONANCE CURVE
as in Fig 2.13 by taking the value at the apex as unity and scaling the other
ordinates accordingly. If the frequency axis is calibrated in units of frequency
  
shift from the resonant frequency, or Q where  =  where  is

the frequency in question.
It can then be shown that the ordinates can be calculated by the formula


Y = 
   


The maximum occurs when  = 0 and when Q =  ± 0.5, Y =  i.e

the half power points which define the bandwidth. Since this parameter  is a
ratio it is not linear with frequency, it will not be quite sym etrical if the
resonance curve is plotted to a linear frequency scale as we have in our
assignment. Assuming Q stays the same, it will be better if plotted on a
logarithmic frequency scale. In practice over such a small frequency range, the

- 47 -
difference is small. Also this matches up with the logarithmic dB scale used for
the ordinates.

The phase shift of the output is also ahown in Fig 2.13.

Below and above resonance, when the circuit is inductive or capacitive, the
phase shift of the output is obviously ±90 . At resonance we have proved
that the circuit appears resistive and the phase shift is zero. THe phase shift
changes through resonance, the rate of change being a maximum at resonance
where the rate of change of reactance is largest. Notice that the phase shift at
the half power points is ±45 . This may mean that different harmonics of the
signal within the bandwidth are phase shifted by different amounts. This is not
too important in sound broadcasting as the ear is relatively insensitive to phase
changes. You will see this happening later when we cover modulation. There
may however be occasions where this must be considered.

Fig 2.13

Fig 2.13 also shows the slight differences in the curve between Q of 10 and
very large values. As we have said, it is only below about 10 that it begins to
affect the results. Whilst on the subject of Q, you may find Q-values given

as figure of merit for a coil. Remember that Q =   so a frequency of

operation must always be implied, if not specified when such a statement is
made.

- 48 -
You have also seen that using mutual inductance to couple the signal into our
tuned circuit makes use of the impedance matching characteristics of the
transformer, as well as its voltage transformation. You will have learnt in your
other work and we reminded you in the practical, that an impedance connected
onto one side of a transformer is reflected into the other side. Its impedance
is effectively multiplied by the square of the turns ratio. Thus the 600
source impedance appears across the secondary as


600 ×  


or 31.5k .

This is still sufficient to damp the Q, so the Q you have measured is in fact
the effective Q, Qeff of the circiut. You should have a value of about 40
(32dB). The actual Q of the circuit is about 90. This also makes calculations of
dynamic impedance meaningless, which is why we have not stressed this point
here. We will return to it in Assignment 4. It is worth nothing that similar
impedance matching effects can be obtained by using a tapped inductor,
although this does not offer electrical isolation between the two circuits.

You will use a tuned circuit on almost every assignment from now on, which is
why we have dwelt at length on its properties. A high Q means good
selectivity, i.e a narrow bandwidth. Signals on one frequency may be sharply
amplified and those on adjacent frequencies rejected. This also helps to reduce
noise. If one stage does not provide sufficient selectivity, several stages in
cascade may be used. However, extreme selectivity is not alwyas what is
required. For intance, some television and radar signals, and even some FM
signals have quite a large bandwidth. In these cases the Q may be deliberately
lowered by introducing extra resistance or a pair of coupled tuned circuits may
be used.

An explanation of implied tuned circuits was given to you at the end of

practical 2.6 and your sketches of primary current and secondary output should
look like Figs 2.14 and 2.15 since it is usually the voltage across the
secondary capacitor that is used as the input to the following stage, it is the
curves of secondary output Fig 2.15 that are of more interest.

- 49 -
 Fig 2.14 Fig 2.15

In the overcoupled cases, one peak may be higher than the other because of
the change is capacitive reactance with frequency. These circuits are commonly
used in the intermediate frequency amplifiers of radio receivers where fixed
frequency operation is employed. This avoids the difficulties which would
otherwise arise from the necessity for simultaneous tuning of two coupled, but
separate, parallel resonant circuits. Their other advantage is the electrical
isolation between the primary and secondary.

The ideal wide band response is a flat top with steep sides, Fig 2.15(a). This
can never be achieved in practice, but is approached by a pair of critically
coupled tuned circuits. A wider band can be achieved by using overcoupling in
conjunction with another single tuned circuit of half the Q value of either of
the overcoupled circuits. The peak of the single hump approximately fills the
trough of the other. Double humps may also be obtained by tuning the two
circuits of a coupled pair to different frequencies. If the several tuned circuits
of a multi-stage amplifier are tuned to different frequencies the result can be a
substantially flat-topped wide-band response curve. This is known as
stagger-tuning. The amount of
coupling determines the exact form of the response. Much work beyond the
scope of this manual has gone into the mathematics of such reponses,
especially to produce the Butterworth, or flat-topped response of Fig 2.16(c)
and the Chebyshev, or equal-ripple response of Fig 2.16(d).

- 50 -
 Fig 2.16

Similar theory is used in filter design and will be mentioned again in

Assignment 5. Note also the similarity between this and the waveform analysis
of Assignment 1.

Variable selectivity is often a necessary feature in equipments, particularly in

communications receivers. It can be accomplished in various ways, for instance.

1. By physical movement of one of the windings providing variable coupling

between the coils of coupled circuits, exactly as you did in your practical.

2. By the introduction into the coupled circuit arrangement of a third winding

through which the transfer from one circuit to the other is effected. A
variation of the number of turns on this coil, by means of a tapping switch
of the value of a resistor across its ends, gives a change in the value of
coupling.

It can also be arranged that the selectivity shall vary automatically according
to the strength of the signal or the interference level.

- 51 -
It is worth mentioning that the degree of coupling in coupled circuits is much
less than that between the windings of power transformers. In these nearly all
the flux cuts nearly all the turns of the primary and secondary. With coupled
tuned circuits, loose coupling is about 1% and even what is described as tight
or close coupling is only a few percent.

The common impedance required for coupling may also be obtained by series
or shunt capacitance as shown if Fig 2.17. These can also be used in
combination to obtain a desired response, especially advantageous is to
combine the types of figures 2.17(a) and (c).

Coupled circuits
Fig 2.17

- 52 -
ASSIGNMENT - 3
- THE CRYSTAL

Object
To investigate the characteristics of a quartz crystal unit.

Equipment required
Power Console ED-2900
0-4 volts d.c. variable, metered.
+15 volts d.c
-15 volts d.cs

Oscilloscope,double beam,5MHz,calibrated,long persistance.

Digital Frequency Counter, 1MHz
High Impedance d.c voltmeter

Modules type 2950A, E

Tuned Circuit Assembly Type 2950H
Crystal Filter Assembly Type 2950J

Approximate Time Required

One hour

Prerequisite Assignments
1 and 2

Theory Assumed
As covered in Assignments 1 and 2.

- 53 -
Preliminary Procedure

1. Attach the console ED-2900P.

2. Connect the power supply unit(s) to the mains supply line. DO NOT
SWITCH ON YET.

3. Turn the variable d.c control knob to minimum.

4. If you are using any of modules 2950A, B, C, D, or E, plug them into the
sockets on the Console on alphabetical order. If any change is required, it
will be indicated in the assignment. The act of plugging in the modules
connects them to the power supply in the correct manner. All sockets
are identical.

5. If you are using either of the modules 2950F or G, or the Assemblies

2950H, J, or K, these do not require any power and may be stood
wher convenient or as directed in the assignment.

6. This completes the preliminary procedure.

Experimental Procedure

You have investigated electrical resonant circuits in Assignment 2 and seen

how they are frequency dependent. Similiar effects occur in mechanically
vibrating systems and can be used in a similar fashion.

The Quarts Crystal Oscillator

Certain substances, notably quartz, exhibit what is known as the
PIEZO-ELECTRIC effect; a mechanical strain applied to a suitably cut piece of
that substance causes a voltage to be developed across tow surfaces of that
piece. Conversely, a voltage applied across two faces causes mechanical
deformation. These effects are used in gramophone pickups and similar
devices. If an alternating voltage is applied to such a crystal, it behaves
electrically as a resonant circuit, resonant at the natural frequency of
mechanical vibration of the piece. The low mechanical damping inheret in
quartz, results in a very high Q. Since quartz is a very stable material, the

- 54 -
frequency of mechanical oscillation of the piece is also very stable.

If such a crystal is mounted between two flat metal plates to which an

electrical connection can be made, Fig 3.1(a), it is found to behave like the
equivalent circuit of the Fig 3.1(b). the conventional circuit symbol for such a
crystal is shown in Fig 3.1(c). Components C, L and R are merely the
electrical equivalents of the mechanical properties of the crystal, their values
being determined by the dimensions and physical properties of the crystal
itself. Capaciator C' is the electrical capacity between the plates holding the
crystal and may also include the capacity of the writing. The crystal is
mounted in a vacuum to eliminate air damping.

You will notice that the circuit composed of C, L and R in Fig 3.1(b)
corresponds to a series resonant circuit. We mentioned in Assignment 2, and
you should recall from your previous studies, that a series resonant circuit has
a minimum impedance at resonance and, if a fixed voltage is applied, the
current flowing will be a maximum at resonance. Thus the considerations of
the universal resonance curve will apply. Let us see if we can demonstrate
this.

Fig 3.1

- 55 -
Crystal Unit module 2950J

Look at the crystal filter in the Assembly

type 2950J in Fig 3.2 and satisfy your-
self that it conforms with the description
given. Note the evacuated glass envelope
containing the crystal and marked with the
precise resonant frequency.
The function of the variable capacitor will
be explained in Practical 3.3.

Fig 3.2

Practical 3.1 Estimation of Q value of Quartz Crystal

A. Resonance in the crystal

To observe the crystal acting as a resonant circuit, connect up as in
Fig 3.3.

On the module 2950A

Set the variable attenuator control to 0dB.
Set the B sensitivity control to zero.
Set the manual frequency control C to zero.

On the oscilloscope:
Set the timebase to 1 s/cm and the amplifier to 2V/cm.

On the Console:
Set the Var.d.c to +4V and the control to zero.
Switch on and SLOWLY turn the manual frequency control C.

- 56 -
 Fig 3.3

Q 3.1 What happens to the output voltage from the crystal?

If you did not see anything, repeat more slowly.

Q 3.2 What does this tell you about the Q factor of the crystal?

Because the Q is very high, in order to see the resonant peak, we must
examine the output over a very narrow frequency range. We can do this with
the help of the external d.c input.

Set the manual frequency control C just above the resonant frequency: do this
by observation of the disply.

Set the B sensitivity control to 0.01. Then slowly reduce the frequency by
turning up the variable d.c control until the crystal resonates. If this does not
occur reset the control and reduce the manual frequency control C setting by a
vey small amount and try again. This adjustment is cristical.

Q 3.4 What shape is the output waveform at resonance? Why?

- 57 -
Q 3.5 Why do you think there is amplitude jitter present?

B. Estimation of the Q value of the crystal Voltage

Measure with an accurate voltmeter the variation Upcer value
in d.c control voltage required to shift the Lower
crystal through resonance. We can take this as the value
bandwidth b of the crystal and record your results Fig 3.4
as in Fig 3.4.

Exercise 3.1

Estimate the Q of the crystal from the formula
Q =  where fo is the

resonant frequency of 465kHz and b the bandwidth.
From Assignment 1 you know the change of frequency produced by a
voltage input of +4. Sensitivity control B will scale this by 0.01. The
chage of voltage obtained in Fig 3.4 can therefore, be given as the
bandwidth b .

Q 3.5 How does this compare with the Q factor of the electrical tuned circuit
of Assignment 2?

C. Effect of load on crystal resonant frequency.

On the digital frequency counter:
Set the range so that it resolves the last digit in tens Hz.
Now record as in Fig 3.5, the peak-to-peak
display voltage at resonance and the freq-
uency counter reading.

Refer to your attenuator calibration from Display voltage Frequency

Assignment 1 and look up the voltage being Vp-p Hz
fed into the crystal from the 0dB output for
the frequency recorded in Fig 3.5.
Fig 3.5

Q 3.6 Why is the voltage indicated on your oscilloscope larger than that being
fed into the crystal and why is the precise frequency measured not

- 58 -
 quite the same as that marked on the crystal?

The input of your oscilloscope has an input capacitance and resistance. This is
shown together with the crystal in Fig 3.6.

Fig 3.6

As you will learn later, the input capaciatance of your oscilloscope Cin is large
compared with the crystal capacitance C. Thus it reduces the effective
capacitance of the tuned circuit and increases the resonant frequency slightly.
It is this you have measured.

As the capacitor Cin forms part of the frequency determineing capacitance of a

tuned circuit the voltage across it at resonance may be higher than the input
voltage. Similarly Rin reduces the effective Q from that theoretically obtainable.

D. To demonstrate damping effect of a resistor

On the 2950C module:
Connect the other end of the 4.7k resistor (point A) across the input
to the oscilloscope.
Return for maximum output and measure the output voltage and precise
frequency.

Q 3.7 How do these now compare with Vin and fo as marked on the crystal?

You should find the agreement is better.

- 59 -
Practical 3.2 Anti-resonance

To use the swept frequency display technique to look at the resonance of the
crystal, connect up the circuit of Fig 3.7.

Fig 3.7

On the Module 2950A

Set the variable attenuator to 0dB.
Keep the setting of the frequency control C.
Just off resonance and sensitivity control.
B at 0.01 as in Practical 3.1.

On the oscilloscope:
Set the amplifier to 0.1V/cm and the timebase to 500 ms/cm.
Adjust the sweep width by the sensitivity control B and position with
with control C.
If your oscilloscope timebase output is large, you may have to extern-
ally attenuate it to get the required sensitivity. You should be able to
obtain a display as in Fig 3.8.

Q 3.8 Is the display symmetrical?

Increase the Y gain to 0.2 volts/division and look carefully at the center of
the display.

- 60 -
Q 3.9 Can you notice a small 'notch' to
one side of the main peak?

Q 3.10 Is the notch frequency higher or

lower than the resonant frequency?

Q 3.11 Looking at the equivalent circuit

Fig 3.1(b), why do you think these
effects are caused? Fig 3.8

The circuit has another, slightly higher resonant frequency caused by the
holder capacitance C'. The crystal and holder form a parallel resonant circuit.
At resonance the capacitance current passing via C' is equal and opposite to
the inductive current through C, L and R. The current flowing and voltage
across the oscilloscope is a minimum, causing a 'notch' in the trace. This point
is called the ANTI-RESONANCE point of the crystal.

Q 3.12 How can we compensate for the anti-resonance caused by the

capaciator C'?

To neutralize the effect of the holder capacitace C' we need to balance it out
in phase as well as magnitude.We can do this with a centre-tapped transformer
as in Fig 3.9 and a variable capacitor CB as the fourth arm of a bridge, known
as the Lamb bridge.

Fig 3.9

- 61 -
Practical 3.3 Use of the Lamb Bridge

To show the effect of neutralizing the holder capacitance C', switch off and
connect up the layout of Fig 3.10.

Fig 3.10

Switch on,
On the oscilloscope:
Set the timebase to 500ms/cm and the amplifier to 0.5V/cm. Slowly adjust
the timing capacitor in the secondary of the transformer in the tuned
circuit assembly until the display is a maximum. You may temporarily
increase the timebase speed to do this, but remember to return it to at
most 500ms/div. You should again have the display of Fig 3.8 and you
should be able to see the anti-resonance clearly.

On the 2950H:
Turn the setting knob so that the
vanes of the balancing capacitor CB
are fully open.
Connect its other side, terminal A,to
terminal B on the primary transformer
side of the module 2950H. In order
to make the anti-resonance notch more
clearly visible, increase the Y gain
to 50mV/division. Contrive to use very
slow sweep speeds on your oscillos- Fig 3.11(a)
cope. Tou should now have the display
of Fig 3.11(a).

- 62 -
 Fig 3.11(b) Fig 3.11(c)

Turn the balancing capacitor CB to its maximum position with the vanes fully
meshed.

Q 3.13 Where is the anti-resonance notch now?

Find the setting of the balancing capacitor CB so that your display is

symmetrical.

Q 3.14 Where is the anti-resonance notch now?

Think about this question before you answer it.

Finally reduce your Y gain setting back to 0.5 volts/cm and observe the full
swept resonance display.

Exercise 3.2

Make another estimate of the bandwidth of the crystal from your knowledge of
the sweep voltage, the sensitivity setting and the signal source frequency

calibrator curve. Estimate Q from the formula   


- 63 -
Practical considerations & applications

Quality Factor of crystal

The striking fact emerging from this assignment is the exceptionally high Q of
a crystal. It is very difficult to measure accurately but you should have been
able to estimate a value of about 2500, which is about a hundred times higher
than that of electrical tuned circuit as measured in Assignment 2. This arises
from the values of L, C and R in the equivalent circuit of Fig 3.1(b), which are
determined by the mechanical properties of the crystal itself. We already know
how the crystal oscillates, but it will help you to understand the magnitude of
the figures involved if we discuss a little of how the crystals are made.

Preparation of crystal

Quartz occurs naturally in crystal form, but it can also be synthetically

cultured. A typical crystal and its outline are shown in Fig 3.12(a) and (b).
Slices are then cut from this large crystal, and described in terms of the angle
the cut makes with one of the major axes. These cuts which are given code
letters, determine the mode of vibration and the frequency range. The precise
frequency is determined by grinding, lapping and etching the slice to the
required thickness. The contacts are often mades by depositing the electrodes
directly onto the crystal faces and/or clamping it between metal electrodes to
which the lead out wires are attached. The mounted crystal is enclosed in a
container and often evacuated to eliminate mechanical damping due to the air.
Virbration may take place in thickness, in bending or in shear. Crystal design,
cutting and mounting involve the selection of dimensions, such that only one of
these modes will give a resonant frequency in the desired range of operation.
The resonant frequency is a function of temperature, but the coefficient can be
made zero, positive or negative as required by proper orientation of the cut
slice.

Operating frequenies and Q

Crystals can be made with fundamental frequencies ranging from about 200Hz

- 64 -
to 20MHz. Above this the quartz plates become very thin and fragile but it is
possible to make crystals vibrate in harmonic overtones, raising the upper
limits to about 200MHz.

Quartz crystal
Fig 3.12

The parameters R, L and C in the equivalent circuit of Fig 3.1(b) are analogues
of the mechanical properties of the crystal. It has a very high ratio of mass to
elasticity (equivalent to a high L/C ratio) and a very high ratio of mass to
damping (equivalent to a high Q ratio). Although the actual Q for the quartz
itself may exceed 106 (equivalent to a low value of R) it is usually between
about 2000 and 200,000 (equivalent to increasing R). Even these values are
appreciably higher than normal tuned circuits.

For example, a typical crystal resonating at 456kHz may have equivalent circuit
values as follows:

- 65 -
 L = 3.66H R = 450
C = 0.0316pF C'= 5.75pF

Anti-resonance

You have seen the effect of the parallel or anti-resonance caused by the
holder capacitance C' and how it causes a 'notch' very close to the operating
frequency. It can be shown that the anti-resonant frequency is given by


fa = fo ( 1 +  )
 ′
the ratio C/C' is about 1/150, so fa = 1.003 fo

Effect of the Lamb Bridge Circuit

By using a Lamb Bridge circuit the current flowing through the balancing
capacitor CB in Fig 3.9 is approximately in anti-phase with that through the
holder capacity C' and the effect is therefore the same as reducing C'. This
increase the anti-resonant frequency up to infinity, when CB = C' to produce
the neutralization point. If CB is further increased, this effectively makes C'
negative and raises the anti-resonant frequency from zero until it again
approches the main resonant frequency. The mathematical formulae to prove
this are beyond the scope of this manual but are included in most textbooks
including a section on crystals.

Effect of loading crystal

Another factor you have seen demonstrated is the effect of the input
capacitance Cin of any circuit following the crystal. Here it was the measuring
oscilloscope, but it could equally well be an amplifier following the crystal.
This capacitance is always large in comparison with the capacitance C of the
crystal. Although it causes a slight shift in the resonant frequency (as well as
voltage magnification), changes in this capacitance C will only cause minute
percentage chages in the resonant frequency. However the crystal unit can be
adjusted if the load capacitance is known, or assumed to have a standard value

- 66 -
of 30pF, so that the marked frequency is achieved when the load is connected.

Effect of temperature chage on performance

Crystals are often used in small temperature controlled ovens for stability and
under such conditions with stablized supply voltages, the stability of frequency
possible may be of the order of 1 to 10Hz per MHz over long periods of time.
With such stability it is only natual that crystals be employed as the frequency
dertermining element in an oscillator circuit, and such oscillator are used for
the preicse measurement of time. You will have heard mention of 'Quartz
crystal clocks' and it is also likely that a crystal provides the reference for
your digital frequency counter. You will construct a simple crystal oscillator in
Assignment 4.

In this Assignment, no great care was taken to ensure good stability and
because of the high Q, this appeared as amplitude jitter at resonance.

Use in band-pass filters

The high Q of crystals also makes them usefull as very narrow band filters,
where it is necessary to pass only one frequency and sharply attenuate all
others. By suitable choice of circuitry the bandwidth can be increased by
employing two or more crystal, possibly in a pair of coupled tuned circuits as
you saw in Assignment 2.

If a pair of matched crystals is required, then two pairs of electrodes may be

deposited on a single crystal slice to provide a matched pair of crystal
elements. The theory behind such filter circuits is fully covered in relevant
textbooks. You will learn a little more about filters in Assignment 5.

- 67 -
ASSIGNMENT - 4
- THE AMPLIFIER

Object
To investigate the use of a single transistor as a linear-tuned or
voltage-controlled amplifier.

Apparatus required
● Power Console ED-2900P
0-4/0-20 volts d.c, variable, metered.
+15 volts d.c
-15 volts d.c
● Oscilloscope, double beam, 5MHz, calibrated.
● Digital Frequency Conter, 1MHz
● Signal Generator
● High Impedance d.c voltmeter and
● Modules Type 2950A, B, C, D, E, F, and G.
● Tuned Circuit Assembly Type 2950H.

Approximate Time Required

Two hours

Prerequisite Assignments
1,2

Theory assumed
Knowledge of transistor characteristics and use of load lines.

- 69 -
Preliminary Procedure

1. Attach the console ED-2900P.

2. Connect the power supply unit(s) to the mains supply line.

DO NOT SWITCH ON YET.

3. Turn the variable d.c control knob to minimum.

4. If you are using any of modules 2950A, B, C, D, or E, plug them into the
sockets on the Console in Alphabetical order. If any change is required, it
will be indicated in the Assignment. The act of plugging in the modules
connects them to the power supply in the correct manner. All sockets are
identical.

5. If you are using either of the modules 2950F or G, or the Assignments

2950H,J, or K, these do not require any power and may be stood where
convenient or as directed in the Assignment.

6. This completes the preliminary procedure.

Experimental Procedure

In many electronic applications we need to raise the level of a signal. You

remember we said in Assignment 2 that small outputs could be made bigger.
To do this we use a circuit whose output is greater than its input. Such a
circuit is called an AMPLIFIER. One type is the voltage amplifirer which has
the output voltage greater than the input voltage. There are also current
amplifiers and power amplifiers.

For instance in a communications system the signal may be of very low value.
We may want to amplify a radio frequency signal in order to provide sufficient
signal to or from the aerial. In the receiver we need to amplify the audio
signals to provide power to operate a loudspeaker.

The frequency response of amplifiers is important. If they do not amplify all

frequencies present, by the same amount, some harmonics in a complex signal
will be attenuated and the output will suffer DISTORTION. Conversely, the

- 70 -
amplifier should not add any harmonics that were not present in the original
signal. In other words the output waveform should be as faithful a replica of
the input as possible. The fact that the output power may be larger than the
input power does not contravene any natural laws since the extra power is
derived from the amplifier power supply.

However in certain uses it may be adventageous to use an amplifier that

responds only to a certain range of frequencies, as this will allow us to select
one wanted signal from several unwanted ones. The differences between the
types of amplifier, the effects of the points we have touched on, and some of
their uses will be made clear during this assignment.

The Transistor as an Amplifier

You already know from your other studies that a TRANSISTOR exhibits the
property of amplification when properly connectd. We will only summarize and
use the results here.
One method of using the transistor as an amplifier is to connect it in COMMON
EMITTER configuration as shown in Fig 4.1. We are using an npn transistor,
Therefore a positive supply voltage is required. Resistors R1 and R2 supply
the bias to the base and determine the quiescent or operating point which
provides for good thermal stabilization of the circuit. The resistor RL is the
collector load resistor. Capacitors CB and CC are coupling capacitors to
isolate the d.c conditions of the circuit from the signal path. Capacitor CE
decouples the emitter resistor at the operating frequency.

- 71 -
 Transistor in common emitter
Fig 4.1

The characteristics of the transistor itself, type ZSC 1815 are shown in Fig
4.2.

Q 4.1 Look at the output characteristic (IC against VCE). Is the spacing
between the curves the same for equal increments of base current
IB? What does this tell you about the amplification provided by this
transfer?

Q 4.2 Look at the transfer characteristic (IC against IB). Why does this
confirm your answer to Q4.1?

Q 4.3 Look at the input characteristic (VBE against VB). The input resi-
stance of the transistor is dete- rmined by the slope of this curve.
Is it constant?

- 72 -
 Fig 4.2

With these points in mind let us begin to look at our amplifier module.

Amplifier Module

Look carefully at the circuit diagram on the front of the amplifier module
2950B and compare it and the component values with that suggested in the
previous section. The module circuit diagram is repeated here as Fig 4.3 and
there are a number of comments to be made on its differences and similarities
with the theorectical diagram of Fig 4.1.

- 73 -
 Fig 4.3

a) For the npn transistor in the module the collector should be connected to
the positive side of the supply as we indicated in Fig 4.1. On the module
we have chosen to earth the collector circuit and connect the emitter
circuit to -15 volts. This has the same effect but has the advantage that
any external loads which may be connected in the collector circuit, will
have one side earthed. With this connection stray capacitances are reduced
together with the risk of pick-up on the long lenths of wires associated
with ED-2900 series.

b) As far as the a.c signal is concerned all supply lines are at the same
potential, in this case ground, so the circuit is still a common emitter
circuit when we take our input and output a.c signals relative to ground.
Similarly the emitter decoupling capacitor can be returned to ground.

c) A choice of values for the emitter decoupling CE is provided. When

connected in series, the 25 F has little effect and the capacitance can be
considered as 47nF Similarly the input coupling capacitor CB can be varied.
A 620 resistor is permanently connected in the emitter in series with 10k
and 47k. These may be shorted out by the sockets to select 620 , 10.6k ,
or 57.6k . This provides for varing the emitter resistor RE.

- 74 -
d) A 620 load resistor R7 can be connected in the collector circuit. The
output is then taken form either side of the capacitor C5. Alternatively an
external collector load may be connected.

e) Resistors R1 and R2 provide base bias. Resistor R1 is for providing

external base bias and will be dealt with later.

Practical 4.1 Amplifier Gain

Now that we have examined the amplifier module, let us see what it can be
made to do. Connect up the circuit as shown in Fig 4.4.

Fig 4.4

On the signal source module 2950A

Set Manual Frequency Control C to the dot approximately 465kHz
Set Output Attenuator to -12dB
Connect a lead from 0dB output to oscilloscope external trigger
Connect -6dB output socket to amplifier module input.

On the amplifier module 2950B

By use of shorting link, select RE = 620 .
Link in the 620 load resistor and the 25 F decoupling capacitor.

On the oscilloscope
Set Y1 and Y2 inputs to 1 volts/division, d.c coupled
Set timebase to 1 s/div

- 75 -
 Select external trigger

Switch on the power and set the d.c level of channel 1 to the top of the
screen. Adjust the oscilloscope trigger control until you obtain a stable display
of the signal on channel 1.

Q 4.4 Is the output bigger than the input? Where is its d.c level?

Q 4.5 What happens to the output if you temporarily remove the link which
connects in the emitter decoupling capacitor CE? Does the d.c level
alter? Temporarily select a decoupling capacitor of 47nF and explain
why the output square wave becomes distorted.

Replace the 25 F capacitor.

Q 4.6 Is there a 180 phase chance between input and output as you would
expect from a common emitter amplifier?

Q 4.7 Increase the input by turning the signal source output attenuator
clockwise. What happens to the output?

In answering these questions, remember that our amplifier is 'upside down' to

the theorectical one, thus the zero level at the top of the screen is actually at
+15 volts with respect to the negative emitter circuit. If you have an inverting
input on your oscilloscope, you may appreciate this by temporarily using it and
setting the d.c level to the bottom.

We will now plot a graph of the output of the amplifier for various values of
input for all three values of emitter resistor. We know that the output is nearly
limitimg with -18dB input so we will only use inputs of -9dB and smaller.
Quite simply record the peak-to-peak output voltage from your oscilloscope at
3dB steps in the input. Do this for all three values of RE. Record your results
in a table as in Fig 4.5. At high inputs you will find that the lower peak of the
waveform becomes distorted. Sketch this and to measure its value take the
centre of the period. Mark these value with an asterisk (*) in your table. Note
the position of the flat top peaks relative to the d.c zero level when this
occurs.

- 76 -
 Fig 4.5

Exercise 4.1

a) On 3-cycle logarithmic graph paper plot graphs of output against input for
each of the three emitter resistors used. Use axes as shown in Fig 4.6.
Copy on to this graph the attenuator calibration graph you plotted in
Assignment 1, Exercise 1.2.

b) By referring to your table of Assignment 1 for the peak-to-peak output

voltage of the signal source at -30dB, calculate the voltage gain of the
stage for each of the three different emitter resistors. At -30dB input all
three graphs should be on a nearly linear portion. Express these gains in
dB.

Q 4.8 What happens to the gain of the amplifier as you increase the value of
the emitter resistor?

The actual reason for this is quite complicated. Larger emitter resistors mean
less current flowing through the transistor. This causes the parameters of the
transistor to change. You remember we pointed out initially that gain and input
resistance were not constant. These effects combine to vary the gain of the
amplifier.

- 77 -
 Fig 4.6

Q 4.9 How do your graphs show what happens if the input is too big? Can
you explain this with reference to the transistor characteristics? With
reference to your sketches, explain why the output distorts when this
occur. Remember the phase inversion in the amplifier.

Q 4.10 Think back to the assignment 2 and see if you can suggest a method
of eliminating this distortion.

Because we have mentioned Assignment 2 it is apparent that we want you to

consider the use of a tuned circuit. Remember this only responds to the
fundamental component in the waveform so it should be able to reduce the
distortion produced at limiting.

Q 4.11 Why is it not a good idea to connect the tuned circuit across the
existing 620 load resistor?

- 78 -
Practical 4.2. Distortion in a Tuned Amplifier

Let us therefore investigate what happens if we use the tuned circuit itself as
the collector load. Such a circuit is called a TUNED AMPLIFIER.
Connect up the circuit of Fig 4.7.

On the 2950A module

Set the manual frequency control C at the white dot, about 465kHz.
Set the manual attenuator to 0dB.

On the oscilloscope
Set the Y-amplifiers to 5V/div
Set the timebase to 1 /div

Fig 4.7

Tune the tuned circuit for maximum output. With input set on 0dB, see, if you
can notice any distortion present, particularly at the peaks of the waveform.

Reduce the input setting and note that the output stays approximately constant
and undistorted for a while. Record the value of input setting where the output
beings to drop sharply.

Repeat this procedure for RE = 10.6k with the oscilloscope at 0.5V/div and
then for RE = 57.6k at 0.1 volt/div, nothing in each case the input setting
where the output begins to drop sharply.

Q 4.12 Do these figures at which the output beings to reduce correspond with

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 the break points of the amplification curves you drew in exercise
4.1?

Q 4.13 Did you notice any distortion at limiting for any value of RE?

Q 4.14 Is there any reason to suppose that you would not get the same
amplifier characteristics (i.e linear amplification at low inputs,
limiting at high inputs, the break point being determined by RE ) if
you plotted them for the tuned amplifier?

The characteristics are of course very similar so it is not necessary to plot

them again. However, if you Practical 4.1 with the tuned amplifier.

Q 4.15 What did you notice about the output voltage level of the tuned
amplifier? Why do you think this is so?

Practical 4.3 Q-factor in a Tuned Amplifier

The extra output is provided by the Q-factor of the tuned circuit. Let us see if
we can measure it. We will use the same procedure as we did in Assignment
2, that is measure the bandwidth of the tuned amplifier and divide it into the
resonant frequency.

Connect up the circuit shown in Fig 4.8.

On the 2950A module

Set the sensitivity control B to zero.
Set the manual frequency control to 470kHz using the frequency counter.
Set the variable attenuator to -3dB.

On the Console
Set the 4-volt variable d.c supply to zero.

On the oscilloscope
Set the Y-amplifier to 5V/div.
Set the timebase 1 s/div.

- 80 -
Note the maximum output in volts peak-to-peak.

Fig 4.8

Now increase the input by 3dB to -24dB and, by adjusting the variable d.c in
both directions, measure the frequency at which the output decrease to the
value you have just measured.

Subtracting these gives the bandwidth. Divide the bandwidth into 465kHz to
obtain a value for Q.

Q 4.16 How does this compare with the value of Q from Assignment 2?

Q 4.17 What resistance is present to provide damping of the tuned circuit?

Q 4.18 Would you expect the damping and Q to change if you change RE?

Q 4.19 Why are we using a signal level of -24dB? Would you get a valid
answer if you used an input of 0dB?

Q 4.20 Is a tuned amplifier of use if the amplitude of the input is varying, i.e if
it is modulated?

Practical 4.4 Gain Control

You will examine this last question fully in future assignments. Now, do you

- 81 -
remember that we said that there is another way of varying the gain of the
amplifier.

This involved injecting extra current into the base through the auxiliary
resistor R1. Remembering that our circuit is 'upside-down' the actual voltage
on the base in our module is about -111/2 volts. Thus to inject extra base
current into this point we need to mark the other end of R1 positive with
respect to the base.

Q 4.21 What may happen under these conditions with large inputs?
(Hint : refer to your answer to Q 4.9)

Let us see if you are correct.

Reconnect the modules up again as in Practical 4.1 Fig 4.4.

Set the oscilloscope to 1 volt/div and 1 s/div and adjust the input level until it
is just on the point of limiting. Trigger the oscilloscope for a stable display
then connect the control input socket at the top end of resistor R1 to ground.
This is positive with respect to the negative voltage on the base, so extra
base current will flow.

Q 4. 22 What happens to the gain and shape of the output? Why?

Sketch the output.

Now transfer your input lead to the 0dB output socket and slowly reduce the
variable attenuator to 0dB.

Q 4.23 What shape is the output waveform now? Sketch it.

Q 4.24 Can you explain this with reference to the transistor characteristics?

Remove the link to the control input and note that the output returns to a
square wave, albeit limited.

Q 4.25 How can we reduce the distortion caused by controlling the gain by
this method?

This is where our tuned amplifier comes the rescue again. Disconnect the load

- 82 -
resistor and connect half the primary of the tuned circuit as the collector load
and the oscilloscope to the secondary output as you did in Practical 4.2. With
no connection to the control socket and 0dB input attenuation, adjust the tuned
circuit for maximum output.

Transfer the input lead to the -24dB socket to ensure you are on the linear
part of the amplifier characteristic and not limiting. Earth the control socket on
the amplifier.

Q 4.26 What happens to the output voltage? Can you see any distortion?

We will now plot the complete characteristics. We can obtain a variable by

using the d.c amplifier on the detector module, fed from the variable d.c power
supply and with the -12V offset linked in so that it is suitable for the voltage
on the base of our amplifier transistor. In this way we can vary the control
voltage from about -12 volts to +12 volts.
Connect up the circuit as shown in Fig 4.9.

Fig 4.9

- 83 -
On the 2950A module
Set the manual frequency control C to the dot position, about 456kHz.
Set the variable attenuator to 0dB.

On the 2950C module

Set the d.c amplifier control to minimum.

On the Console
Set the 4-volt variable d.c control to minimum.

On the oscilloscope
Set the Y-amplifier to 2V/div.

Tune the circuit for maximum output.

Turn the variable d.c to its maximum on the 4-volt range.

Adjust the gain of the d.c amplifier so that your voltmeter reading is just the
maximum. (This will give maximum sensitivity over the range of control).

Exercise 4.2

Take a set of readings of output voltage pk-pk against the voltage at the
amplifier control socket for different input levels. Record your results in a
table as in Fig 4.10. Plot a graph of these variations on the same sheet of
linear graph paper.

Q 4.27 Can you explain the general shape of these graphs with reference to
the transistor characteristics?

Q 4.28 Why do you think the output increases before it begins to decrease,
especially at higher input levels, with increasing control voltage?

Q 4.29 Is the reduction in gain (as distinct from output voltage) greater or
smaller for large inputs than small inputs.

- 84 -
 Fig 4.10

Keep your graphs carefully as you will need to refer to them in Assignment 8,
when one of the practical applications of this result will be demonstrated to
you. Meanwhile, we will consider in general what you have learnt in this
assignment.

- 85 -
Practical considerations & applications

As a general rule, in this kit, we have avoided giving details of the actual
circuits used to carry out the functions which you have investigated as it is
more important to learn of the effects and their applications than how these
effects are actually produced. However, here you meet one of the exceptions
to this rule, where a knowledge is needed of the amplifier in its different
modes, and this is especially the case of the amplifier in voltage control mode.
A full description of the circuit theory is outside the scope of this manual but
it is fairly standard and you will meet it in the course of your other studies.

You will, therefore, know how this single transistor can be made to operate in
the various modes. With a resistive load, you should have found that the actual
value of amplification depended upon the value of the emitter resistor RE and
that it was linear for low input levels, limiting (and distorting) at high input
levels.

You next investigated how this distortion could be reduced by using a tuned
amplifier which would amplify only the fundamental component of frequency of
the input waveform and you obtained a sinusoidal output with a sqare-wave
input. The Q value you measured should be about double that found for the
tuned circuit alone in Assignment 2, since the resistive damping of the tuned
circuit is much less. Many of the comments we made about tuned circuits in
Assignment 2 can also be applied to the tuned amplifier and even coupled
tuned circuits can be used to modify the response. Our component values have
been chosen so that the dynamic impedance of the tuned circuit is nominally
600 ohms, which is the designed load of the amplifier.

A tuned amplifier, therefore, amplifies only a narow band of frequencies

whereas the resistance-loaded amplifier has a much wider frequency response.
this frequency response is of impotance in the choice of which type of
amplifier to use. You must know for example if the amplifier is to be used for
the whole of the audio spectrum (about 20Hz-20kHz) or if it is required to
respond only to a particular frequency(e.g 465kHz). In this way a tuned
amplifier can pick out the signal of one radio station from many others.

You remember that we chose to earth the collector circuit so that one side of

- 86 -
our tuned circuit was at earth potential. This reduces pick-up and if the tuning
capacitor is in the collector circuit, it eliminates the need for insulation
between the spindle and chassis, and stability is improved. In general a similar
advantage may be had if the tuned circuit is connected to the input of the next
stage, suitably decoupled. By using a transformer we have in fact a similar
arrangement.

In addition power is wasted in the load resistance of an amplifier, if the

amplifier is a power amplifier in a transmitter; for example the waste power
that is dissipated as heat may not be tolerated. On the other hand a tuned
amplifier providing much greater efficiency.

Another factor to be considered is coupling to the next stage. A

resistance-loaded amplifier must be coupled via a capacitor to prevent the d.c
voltage on the collector from interfering with the bias conditions of the
following stage. this capacitor also affects the frequency response. A
transformer-coupled tuned amplifier needs no necessary isolation. the
transformer can also be used for impedance matching by suitable choice of the
number of turns and this sometimes achieved by a tap on the coil. You will
meet further examples in future assignments.

In a transistor amplifier the parameters of the transistor itself (e.g gain, input
and output resistance) vary with the current passing through the transistor.
These effects are exploited to vary the gain - by varying the emitter resistor
and also by injecting extra current into the base. The transistor type is
specially designed to perform this latter function and you will use it in
Assignment 8 to demonstrate automatic gain control. There are many other
applications where the gain of an amplifier has to be controlled by a d.c
voltage. Examples occur in multi-channel sound reproduction, computer control
of processed and so on. As you have seen this single transistor does not
provide very linear control. If this is important, multi-stage amplifiers can be
designed to have much better characteristics since each stage can then operate
over a small and more linear range. Such voltage controlled amplifiers can also
be obtained as a single integrated circuit.

- 87 -
ASSIGNMENT - 5
- FILTERS

Object
To investigate the properties of passive and active low pass filters.

Equipment Required

● Power Console ED-2900P

0-4/volts d.c, variable, metered
+15 volts d.c
-15 volts d.c

● Oscilloscope, double beam, 5MHz, calibrated.

● Digital Frequency Counter, 1MHz.
● Modules Type 2950A and E.
● Passive Filter Assembly Type 2950F.

Approximate time required

One and a half hours

Prerequisite Assignments
1 and 2

Theory assumed
Basic Knowledge of electricity

- 89 -
Introduction

A filter is a network which transmits signals within a given frequency ranges

and attenuates signals within another range. It has obvious uses in the
selection of wanted signals and the rejection of unwanted signals. A filter
therefore possesses at least one PASS BAND - a band of frquencies in which
the attenuation is theoretically zero and at least one STOP BAND (sometimes
called attenuation band) - where the attenuation is theoretically infinite. The
frequencies that separate the various pass and stop bands are called the
CUT-OFF frequencies. The ideal filter has a sharply defined cut-off, as shown
in Fig 5.1(a). These theoretical definitions can never be achieved in practice
and this has been considered in drawing Figs 5.1(b) to (e)where the
characteristics of four different types of simple filter are shown. As indicated
above, more complex filters can be achieved by using these in multiples and/or
in combination.

You have already met a filter of type (e) band pass - the tuned circuit you
studied in Assignment 2. You will recall that we said the ideal response of a
flat top with steep sides could never be achieved in practice and that we
mentioned methods of improving this. We introduced you to the Butterworth
and Chebyshev responses. these are named after the men who did the complex
mathematic analyses.

- 90 -
 Fig 5.1

A Chevyshev response is characterized by a maximum increase of attenuation

into stop band. This is achieved at the cost of some ripple in the pass band,
usually less than 1dB. A Butterworth response is maximally flat in the pass
band.
This is achieved at the cost of a less sharp cut-off into the stop band. Typical
responses are shown in Fig 5.2. The cut-off frequency o is defined as the
angular frequency at which the response is 3dB below its maximum.

- 91 -
 Fig 5.2

Passive Filters

We have chosen to investigate low-pass Butterworth filters. If one is to be

constructed from passive components it is desirable to use purely reactive
elements, otherwise the attenuation would never become zero. In practice,
capacitors and especially inductors have a certain amount of resistance so the
attenuation in the pass band will not be zero and we will have an INSERTION
LOSS. Also mismatch losses will have to be considered.

Fig 5.3(a)

- 92 -
The slope of the response is determined by the type or order of the circuit
used and the actual response by the component values. The circuit of Fig 5.3
has been found to be suitable for a low-pass filter. The mathematics of

the circuit analysis is beyond the scope of this manual but the use of
computers has simplified matters and tables of component value exist for
different conditions. Other circuits are also possible and each has its own table.

The reactive components which comprise the filter are numbered in sequence
from the input side of the circuit. The order of the filter as set by the
required attenuation at a given frequency in the stop band. Here the graph of
Fig 5.4(a) is used. For example, if we required an attenuation of 60dB at four
times the cut-off frequency o we see that a fifth order filter has to be used.
We have in fact used a fifth order filter in the Passive Filter Module 2950F
Fig 5.3(b). You can see from the circuit and module of Fig 5.3 that this
requires three capacitors and two inductors.

Fig 5.3(b)

The component values are taken from tables, Fig 5.4(b) that are normalized
and use relative values. Tables exist for signal source resistance Rs =

- 93 -
1.0(relative) and Rs = 0. This latter case corresponds to a low impedance
input. The conversion relationships are :


 
 

Creal =          
 ∙   
where Ro is the terminating resistance.

Exercise 5.1

Given that Ro = 600 and  o = 18850 rad/sec (i.e a frequency of 3kHz),

check the component values marked on the module using the table of Fig
5.4(b) for a fifth order filter. Similar graphs and tables exist for Chebyshev
responses.

We are now going to look at the frequency response of this low-pass filter,
since the practical is quite straightforward and you have met the techniques
before in Assignment 2 (tuned circuits), it will be presented as a series of
laboratory notes telling you what to do and no formal questions will be asked.
If in doubt or difficulty, consult your instructor.

- 94 -
Fig 5.4(a)

Fig 5.4(b)

- 95 -
Practical 5.1 Frequency response of a low-pass passive filter

We will now look at the frequency response of the passive filter module and
see how it performs the function of a low pass filter as indicated.

Connect up the filter to the signal source as shown in Fig 5.5.

On the 2950A module

Set the variable attenuator to 0dB.
Set control A so that the signal sweep over a range of ≈ 1.5kHz for a
timebase
input of ≈ 4 volts.

On the oscilloscope
Set the timebase to 10ms/div.
Set the Y1 amplifier to 1V/div.

Note the shape of the response without the 620 load resistor Ro. This is not
correct, so connect in Ro, increase the Y gain to 0.2V/div and you should see
a characteristic corresponding to Fig 5.1(b). Think which is the low frequency
end of your trace. Note the stop band and the pass band and that the cutt-off
frequency between them is not sharply defined. Depending upon your
oscilloscope timebase output, which controls the range of sweep, you may have
to adjust the controls on the signal source module to view the whole range of
the frequency response trace.

- 96 -
 Fig 5.5

You may notice a tiny amount of ripple in the pass band. This is very difficult
to remove altogether in practice but should be less than about 0.2dB. If you
look at the extreme low frequency end of your trace you may see something
strange happening. To examine this more closely you should disconnect the
sweep input and reduce the frequency manually. Notice that at frequencies just
below cut-off and abvove, a sine wave is displayed but at very low
frequencies you see a distroted square wave. It was this that affected the low
end of your swept display.

Q 5.1 Can you think why these effects happen?

Q 5.2 Compare the size of input with the size of the output from your
filter at a point in the middle of the pass band. Is there any
attenuation in the pass band?

Q 5.3 Estimate from the scale on the signal source module and your
calibration graph from Assignment 1 the frequency when cut off just
begun to take place.

We will not plot the response manually. We are going to use the calibrated
attenuator as in Assignment 2. Connect your digital frequency counter
to the input of the filter.

- 97 -
Q 5.4 Why connect the frequency counter to the input?

Fig 5.6

Increase the scope timebase to 200 s/div and leave the gain on 0.2 volt/div.
Set the frequency to position 2 on the scale and adjust the oscilloscope for a
stable display. To ensure that you are in the pass band, reduce the frequency
and note the minimum frequency at which a good sinewave is still displayed.

Now set the frequency to 5kHz and increase the scope gain to 20mV/div. Note
the peak-to-peak output voltage. You may find it helpful to adjust the
oscilloscope fine gain control so that this fills an exact number of divisions.
Now reduce the frequency to that minimum point you have just noted and
increase the setting of the output attenuator on the signal source module until
the same peak-to-peak value is displayed. Check that the waveform
deteriorates if you decrease the frequency. Increase the frequency slowly and
note that the output stays constant for a while, then decreases. This

- 98 -
represents the transition from the pass band to the stop band.

Reduce the attenuator setting in steps of 2dB from the next step down from
the maximum you have just obtained. Increase the frequency until the
peak-to-peak value is the same as you had noted and read the frequency from
your frequency counter. Record your results in a table as in Fig 5.6. Below
10dB, 2dB steps should suffice.

Exercise 5.2

a) Plot a graph on 2-cycle logarithmic graph paper of attenuator settingagainst

frequency (remembering this is equivalent to the response) for the whole of
the frequency axis from 1.5kHz. Plot a constant level for frequencies below
the first setting in your table.

b) Why do you think the slope of the cut-off slope on this graph is linear
when it appeared curved on your swept display?

c) From your graph measure the value of this slope and express it in
dB/octave (an octave double the frequency, i.e from 2kHz to 4kHz is an
octave).

D) Also from your graph, record the frequency at which the response is 3dB
down from the maximum in the pass band. This is the normal point for making
measurements as discussed Assignment 2 and is therefore the nominal
CUT-OFF frequency).

Active Filters

The passive filter you investigated Practical 5.1 possesses certain

disadvantages. The main ones are that the cut-off frequency cannot be varied
easily and low cut-off frequencies mean high values of inductance. This in turn
means an increase of winding resistance which affects the performance and
makes the filter heavy, large and expencive.

- 99 -
Operational amplifiers, it can be shown that filter circuits can be synthesized
using RC circuits in combination with operation amplifiers. The elimination of
inductors and the use of integrated circuits means that this circuit can be
miniaturised. Amplification is also possible. Variation of the cut-off frequency is
easier as a variable resistance can be used.

As with passive filters the slope at cut off is determined by the order of the
filter. This is achieved by cascade connection of first and second-order stages.

For calculation of the desired order we use the curves of Fig 5.4(a) as for
passive filters. Fig 5.7 shows how a fifth-order filter may be made.

It is still the number of reactive components (in this case capacitors) which
decides the order of the filter.

Fig 5.7

The following circuits have been derived. A first-order active low-pass filter is
shown in Fig 5.8. It is first order since it constants a single capacitor.
Fig 5.8(b) shows a second-order active low-pass filter with two capacitors.

Fig 5.8(a) Fig 5.8(b)

- 100 -
The '+1' sign inside the amplifier symbol indicates that it is connected as a
voltage follower or Vf. This means that the output voltage equals the input
voltage while retaining the valuable property of the operational amplifier of
taking negligible input current. As we have stated, the mathematics of active
filter theory is beyond the scope of this manual. However, in practice it is
usual to make all the resistors the same. Then the capacitor values can be
calculated to give the required response. Again, use is made of tables of
relative values. In this trainer the active filter module has been designed to be
a fifth order (5-pole) Butterworth filter, the same as the passive filter, and to
have the same response at the same cut-off frequency. However the cut-off
frequency can be lowered. The circuit is shown in Fig 5.9(a) and the module
layout in Fig 5.9(b). Variation of resistance is achieved by shorting links. Note
that all resistors must be symmetrical, or the filter characteristics will not be
correct.

Fig 5.9(a)

We are now going to examine the frequency response of the filter.

Connect the active filter into the test circuit as in Fig 5.10.

On the 2950A module

Set the variable attenuator to 0dB'
Set control A again so that the signal sweeps over a range of ≈ 1.5kHz
for a timebase input of 4 volts.

- 101 -
On the oscilloscope
Set the timebase to 10ms/div.
Set the Y1 amplifier to 1 V/div.

On the 2950E module

Initially short out all the resistor ladders so that only the basic 10
k resistors are in circuit.

Use a sweep display as you did in Practical 5.1 to display the response and
check that it is of the same from as for the passive filter. Check that this time
there is no attenuation in the pass band. (You may think there is a slight
amplification but this is due to the fundamental component of frequency being
larger than the resultant square wave - see Assignment 1.)

Fig 5.9(b)

- 102 -
 Fig 5.10

Exercise 5.3

Following the same pattern as before plot the frequency response of the active
filter at this setting. Use the same piece of graph paper as for the passive
filter and distinguish the curves by the use of colored pens. You will find that
a greater setting of the attenuator is necessary. This tells you that there is no
attenuation in the pass band through the active filter and your graph should
show the attenuation in the pass band of the passive filter.

Repeat your measurements. for two more characteristics; the middle one of the
range and the lowest one of the range. These are achieved by shorting out the
33k , 22k and 15k resistors and by removing the shorting link,
respectively. This means that for the middle setting, each resistance is (10k +
4.7k + 6.8k + 10k) ohms and for the lowest setting, each resistance is (10k +
4.7k + 6.8k + 10k + 15k + 22k + 33k) ohms. Also find the cut-off frequency
(3dB point) for each of the other settings. If you have time, of course, you can
do a complete frequency response as each setting. A suitable heading for your
table is shown in Fig 5.11. At lower frequencies you may find it better to set
the frequency using a d.c control voltage from the variable d.c socket.

- 103 -
 Fig 5.11

Plot all graphs on the same sheet of graph paper and note that the slope at
cut-off is the same in all cases. This is determined by the order of the filter.

- 104 -
Practical considerations & applications

These experiments should indicate to you the type of characteristics you can
expect from filter circuits. The two main parameters of any filter are the slope
at cut-off (sometimes called the roll-off rate) and the cut-off frequency itself.
The former is determined by the order of the filter or the number of stages in
cascade and the latter by the component values. As you have seen modern
filter design has been greatly simplified by the use of computers, and standard
circuits and tables of component values can often be used.

You should have found that the roll-off of your filters was 30dB/octave. This
is the usual unit in which it is expressed; an octave simply signifies a doubling
of the frequency. The filters are designed so that the roll-off of each stage is
6dB/octave, so for fifth-order filter you would expect 6 5 = 30dB/octave.
thus you can see how the larger order filter would give a much shaper cut-off,
of course, is a rate dependent on the cut-off frequency and cannot be given
an absolute bandwidth in terms of Hz.

You have also seen something of the difference between passive and active
filters. These can be summarized in the following tables, Fig 5.12.

Fig 5.12

- 105 -
We have started that filters have obvious uses in the selection of wanted
signals and the rejection of unwanted signals. We can differentiate between
these signals by their frequency. You have already met a band pass filter in
Assignment 3, the crystal, you will recognize this as a band-pass filter also;
the pass band being for the selection of a single spot frequency.

In this assignment we have studied low-pass filters, which pass all signals of
frequency below the cut-off frequency. One important use of these is in
speech telephony. Only frequencies below 3.4kHz are transmitted, to save
bandwidth. This nevertheless is satisfactory for speech and still allows a
certain amount of voice recognition. A low-pass filter is used to remove the
unwanted range.

You will meet another use in Assignment 8 where we study demodulation and
we find there may be unwanted high frequencies present as well as the wanted
signal; a low-pass filter removes them. When we study modulation in
Assignment 6 we will discuss how multi-channel telephony is achieved by
suitable choice of carrier frequencies. Here extensive use is made of filters
both at transmitting and receiving ends to select the required channels.

- 106 -
ASSIGNMENT - 6
- TUNED CIRCUITS

Object
To explore the characteristics of a multiplying balanced modulator and to
examine the amplitude-modulated wave.

Equipment required
● Power Console ED-2900P
0-4 volts d.c, variable metered.
+15 volts d.c
-15 volts d.c

● Oscilloscope, double beam, 5MHz calibrated

● Digital Frequency Counter, 1MHz
● Signal Generator
● High Impedance d.c voltmeter

● Modules Type 2950A and D

● Tuned Circuit Assembly Type 2950H
● Crystal Filter Assembly Type 2950J

Approximate time required

Two hours

Prerequisites
Assignments 1,2 and 3

- 107 -
Preliminary Procedure

1. Attach the console ED-2900P.

2. Connect the power supply unit(s) to the mains supply line.

DO NOT SWITCH ON YET.

3. Turn the variable d.c control knob to minimum.

4. If you are using any of modules 2950A, B, C, D or E, plug them into the
sockets on the Console or Manifold in alphabetical order. If any change is
required, it will be indicated in the assignment. The act of plugging in the
modules connects them to the power supply in the correct manner. All
sockets are identical.

5. If you are using either of the modules 2950F or G, or the Assemblies

2950H, J or K, these do not require any power and may be placed where
convenient or as directed in the assignment.

6. This completes the preliminary procedure.

Fig 6.1 Fig 6.2

Experimental Procedure

Amplitude Modulation

- 108 -
You already know that a steady signal, be it a.c or d.c, conveys no information
other than the fact that a signal is present. In order to communicate, we must
vary one or more of the properties of this signal. In the next few assignments
we will deal with methods of modulating the amplitude of a carrier signal, with
the signal we wish to communicate.
If we simply added the two signals together, one signal would vary the
reference level of the other, without affecting its amplitude. In spite of the fact
that these variations may occur at signal frequency, the receiver still sees a
carrier of unvarying amplitude.

To see how we can convey this information on a carrier wave, having a

frequency of  consider Fig 6.1. If the normal amplitude of the waveform
is V1, then V0 = V1 sin  t. If we increase the amplitude by V2, then

Vo = (V1 + V2) sin  t....Equation 1

The process of amplitude modulation then is one of multiplying the carrier

wave by the varying information signal V2.

Taking the case further, if V2 has a frequency of p/2 as in Fig 6.2(a) then it
will mould the shape of the carrier wave as in Fig 6.2(b) and Equation 1 will
now become

V0 = (V1 + V2 sin p t) sin  t...Equation 2

Theory

Equation 2 can be expanded to give

V0 = V1 sin  t + V2 sin p t. sin  t

Using the product formula, the proof of which is given in most text books on
trigonometry

 
V0 = V1 sin                  
 

- 109 -
Double Side-band Modulation (DSB)

Equation 3 thus shows that the modulated waveform of Fig 6.2(b) is the sum of
three frequencies, namely the unmodulated carrier carrying no intelligence and
two other frequencies each half the amplitude of the information signal; These
are called the upper side frequency ( +p) and the lower side frequency ( -p).

The depth of modulation or modulation index 'm' is the relationship between

the maximum value Vmax and the minimum Value Vmin of the
amplitude-modulated waveform.

    
m = 
    

Balanced Modulator module 2950D

This module contains an integrated circuit, type LM 1496 and has an amplifier
connected to the input of one channel so that the impedance of the two
channels is the same. The mimic of the front panel is shown in Fig 6.3.
N.B. The front panel legend reading 'R12,470 ' should read 'R7 10k'.

The action of the balanced modulator is to act signals. It is a standard type of

circuit and in the next assignment we will examine two simple circuits that can
produce A.M. waveforms.

- 110 -
 Fig 6.3

With the balanced modulator connected as in Fig 6.4(a) the output waveform
will appear as in Fig 6.4(b) and will be of the form

V0 = V1 sin  t. V2 sin p t. . . Equation 4

Equation 4, by the same expansion used to obtain Equation 3, contains two

side band terms of angular frequency ( +p) and ( -p) but no carrier term.
Such a signal is called a Double Sideband Suppressed Carrier (DSSC) signal.

Comparing Equation 3 with Equation 4 we can see that to obtain a normal

Amplitude Modulated signal with carrier a bias input must be applied. This can
be conveniently injected.

- 111 -
 Fig 6.4 Fig 6.5

Practical 6.1 Familiarisation with the Balanced Modulator.

To carry out the familiarisation routines set out in this practical, connect up
the apparatus as shown in Fig 6.5.

Balancing Procedure

1. Connect a high impedance 10V d.c voltmeter across the output terminals c

and c.
2. Earth input b (not through the capacitor). Connect the +1 volt socket to
input a. Adjust balance a control until connection or disconnection of
the input causes no change of output.

3. Earth input a.
Connect the +1 volt socket to input b, (not through the capacitor).
Adjust balance b control for zero output change as in paragraph 2.

4. Repeat the above, as the two contorls are slightly interdependent.

The Multiplying Function of the Balanced Modulator.

In Equation 4 and Fig 6.4 we saw that the object of a balanced modulator is to
form an output composed of a product of the input signals.

- 112 -
We will first examine this behaviour with d.c inputs. Connect:

1. Input b to the +1V socket (not through the capacitor).

2. Input a to variable d.c 4 socket.

3. Output c and earth to the voltmeter.

On the Console set the variable +4 d.c to zero.

The output should read some 5.8V or more and if you vary the 4Vd.c on the
Console, the voltage will also vary by a small amount.

Now connect the output c instead of c to the voltmeter and if you have
balanced the module correctly, when the voltage is varied the meter will vary
similarly but in the opposite sense.

To verify the multiplying function of

inputs a and b, connect the meter between
outputs c and c. This will eliminate the
constant d.c level but double the value
of the output. If your voltmeter reads
negative, reverse the leads.

Now take meter readings for different

values of variable d.c as in Fig 6.6. As
the input divides by 4, the voltage will
go up to 1volt. Subtract the reading for
zero input from all readings. Fig 6.6

Exercise 6.1

(a) Plot a graph, on linear graph paper, of the corrected d.c output volts
against d.c input volts, from your table in Fig 6.6

(b) Is the graph a straight line? Does this prove that the multiplication
process is linear?

- 113 -
(c) Remembering that the other input is +1 volt, calculate the value of the
constant in the equation a x b = c x constant.

Q 6.1 What do you think you will get if you reversed the inputs, i.e.
attempted to measure b x a?

We will see if you are right. Set the variable input to 0.5 volt so that you get
an output as in your table above. Now swap over a and b inputs.

Q 6.2 Is the output the same as before? What does this prove?

Now we will see what happens if one input is negative. We have no negative
reference voltages on our module, but we can easily produce the same effect
by using the complementary input a.

Q 6.3 What does the output voltmeter read? Does this correspond with what
you would expect from -a x b = ? (Don't forget to subtract the zero
reading).

This is sometimes called a FOUR QUADRANT MULTIPLIER, since it gives the

correct sign for the output for all four combinations of negative and positive
inputs.

Practical 6.2 Operation of the Module 2950D

From our discussion of amplitude modulation we have seen that to carry out
this action we have to multiply two signals with a bias applied to one of them,
as in Equation 3.

The two terminals (a) and (b) to the module 2950D form the inputs to the
multiplying circuit, while terminal (a) is the bias input as shown in Fig 6.7.

To obtain an amplitude modulated waveform.

On the module 2940A

Set the variable attenuator to -4dB.
Set the frequency control C against the dot 465kHz.

- 114 -
On the module 2950D:

Connect a lead between the VAR.d.c socket and the (a) input.
Set the variable d.c input to 1V.

On the oscilloscope:
Set the timebase to 1ms/cm and amplifiers to 0.1V/cm.
Connect Y1 input to (c) on module 2950D.
Set to a.c.

On the function Generator:

Set for a 1-volt peak-to-peak 300Hz sinewave output (use the oscilloscope
to make the measurement).

Ensure that the Y1 waveform is correctly balanced according to the Balancing

Procedure.

Q 6.4 Is the waveform displayed the same as in Fig 6.2(b)?

Exercise 6.2(a)

Sketch what you see.

Q 6.5 Considering the mimic on module 2950D what is the simple action
required to remove the carrier and turn the output into a DSSC
waveform?

If we examine Equation 3 and compare it with Equation 4, we see that all that
is needed to produce a DSSC waveform, is to remove the bias from the
Balanced Modulator on the module 2950D.

Remove the connection between the VAR.DC socket and the (a) input.

Exercise 6.2(b)

Sketch what you see and measure the peak-to-peak voltage.

Practical 6.3 Measurement of the Modulation Index.

- 115 -
In this practical we will study the depth of modulation of a DSB modulated
waveform.

Check that the apparatus is still connected as in Fig 6.7, together with a

connection on module 2950D of the VAR.DC output to the a input.

Fig 6.7

Carry out the initial instructions of Practical 6.2 but set the Function Generator
and Console variable d.c outputs to zero. Increase the variable d.c output to
0.5 volts, and observe Y1.

Q 6.6 What do you see on the oscilloscope? Does its peak-to-peak output
voltage correspond approximately with what you would expect from
your calibration graph?

SLOWLY increase the Function Generator output amplitude. Note what happens
and continue increasing until the peaks of the modulated output come together.

Connect the Y2 channel of your oscilloscope (0.2V/cm) to measure the signal

input amplitude.

Q 6.7 How does the peak signal voltage which makes the peaks of the
modulated output come together, compare with the variable d.c.
input?

- 116 -
Q 6.8 What value of modulation index does this correspond to?

Increase the signal input and note the effect of modulation depths greater than
100 %. THis is not normally used. If you increase the input too much, limiting
will occur in the integrated circuit.

Q 6.9 What signal input amplitude would be required for a modulation index
of 50%?

Set up this input amplitude. Check the value of the Modulation Index from the
oscilloscope display shown in Fig 6.8.

In practice the usual case is where >>p as we have here, and we cannot
distinguish the cycles of carrier in our oscilloscope trace. In Fig 6.8 we have
shown the individual cycles for clarity, such that = 10p. THis will also make it
clear why the modulation you see is called the ENVELOPE of the waveform.
We will now show this in practice.

Increase the input signal frequency of your function generator to 46.5kHz.

Increase the oscilloscope timebase to 5 s/cm. If you are still triggering from
the input signal p, you should have a display as previously. However if you
now trigger from the output itself on the Y1 channel, you should have a
display more like Fig 6.8 and be able to count the cycles of the carrier.

Q 6.10 How many cycles of carrier are there per cycle of modulation
envelope?
Does this correspond to the ratio of the frequencies?

You can see clearly now how the amplitude of the carrier is modulated by the
input signal. return the signal frequency to 300Hz and the timebase to 1ms/cm.
The same thing is happening still, but you can't see it.

Select in turn triangular and square-wave signals from your function generator
at 300Hz. Note especially that the straightness of the triangular wave trace
gives an idea of the linearity of the modulator.

- 117 -
 Fig 6.8

Exercise 6.3

Sketch the amplitude-modulated outputs of triangular and square waves for a

modulation index of about 50%.

Display methods of examining the A.M. waveform.

As an extension to this Assignment there are some further practicals in the

Application Note on display methods of examining the A.M. waveform. It shows
how to detemine the presence of distortion in the modulation process and how
to use a swept frequency display to pick out the A.M. frequency components.

Practical considerations
& applications

Multiplying function of the Balanced-modulator module

In the first two practicals we were concerned to show that multiplying two
waveforms  and p together produced two side frequencies  ± p and, by
adding a d.c component to the signal, the carrier could be made to appear in
the output. This means that this circuit can also be used to provide voltage
control of the amplitude of an a.c. signal. If you have done Assignment 4 you
will note that this is an alternative, more linear method, to the voltage control
of gain described there.

- 118 -
Modulation Index

In practical 6.3 we started with a fixed carrier and showed you how the signal
varied the amplitude and defined the MODULATION INDEX m as the ratio of
signal amplitude to unmodulated carrier amplitude. You should appreciate that
the maximum value of m is 1, corresponding to the point where the peaks
meet and the carrier is reduced to zero. Because of the balanced modulator,
attempting to increase m in fact produces a negative signal. You will find later
that other types of modulator simply cut off. In both bases, the demodulated
output will suffer distortion. When m is less than unity, both positive and
negative peaks of the information signal cause the carrier amplitude to reach
maximum and minimum intensity levels. THis forms the symmetrical waveform,
both upper and lower halves showing identical characteristics of the signal.
This is referred to as the ENVELOPE and we shall see that its presence
creates an easy method of detection. Notice that no definition of modulation
index is possible for a suppressed carrier wave.

Double Sideband Modulated Transmission

In the appendix on Amplitude Modulation we show how inefficient is this form

of transmission. The reason that there are so many full carrier-wave A.M.
broadcasting stations is that this was the first system to be developed. In this
assignment we artificially produced it by adding a d.c component to our original
signal, but you will see in the next assignment that other types of modulators
produce it directly. Much capital is invested in transmission equipment.
However, receivers are simple and although transmitters are expensive and
wasteful of power, this system is economical where a single transmitter serves
many thousands or even millions of receivers, such as in medium wave
broadcasting.

Double Sideband Suppressed Carrier Transmission.

When cost is not of prime importance, but making the best use of a limited
transmitter power is, for example when a single transmitter serves only a few
receivers, suppressed carrier modulation is preferred. Examples occur in
military, industrial, and amateur radio communication systems. The balanced

- 119 -
modulator circuit as employed here is easily used, but demodulation is more
difficult.

As Fig 6.4 shows, the envelope of the suppressed carrier wave is not simply
that of the original waveform, some carrier is required for demodulation. This
can be derived from the side frequencies or generated locally. In this case,
transmission of a small fraction of the original carrier helps the receiver. This
is called a vestigial carrier or pilot carrier. We will discuss the merits of the
various methods of dealing with DSSC when we cover demodulation and
detection.

Although the DSSC method saves power, the bandwidth required is still twice
the highest modulation frequency. In crowded transmission media, such as
occurs in aircraft, radio telephone and some amateur radio bands, this is a
disadvantage.

Single Sideband Transmission

It is apparent from the equations that each sideband carries all the information
present in the original signal, so half the bandwidth can be saved if only one
sideband is transmitted. This is known as SINGLE SIDE BAND transmission
(SSB). There are several ways to produce this. The phasing method of SSB
generation develops two sets of sidebands without carrier, but with the
sidebands out of phase. Addition or subtraction of the sidebands then removes
the upper or lower sideband as required. This method can be employed at
radio frequencies as high as a few megahertz. A simpler method, called the
filter method, generates DSSC then discards one sideband by the use of a
narrow band filter. Many filters of satifactory narrow pass band are suited only
to relatively low frequencies so that modulation and separation of the
sidebands is usually carried out at frequencies below one megahertz, and the
output sideband is translated to the transmission frequency.

Heterodyning

This process of frequency translation is known as HETERODYNING and one

method (there are others) is to use modulation in a balanced modulator.

- 120 -
Suppose we have a frequency of 1000kHz that we wish to translate to 465kHz.
If we have an oscillator at 1465kHz and we feed the two into a balanced
modulator the output side frequencies will be (1465 1000)kHz. The 465Khz
LSF is our required output and the 2465kHz USF is called the image channel.
This is rejected by a filter, or the tuned circuit used to pick out the required
465kHz signal. If sidebands are present on the signals, they too will be
translated into the required frequency band. This process is commonly used in
radio receivers, superheterodyne receivers, and will be discussed fully in
another assignment, together with consideration of the frequencies and
bandwidths employed.

Frequency Division Multiplexing

The method of frequency translation is very useful when it comes to sending

many sets of information along the same channel, as in telephone circuits
between exchanges. If you have done Assignment 5 you will know that
intelligible speech can be transmitted with a very small bandwidth. In fact the
bandwidth chosen for telephones is 300Hz to 3400Hz. This is known as the
BASE BAND. Firstly, different base bands (say - 4kHz for ease of description)
are separately modulated with twelve different carrier frequencies in the range
64 to 108kHz. Each of the lower sidebands is selected by a crystal filter. The
total range of the group of twelve sidebands is from 60 to 108 kHz. Five
separate groups are then modulated onto a carrier frequency between 420 and
612kHz and again the lower sideband is selected. This represents sixty
channels which together with nine more 60-channel super-groups is further
translated into a frequency band suitable for tansmission along a cable.
Amplifiers in the cable make good the losses and at the receiving end, filters
are used to sort out the channels for demodulation.

The frequencies are all produced by dividing down a master oscillator for
stability. This example showing how 600 telephone calls may be sent along a
single cable is illustrated in Fig 6.9 (see overleaf).

The frequency range required of the cable is 600kHz to 2.5MHz. This is

referred to as CARRIER TELEPHONY and is a FREQUENCY DIVISION
MULTIPLEX system.

- 121 -
- 122 -
ASSIGNMENT - 7
- SIMPLE AMPLITUDE MODULATORS

Object
To examine the use of a correctly biased transistor amplifier and also a diode
ring bridge circuit, as amplitude modulators.

Equipment Required
● Power Console ED-2900P
±15 volts d.c

● Oscilloscope, double beam, 5MHz, calibrated

● Signal Generator

● Modules Type 2950A, B, F, G.

● Tuned Circuit Assemblies Type 2950H : 2ea

Approximate Time Required

Two hours

Prerequisite Assignments

1, 2, 4 and 6

- 123 -
Preliminary Procedure

1. Attach the console ED-2900P.

2. Connect the power supply unit(s) to the mains supply line.

DO NOT SWITCH ON YET.

3. Turn the variable d.c control knob to minimum.

4. If you are using any of modules 2950A, C, B, D or E, plug them into the
sockets on the Console in alphabetical order. If any change is required, it
will
be indicated in this assignment. The act of plugging in the modules connects
them to the power supply in the correct manner. All sockets are identical.

5. If you are using either of the modules 2950F or G, or the Assemblies

2950H, J or K, these do not require any power and may be stood
where convenient or as directed in the assignment.

6. This completes the preliminary procedure.

Experimental Procedure

You saw in Assignment 6 that a double sideband suppressed carrier (DSSC)

amplitude-modulated wave could be produced by mutiplying together the carrier
and information signals. The circuit is thus known as a balanced modulator.
When you added a d.c component to the original signal p, this produced the
full AM wave with carrier  and side frequency  ± p. You learned the
definition of modulation index and saw that it governs the relative amplitudes
of carrier and side frequencies.

In this assignment we are going to show how a transistor connected as a

common emitter circuit can be used as an amplitude modulator and then how
diodes connected to form a ring bridge can operates as a balanced modulator.
To examine the performance of these circuits we shall be using the technique
that we have already learned.

- 124 -
The Transistor as an Amplitude Modulator

From Assignment 4 and you other studies of transistor you will know that the
input cahracteristics show in fig 7.1(a) will have a higher quiescent operating
point the lower the value of the emitter resistor RE in fig 7.1(b). Taking a
value of RE such that operating takes place at Q1, there will be a high
standing current flowing in the emitter circuit. Due the low value of carrier
injected at the single input to the base will be simply added to this voltage and
because the total variation in Vbe is small, operation will take place over a
small section of the characteristics.
The base current is therefore all most linearly dependent on the value of Vbe.
Amplification will take place by normal transistor action, and the
output will be proportional to the sum of the inputs, recognisable by the peaks
going in the same direction.

Fig 7.1 (a)

- 125 -
 Fig 7.1 (b)

Increasing RE moves the quiescent point to Q2. This reduces the loading effect
and more carrier becomes available at the emitter. Let us see what this
implies. The input characteristic can be redrawn in fig 7.2 as a straight line
approximation. Portion OA is the forward conduction part; OB is the reverse
blocking part.

Fig 7.2

When the carrier only is applied to the emitter, increasing its peak- to-peak
voltage beyond OC will cause the negative portion behind O to be cut
off. Since the emitter is not directly
decoupled, as explained in Assignment

- 126 -
4, the resultant d.c level acts against
the bias and moves the quiescent
point nearer to O. The output thus
tends to limit at a value of OC.
Further increa- sing the carrier input
to the emitter varies the bias further
towards B, so that conduction occurs
only for samll part of the carrier
cycle. You will recognize from your
other work that these conditions
correspond to class B and class C
operation respectively. When a signal
is added to the base and the carrier
is large enough, the sum of carrier
and signal will only have part of the Fig 7.3
waveform in the conducting portion
A of the charateristic.

Hence only this part will be amplified, as shown in fig 7.7.Carrier current can
flow when the signal varies the bias during its cycle. The output therefore
consists of pulses of carrier, with an envelope equal to the modulating signal.
If the primary of a suitable transformer is connected in the collector of the
transistor waveform envelope will appear as in fig 7.3

Diode Ring Bridge Balanced Modulator

A diode only connects when forword-biased. It can therefore be considered as

a switch, conducting or nor depending upon the polarity of the bias voltage.
Typical bias voltage charateristica of Germanium and Silicon diodes are shown
in fig7.4. If we use our carrier signal as the bias voltage, provided it is much
greater than the information signal level, it will switch a diode on and off and
allow the signal to pass or not, at carrier frequency.

- 127 -
 Fig 7.4 Fig 7.5

To form a suitable circuit, as in fig 7.5, four silicon diodes are used, arranged
in a RING BRIDGE. The carrier is fed in, and the output taken out through
transformers. These provide isolation from the signal which is fed in through
the centre taps on these transformers. Notice that the diodes are arranged in a
ring, not in the usual full-wave bridge configuration. When the amplitude of the
carrier is large compared with the modulation signal, it can be considered as a
square-wave switch, switching the diode alternately forward and
reverse-biased at the carrier frequency. It is assumed the carrier is greater
than the forward voltage drop of the diodes.

The output is then caused to vary as the product term for the valanced
modulator in Equation 4 of Assignment 6 where:

V0 = V1 cos  t. V2 cos pt. Equation 1

where /2 = carrier frequency and P/2 = signal frequency. This will result in a
suppressed carrier double side-band waveform.

Practical 7.1 Use of the transistor as a modulator

Initially we are going to demonstrate how modulation can be produced by the

modulating effect of an input the carrier wave into the emitter and the signal
wave into the base of a transistor.

Connect up the layout of fig 7.6

- 128 -
On the 2950A module:
Set the manual frequency control C to about 465kHz.
Set the variable attenuator control to 0dB.

On the Function Generator:

Set the output to a 300Hz sine wave.
Set the amplitude control to zero.

On the Oscilloscope:
Set the timebase to 1s/cm and the amplifier to 0.5V/cm.
Switch to d.c coupling.
Set the trace so that its d.c zero level is at the top of the screen.
Switch on the Console power. Adjust the oscilloscope for a stable display

Fig 7.6

Q7.1 What do you see? What is its d.c level?

Do the peaks reach the d.c zero level?
Q7.2 Why is this output smaller than you had in Assignment 4?

Measure the carrier voltage input at the emitter with your oscilloscope. Now
decrease the timebase to 1ms/cm. Note that the carrier is still present.
Connect a lead from terminal A on the module2950B in fig 7.6 to terminal B,
the output on the Function Generator.
Monitor the Function Generator output on Y2 input of the oscilloscope, slowly
increase its output to about 1 volt pk-pk.

- 129 -
Q7.3 What happens to the amplifier output? Sketch what you can see.

Q7.4 What happens if you increase the signal output (to the base) even more?

Now set the input exactly 0.6 volt pk-pk and without adjusting anything else,
increase the value of the emitter resistor to 10,620 ohms in the 2950B module
by connecting the lead from terminal C to terminal D on fig 7.6.

Increase the oscilloscope Y gain to 0.2 volt/div so that you can see the trace
better.

Q7.5 Sketch what you can see? How is it different from before?

Q7.6 What waveform would you get if you took the average value at a
large number of points on the waveform? Has this average waveform
got a d.c level present on it?

Q7.7 What happens when you reduce the signal input to zero and when
you increase it over about 1V pk-pk?

Q7.8 Think carefully. Can you see what is happening?

When the signal input is increased, because of the phase reversal of the
common emitter amplifier, the top peak's flattened as the transistor becomes
cut off, i.e no current flows.

If it was possible to increase the carrier, the separation between peaks would
be increased and more the cycle would be cut off. However the available
carrier level is limited because of the loading effects. For greater detail read
the initial section to the assignment.

Reduce the Function Generator output to zero so that only the carrier is
displayed. Slowly reduce the carrier amplutude using the variable attenuator
on the 2950A module. Do you notuce the limiting above a certain value and the
linear amplification below this vlaue? You should find this value occurs at a
setting of about -10dB, This corresponds to the amplitude of OC in fig 7.2.
Reduce the carrier to zero by tuning the attenuator fully anti-clockwise.
Now apply a signal of 0.2V pk-pk. You will see the output signal with a d.c

- 130 -
level on it. Slowly increase the carrier input. Notice how at first the two
signals are simply summed corresponding to linear operation on part OA of the
characteristic in fig 7.2. Further increasing the carrier causes the upper peak
to limit as the negative parts of the input waveform occur on the section OB
of the characteristic.

Practical 7.2 Formation of the modulated waveform

We are now going to see how the output can be made to represent a
modulated waveform by increasing the carrier until the upper section of the
output waveform is flat. (Remember that you are looking at the voltage
developed by the output current across the collector load resistor and there is
a phase change between input and output).

Keep the layout of Practical 7.1

Increase the signal input so that part of the other edge of the waveform now
enters the conduction section OA in fig 7.2 of the charateristic.
Continue increasing the signal input until just before the cycles of full
conduction begin to limit. Increase the carrier input and watch the negative
peaks at the top disappear. Observe that further increasing the carrier has no
effect on the output; it simply moves the negative cut-off section of the input,
marked D in fig 7.3. This is further along OB, corresponding to class C
operation throughout the whole cycle. The output corresponds to half of the
modulated waveform you had from the modulator in Assignment 6, so it would
appear to represent modulation.

Practical 7.3 Frequency and d.c components of modulated waveform

If you observe the waveform you should

be able to see that it includes, in addition
to the other frequency comp- onents
needed to represent the pulses of carrier
in the output, a component of the signal
frequency and a d.c level. You can Fig 7.7

- 131 -
quickly verify this by passing the waveform through the low-pass passive filter
module 2950F, so the carrier is removed.

Connect up as in fig 7.8. Do not alter the other setting.

Fig 7.8

By using the transfer curve plotted in Assignment 4 we can see how

modulation is achieved. The characteristic of the curve in fig 7.2 is not simply
OA, it is BOA, because of the curvature at A. This is similar to the
characteristics of a diode.

As a check set the carrier to 0dB and increase the signal until just before
limiting occurs. Measure the peak-to-peak output voltage. You should be able
to see that this occurs when the pk-pk input is twice OC, that is four times
OQ2.

Exercise 7.1

a) Look at the actual dynamic transfer curve you plotted in Assignment 4 and
read off the collector current corresponding to 4 x OQ2. Calculate the

- 132 -
 voltage this current drops across the collector load of 620 .

b) Does this compare with the output on the oscilloscope?

Normal amplification occurs in the transistor. We could get more amplifi-

cation if we had our quiescent point higher up the curve, but then we would
need more carrier available. The value of RE chosen thus represents a
compromise, but displays all the charateristics. This effect also happens with
RE at the maximum of 57.6k on the 2950B module but as you would expect
the output is smaller as the amplificaiton is smaller.

Practical 7.4 Distortion in the modualtion process

When the input voltage to a device is connected to the X input of an

oscilloscope and the output to the Y input, the resultant display will be a
straight line if no distortion is immediately evident by curvature of the
displayed line. An advantage of this system is that its operation is not
dependent upon a particular input waveform.

In the case of a modulated output the

apparent straight line is found by the
envelope edge. As fig 7.7 displays only
half the modulation waveform, the disp-
lay, as shown in fig 7.9 will only pro-
duce half a trapezium.
Fig 7.9

Retain the layout of fig 7.8, but:

On the oscilloscope:
Switch to X mode of operation.
Connect the lead from the Y2 input to the X-input (instead of to the
trigger input)

Q7.9 Do you notice any distortion present?

(is the sloping side straight?)

- 133 -
Q7.10 How can we get rid of the unwanted components in this waveform?

Application Note 3 on Display Methods gives a full explaination on how this

trapezoidal dispaly is obtained.

Practical 7.5 Retrieving the full modulation waveform.

If we pass our output signal through a transformer we will transform the

half-wave modulated waveform into a full-wave modulated waveform. Also
by using the tuned circuit of module 2950H we are able to eliminate unwanted
harmonics.

Connect up the layout of fig 7.10

On the oscilloscope:
Set the Y gain to 1V/cm and the timebase to 1ms/cm.

On the 2950A module:

Keep the manual frequency control C at about 465kHz.
Set the variable attenuator control to 0dB.
On the Function Generator:
Keep the output as a 300Hz sinewave with zero amplitude.

On the 2950H module:

Set the tuning control to give a maximum output display.
Then slowly increase the signal amplitude.

- 134 -
 Fig 7.10

Q7.11 Sketch what you see. Does this correspond to the full
modulation waveform you had in Assignment 6?

With your tuned circuit output on the Y-channel and the signal input on the
X-channel. switch your oscilloscope to X-Y display. Increase the signal input
so that over modulation occurs.

Q7.12 Are the sides of the display straight?

Does this support the theory?

Q7.13 Sketch what you see. To what does the flat section of the display
correspond?

Increase and decrease the signal input frequency and note the opening out of
the sides of the display caused by phase shift.

Q7.14 What cause phase shift at high frequencies?

Q7.15 What causes phase shift at low frequencies?

- 135 -
Return to timebase operation at 100 s/cm. Select a 0.6V pk-pk square-wave
input at a frequency of 3kHz.

Q7.16 Sketch what you see. Can you think why the distortion is caused?

Practical 7.6 The diode ring bridge balanced modulator

We have seen how the mutiplying function of the Balanced modulator in

Assignment 6 produces an output having a suppressed carrier wave. In the
initial discussion we have already explained how a simple balanced modulator
can be made from a diode ring bridge.

Fig 7.5 shows a type od circuit that can carry out balanced modulation. For
the diode ring bridge we shall use the arrangement of the diode module
2950G. The transformers can be conveniently provided by the coil of the two
tuned circuit assemblies 2950H.

As each tuned circuit has to be tuned for maximum output, initially connect up
the layout of fig 7.11

Fig 7.11

On the 2950A module:

Set the frequency control C to about 465kHz, that is the dot position on
the scale.
Set the variable attenuator control to 0dB.

- 136 -
Switch on and tune the control of the tuned circuit 2950H module for maximum
output. Now connect up the main circuit as in the layout of fig 7.12.

Do not touch the 2950A and 2950H (A) module settings.

On the Function Generator:

Set to a 2V pk-pk 300Hz sinewave output.

On the oscilloscope:
Set the amplifiers to 0.5V/cm and the timebase to 1ms/cm.

Fig 7.12

Position the two tuned circuits at right angles to avoid any mutual inductance.
Switch your oscilloscope to 0.2 volt/cm and reduce the input signal amplitude
from your function generator until the limiting just disappears.

Q7.17 Sketch what you see. Does this represent modulation? If so, what
kind?

Q7.18 Why did limiting occur when the input was quite high?

- 137 -
Q7.19 What happens if you now reduce the carrier level?

Reducing the carrier level means that it can no longer be considered as a

square-wave switching function and distortion occurs. Reduce the carrier to
zero and display the signal input on the Y2 of the oscilloscope at 0.2 volt/ cm.
Reduce Y1 to 0.5 volt/cm so that both traces are on the screen. Reset the
variable attenuator on the 2950A module to the 0dB position.

Q7.20 Why does the displayed input signal become distorted?

Practical 7.7 Distortion in the ring bridge signal

To note the effect created by the distorted signal, switch your oscilloscope to
the X-mode of operation so that you obtain a trapezoidal display.

Use the sinewave monitor output on your function generator as the X input, or
alternatively pass the distorted signal through the Low Pass Passive Filter Unit
(refer to Assignment 5) as a pre-input to the X channel. Either method should
give you a good trapezoidal display. Increase the signal input and note the
effect of limiting. Decrease the carrier input and note the non-linearity.

When you increased the signal too much there was limiting on the peak - the
combined voltage on negative signal peaks was not sufficient to forward bias
the diode. When the carrier input was reduced, the assumption that it is a
square wave became no longer valid, as we are passing it through a tuned
circuit. The sine wave thus formed did not provide a sharp enough transition
from forward-to-reverse bias amd distortion occured. Reducing the carrier
level caused even more limiting.

Referring to fig 7.5, let us say that on positive half cycles of the signal input,
point A is positive with respect to point D. During positive half cycle of
carrier, point X is positive with respect to point Y.

Exercise 7.2

a) Fig 7.13 shows one cycle of the input waveform, together with 10 cycles

- 138 -
 of carrier waveform. On graph paper graphically multiply the two signals
together on the same axis as shown in fig 7.13(c).

b) State the diodes that are forward -biased during the time period 0-1, 1-2
and 10-11.
c) Is the output the same or of
opposite polarity with respect
to the carrier input for the
time period 0-1?

d) In the time period 10-11 is the

output the same or of opposite
polarity to that during the time
period 0-1?

e) Does your graphically construc-

ucted output look the same as
the displayed output on the osc-
illoscope?

f) Why is there no carrier persent

in the output waveform? Fig 7.13

Practical consideration
& applications

The purpose of this assignment was not to analyse specific modulators in detail
but to see how the non-linearity of a device can be incorporated in a circuit to
produce modulation. Many such circuits can be designed and you should be
able to easily understand their operation, with a grasp of the basic principles.

Active device modulators

Modulation can be achieved with any non-lineary transustor circuit; even any
slight non-linearity which may appear in the characteristics, say a class A

- 139 -
amplifier will cause slight modulation. When the transistor is biased close to
the very sharp non-linearity in the characteristic at the origin, the efficient in
increased and the modulation is easily visible.
It is important that you realise that modulation actually taked place in the
base/emitter junction of the transistor. The two signals are summed as Vbe,
and the sharp non-linearity at the origin of the input charateristics causes the
base current Ib to contain the additional frequency components. This is then
amplified by normal transistor action and we observe the variations in collector
current. Some references call this linear modulation. It is not essential that
the amplification be linear as any of the extra frequency components
introduced will be rejucted by the tuned circuit. The exceptions are any
sidebands of the harmonics of the signal frequency which fall within the
bandwidth of the tuned circuit, and this will introduce distortion. At the
frequency of 300Hz used, any 2nd and 3rd harmonic distortion introduced will
be very small, until limiting or cut-off is reached.
Modulating can either occur at low or high power levels. Both methods have
advantages, and they are set out in the block disgram shown in fig 7.14.

(a) low power modulation (b) high power modulation

Fig 7.14

- 140 -
At low power levels, little power is associated with the carrier or information
signal, consequency the output power of the modulator is low. Power
amplifiers are necessary to boost the amplitude modulated signal up to the
desired output level. Wideband power amplifiers are required to preserve the
side frequencies of the modulated signal. Class C amplifiers cannot be used.
Several stages of amplification may be necessary.
In high level modulation, high gain narrow band amplifiers can be used on the
carrier to achive the desired power level. However the information signal
must also be at a higher level and this needs a separate power amplifier.
High efficiency class C operation can be used.

Passive devices as modulators

Passive devices can also be used as modulators - the diode ring bridge is an
example of this. The circuit used in this assignment produced a DSSC
waveform. However, if in fig 7.5 in the inputs A and B, one terminal is taken
from a signal source and the other is taken from a variable d. c source then
the carrier waveform will appear in the output. The carrier can be considered
as a squarewave switching function operating the diodes as switches so that
the transition from the conduction to non-conducting states moves the
operating point over the most sharply curved portion of the characteristic,
resulting in the maximum efficiency. However the carrier amplitude must be
made much greater than the signal amplitude in order sides of the
characteristic. If this is not so, distortion occurs.

The ring bridge modulator is commonly used in line telephony as it is a simple

method of generating the DSSC waveform required, as discussed in Assignment
6. It is also small and highly reliable requiring minimum main- tenance as it is
a passive circuit and it has no power supply requirements.

- 141 -
ASSIGNMENT - 8
- DETECTION AND DEMODULATION

Object
To investigate the process of demodulation and to exanmin the performance of
some AM detectors.

Apparatus required
● Power Console ED-2900P
0-4 volts d.c, variable, metered.
+15 volts d.c
-15 volts d.c

● Oscilloscope, double beam, 5MHz, calibrated.

● Signal Generator
● High Impedance d.c voltmeter

● Modules Type 2950A, B, C, D, E, F and G

● Tuned Circuit Assembly Type 2950H

Approximate time required

Two hours

Prerequisite Assignments
1, 2, 4, 6

Theory Assumed
Diode characteristics

- 143 -
Preliminary Procedure

1. Attach the console ED-2900P.

2. Connect the power supply unit(s) to the mains supply line.

DO NOT SWITCH ON YET.

3. Turn the variable d.c control knob to minimum.

4. If you are using any of modules 2950A, C, B, D or E, plug them into the
sockets on the Console in alphabetical order. If any change is requir-ed,
it will be indicated in this assignment. The act of plugging in the
modules connects them to the power supply in the correct manner. All
sockets are identical.

5. If you are using either of the modules 2950F, or the Assemblies 2950H
J or K, these do not require any power and may be stood where convenient
or as directed in the assignment.

6. This completes the preliminary procedure.

Experimental Procedure

You have investigated in previous assignments the amplitude modulation which

we said was necessary to impress a signal onto a carrier suitable to the
transmission channel used, The question arises as to how to recover the
information at the receiving end.

Recall the two types of AM wave which you have studied previously. These
are shown in fig 8.1.

Fig 8.1

- 144 -
Q8.1 In which of these two does the envelope most closely resemble the
original signal?

Q8.2 From which do you think it will be easiest do recover the original
signal?

Q8.3 Do you remember the display method you used with the tuned circuits
in Assignment 2?
Can you see how this can be applied to demodulation?

Let us see if you are correct. You must first obtain a modulated wave as you
did in Assignment 6. First balance the modulator according in the instructions
in Assignment 6.

Connect up the circuit of fig 8.2.

Fig 8.2

On the oscilloscope
Set the Y-amplifier to 0.1V/cm.
Set the timebase to 1ms/cm.

On the 2950A module

Set the manual frequency control C aginst the dot 465kHz.

- 145 -
On the Console power supply
Set the variable d.c to about 2V.

On the 2950H module

Tune for maximum output on the oscilloscope.
Monitor the d.c input and set it at 0.5V.
Monitor the tuned circuit on the oscilloscope.

On the function generator

Set the waveform to sinewave.
Set the frequency to 300Hz.
Set the amplitude to zero.

Slowly increase the output amplitude to about 1 volt pk-pk, when an AM

waveform with a modulation index of just under 100% will be displayed.

Q8.4 Is he signal information contained equally in both halves of the

wave-form?

Q8.5 How can we remove one half of it?

Detector Module2950C

Obviously we can use a diode. Such a diode is available on the Detector

Module 2950C. If you look at the mimic of the module in fig 8.3, you will see
that it contains a Germanium diode D1 together with all the resistors and
capacitors that you will need to construct a complete detector. There are two
further diodes D2 and D3 that you will use when you study Fm.
You will already have used the d.c amplifier on the module during your
experiments in Assignment 4 but you will notice that a 0.5V delay can be
placed in the input and you will further use this amplifier to study AGC later in
this assignment.

The section marked 'Audio Amp' is an a.c amplifier for processing the signals.
You will find out more about these circuits as we progress.

- 146 -
 Fig 8.3

Practical 8.1 Diagonal Peak Clipping

To see how we remove one half of the waveform, connect up the 2950C
module as shown in fig 8.4 to the layout of fig 8.2. Do not disturb the way
you have set up the modulator.

You can now monitor the input and output to the diode on the both beams of
the oscilloscope.

Fig 8.4

Q8.6 Why do you think we have taken our output across only half the
tuned circuit?

- 147 -
Q8.7 Why do you think we are using a Germanium and not a Silicon diode?
Sketch the output of your Y2 trace.

Q8.8 How can we remove the carrier frequency component still remaining?

Correct, a capacitor across the load resistor of the diode will remove the high
frequency carrier component.

Connect in turn the 1nF, 5nF and 20nF capacitors across the resitors, as
shown dotted in fig 8.4 and sketch the outputs you see with the 1nF and 20nF
capacitors. Notice the filtering action of the capacitors and keep monitoring
the diode input waveform on Y1.

Exercise 8.1(a)

With C1 = 20nF, measure the peak-to-peak value of the modulation on your

diode input waveform (shown as Vi in fig 8.1(a)) and measure the peak-to-
peak output voltage V0 across the capacitor.

Calculate the detector efficiency from the formula


 =  . 100%
 

Now remove the 4.7k resistor and connect the 1nF capacitor so that the
diode load is 100k in parallel with 1nF.

Q8.9 Is the output waveform the same as before?

Exercise 8.1(b)

Repeat Exercise 8.1(a) to calculate the detector efficiency.

Q8.10 Is this efficiecy greater or less than before? Why do you think this
is so?

- 148 -
Q8.11 You have seen that reducing the value of capacitance and increaseing
the value of resistance produces no change in the shape of the out-
put. What property of the CR circuit do you think is of importance
for proper detection?

You should be able to see that it is the product of CoRo that is important.
This is called the TIME CONSTANT of the circuit and is denoted by = CoRo
and is measured in seconds. If is too small, we have seen that there is still
some RF ripple left on the output. What happens if is too large?

Connect the 5nF capacitor across the 100k resistor as the diode load.

Q8.12 Do you notice anything odd about the output waveform? Sketch what
you see.

Reduce the amplitude of the input signal and measure the value of modulation
index m at which the distortion disappears. If you are in any doubt as to
how to do this, refer to Assignment 6, fig 6

Return m to 100% and decrease the frequency slowly, noting when the
distortion disappears. Then increase the frequency slowly to 1kHz and note
the increase in the distortion.

Return to m = 100% at 1kHz and reduce the time constant by using the 1nF
capacitor in parallel with the 100k resistor. Notice that the distortion has
decreased. Now reduce the signal frequency to 300Hz and the distortion
should disappear.

Finally select a square wave input at 300Hz and set m = 100%. With the 5nF
capacitor notice the trailing edges of the demodulated waveform. Vary m and
f (frequency) and notice similar effects to those above.

Q8.13 By deduction from these observations, can you explain what is

happening?

Q8.14 Calculate the value of time constant for R = 100k and 5nF, and
also for R = 100k , C = 1nF, and for R = 4.7k , C = 20nF.

- 149 -
You should be able to see that for a given value of time constant the
distortion increases with frequency and value of m. This type of distortion is
called DIAGONAL (or TANGENTIAL) PEAK CLIPPING. Let us see if we can
explain how it is caused.

Fig 8.5 shows the circuit and the waveform you should have sketched. The
input is shown at (b); (c) also shown how the diode D removes half the
waveform by rectification; (d), (e) and (f) show the action of R1 and C1.

When the modulation signal causes the carrier to increase (portion A of the
waveform of fig 8.5(e), capacitor C1 charges up through the diode and the
output waveform follows the input. When the modulation signal decreases, C1
can only discharge through R1 as the reverse resistance of the diode D is
high, if the time constant (C1R1) is too large.

This does not happen fast enough, as shown on portion B of the waveform of
fig 8.5(e). This effect is most serious at the instant when the rate of
decrease of the modulation envelope is the greatest. This discharge of C1
cannot follow the envelope and is determined by the CR time constant instead.
The higher the modulation frequency and the greater the depth of modulation,
the steeper is the slope of the enveope, and the effect of diagonal clipping is
more severe, as you should have found. Observation of the effect with square
wave signals should have convinced you. It is possible to derive equations
connecting these variable and this will be discussed later.

Practical 8.2 Negative peak clipping

Conntinuing our examination of the detector circuit, keep the circuit of fog 8.4
and the initial circuit of fig 8.2 but:

On the 2950C module

Return your set-up to 100k in parallel with a 1nF capacitor as
the detector load.

On the function generator

Select a sign waveform.
Set to 300Hz.

- 150 -
 Set the output amplitude so that the value of m is about 100%.
(Check that you have the same conditions as when you answered Q8.9)
Now increase the output amplitude so that m is over 100%.
Sketch the display on your oscilloscope.
Turn down the variable d.c so that the carrier component is removed.

Fig 8.5

- 151 -
Q8.15 Does this suggest it is important to have the carrier component
present for correct demodulation?

Now reset the value of m to 100%.

Q8.16 Where is the d.c level on your output waveform? What effects may
this have on subsequent circuits? What determines the magnitude of
this d.c level?

We must obviously try to remove this d.c level as it may upset the bias on
any subsequent stages for example. This can be done by means of C6 and R4
of fig 8.5(a). This then makes fig 8.5(f) into fig 8.5(g). However, this then
introduces extra complications. Until now we have assumed that the detector
is not required to supply any current. R4 in our d.c filter will take current,
not to mention any subsequent circuit, e.g amplifier, which may be connected
across R4.

Such a circuit exist on the detector module. Connect it as shown in fig 8.6
using the a.c amplifier simply to increase the output.

Fig 8.6

Q8.17 Sketch the output waveform you see. Can you explain why this caused?
Note also the effect on triangular and square waves.

The coupling capacitor C6 in fig 8.5(a) has a low reactance to a.c signals, but
blocks the d.c component in the detector output. Thus as seen by the

- 152 -
detector, the d.c resistance is simply R1 this being less than the d.c
resistance. In these circumstances it becomes possible for the a.c component
to exceed the value of the d.c component in the output. The negative peaks
of the a.c will be lost because reverse current cannot flow in the diode D.
This effect is shown in fig 8.5(h) and is called NEGATIVE PEAK CLIPPING (If
you temporarily connect in an additional 5nF capacitor across C, you can
demonstrate negative and diagonal peak clipping together).

Q8.18 Does negative peak clipping depend upon the value of modulation
index m?

To verify your answer, reduce the input signal level and measure the value of
m at which the negative peak clipping disappears.

It is fairly easy to derive an expression for this. If the carrier voltage at the
detector output is, say, E, then the modulation voltage is mE.

The d.c level is proportional to the carrier voltage; the d.c load R1
∴ d.c current = E/R1.

The a.c load is R1 and R4 in parallel

  
∴ a.c currnet = mE. 
  

Since reverse current cannot flow, to eliminate negative peak clipping, d.c
current ≥ a.c current.

   
∴  ≥ mE. 
   


or m ≤ 
  

In our circuit, R1 and R4 are both 100k .

- 153 -
Q8.19 Substitute these values into this equation.
Does the value tie up with the value of m you observed before?

Q8.20 How can we reduce the effect of negative peak clipping, thus
increasing the value of m we can use?

From the last equation you can see that we need to make the a.c and d.c
resistors more nearly equal. This implies a high ratio of R4 to R1. There are
several ways of doing this.

1. Insert a high series resistance in series with C6 to equate the

impedances.
2. Top the output voltage across only a part on R1.
3. Reduce the value of R1.

All these methods lower the output voltage and may damp the tuned circuit.
The detector efficiency is thus low, but additional gain in easily obtained in the
amplifier to compensate for this.

We can demonstrate method 3 by connecting the 4.7k resistor R3 across

the 100k of R1. We will then need to use a capacitance C4 of 20nF so that
the time constant is correct. You should note that the output level is reduced
so turn up the amplifier gain to compensate. There may be still some
evidence of clipping at values of m approaching 100% but the effect should be
much less than before.

Q8.21 Does the distortion introduced by negative peak clipping depend upon
the frequency of the modulating signal p?

Practical 8.3 Automatic Gain Control

Let us consider what happens if the strength of the received signal varies.
This can be caused by a variety of effects and is particularly common in radio
reception. It would be very the convenient if we could automatically vary the
gain of part of the circuit in inverse proportion to the strength of the
received signal. This would provide a constant output power. Such circuits
are called AUTOMATIC GAIN CONTROL (AGC) or AUTOMATIC VOLUME
CONTROL (AVC) circuits.

- 154 -
Q8.22 Which of the circuits you studied in part 1 of this trainer will
allow you to do this? What type of control signal does it require?

Q8.23 From where in our detector circuit can we obtain such a control
signal, proportional to the mean strength of the received signal?

You studied in Assignment 4 how to use the control socket on the amplifier
module to vary the gain by adjustment of the base bias. This required a d.c
control signal.

When an AM carrier has any modulating voltage with equal positive and
negative half cycle, the mean value of the carrier is not altered. If we can
remove the modulation, we can obtain a d.c signal proportional to the mean
carrier level, unaltered by the modulation. This is important as we don't want
the AVC to operate between loud and soft passages of the modulation signal,
but only if the strength of the carrier varies.

From our diode detector circuit we have obtained

1. the wanted modulation evelope

2. a small residual r.f component
3. a d.c component

- 155 -
Fig 8.7

- 156 -
The RC filter removes condition 2 with the time constant correctly chosen to
avoid diagonal peak clipping. We removed condition 3 with a d.c filter, being
carefull to avoid negative peak clipping. We can use an a.f filter to remove
condition 1 leaving the d.c component for subsequent use.

Connect up the circuit as shown in fig 8.7.

Set up the modulator with a carrier frequency of about 465kHz and a sinewave
signal at 300Hz according to the instructions at the start of the Experimental
Procedure.
The modulator output passes to a tuned amplifier, then into the detector.

There may be some evidence of negative peak clipping at values of m

approaching 100% modulation, so adjust the input signal amplitude until this
disappears.

Q8.24 Why have we used 4.7k and 20nF in the detector?

Q8.25 Why have we used RE = 620 in the amplifier?

Q8.26 Does decreasing the input signal level below that just set, vary the
value of the d.c level at point A? What is the value of this d.c voltage?

Q8.27 Does adjustment of the carrier level (by the attenuator contol on
the signal source module) affect the value of this d.c level?

We can now plot the AVC characteristics. It is necessary to process our d.c
signal so that it is in the correct form for feeding into our amplifier. You will
remember from Assignment 5 that our amplifier needs a control signal starting
at about -12 volts and positive-going. Link the two points so marked on the
detector module to introduce a -12V d.c offset into the d.c amplifier. Connect
point (A) (the output of the modulation filter) directly to the input of the d.c
amplifier and set the gain to MINIMUM.

Set the gain of a.c amplifier so that the output is exactly 10 volts pk-pk and
check that there is no clipping or other distortion present. Then take a set of
readings of pk-pk output voltage for 3dB increments of decreasing carrier

- 157 -
level, by using the calibrated attenuator on the signal source module. Record
your readings in a table as in fig 8.8, this set of results is with NIMIMUM
AVC. Now set the gain of the d.c amplifier to MAXIMUM and take another set
of readings. Record thses in the column headed 'Simple AVC'.

While taking reading, keep observing the oscilloscope for any sign of distortion
limiting or clipping and if you notice any, indicate it in your table.
You will notice these outputs are smaller than those with minimum AVC. A
strong carrier signal produces a large d.c voltage, which makes a large
reduction in amplifier gain. A weak one produces only a small control valtage
and allow higher amplifier gain - thus the level of the output is held somewhat
more constant, although smaller.

However, this will still reduce the strength of very weak signals. If the AVC
could be prevented from operating until the signal was above a certain level, a
considerable improvement in constancy of output would be possible.

Connect the circuit labelled '0.5V delay' between the output of the a.f filter
(point A, fig 8.7) and the input of the d.c amplifier. Take another set of
readings of output voltage for different settings of carrier level and enter them
in the third column of table 8.8.

Fig 8.8

- 158 -
Finally remove the AVC feedback link to the control socket of the amplifier
and take another set of readings with NO AVC. You will find that the
amplifier is limiting at high inputs so take care that the readings you have are
correct.

Exercise 8.2

Plot graphs on single-cycle logarithmic graph paper, of output voltage, against

carrier attenuation (dB) for each of the four cases of results in table 8.8. Use
axes as fig 8.9. Should you have no such paper, you can use linear graph
paper if you convert the dB readings to voltage levels, using your calibration
table from Assignment 1. Read from your graph of delayed AVC the 'threshold
level' at which the AVC begins to operate. You should find this is
characterised by a distinct 'knee' in the curve.
For each of three AVC curves, calculate the percentage change in output fot
the change in input level from 0dB to the threshold level you have just read
off.

Q8.28 Which do you consider is the most effective tyoe of AVC?

Q8.29 You notice that with no AVC at all, the amplifier limits.
How have the three forms of AVC offered improvement?

Fig 8.9

- 159 -
Q8.30 How do you think the AVC effect could be improved in a practical
application?

Practical 8.4 Demodulation

Cast your mind back to Assignment 6 where you studied modulation as a

multiplication process. The circuit you used was called a balanded modulator
and with input angular frequencies of a carrier and a signal p you will
remember that the output consisted of two side frequencies  ±p.

Q8.31 What do you think the output frequencies would be if the inputs
were and the lower side frequency ( - p)?

Q8.32 What are the frequencies present in a full AM wave?

What would be the output frequencies of the balanced modulator if
the inputs were a full AM wave (of signal p on a carrier  ) and also
the carrier itself?

Q8.33 Do you think that modulator could therefore be used for demodulation?

Q8.34 How could we remove the unwanted frequencies?

Let us see if you are correct. We must first generete a full AM wave which
we can conveniently do by using the non-linear amplifier as we did in
Assignment 7. We can then feed this and the carrier frequency into our
balanced modulator and look at the output.
Connect up the layout of fig 8.10.(see overleaf)

On the 2950A module

Set the manual frequency conrol to the dot 465kHz.
Set the variable attenuator to 0dB.

On the 2950B module

Select the emitter resistor RE of 10.6k .

On the function generator

Set to zero output.

- 160 -
On the oscilloscope
Set the Y1-amplifier to 0.2V/cm and Y2-amplifier to 0.02V/cm.
Set the timebase to 1ms/cm.

Slowly increase the output from your function generator until a full AM wave
with a modulation index of about 70% is displayed. You should find this
happens at about 0.7 volt pk-pk. Observe the Y2 trace, i.e the output from
the filter.

Try other waveforms and frequencies and see if this method is satisfactory.
You can also try the XY trapezoidal displays as mentioned in Assignment 6.

Practical considerations
& applications

When you studied the properties of the amplitude modulated wave in

Assignment 6 you saw that it contained no components at the signal frequency
p but only components at carrier  and side frequencies,  ± p. Thus in
order to recover the information, some form of DETECTOR must be used.
You saw that, in the full AM wave (carrier + both side frequencies) the
envelope of the wave resembled the original signal so that easiest form of
demodulation is to use what is called an envelope detector.

This process is similar to that of rectification. You first used a diode to

remove half the waveform and make it asymmetrical, then filtered off the
remaining carrier with capacitors. A Germanium diode is used because of its
lower forward voltage drop. The diode is necessary otherwise the symmetry
of the waveform cancels out the signal information.

- 161 -
Fig 8.10

- 162 -
The output of the diode then consists of the following components:

a) an a.f signal corresponding to the envelope.

b) an average d.c level.
c) a residual (distorted) carrier frequency.

Of these, it is component (a) which is required and you saw how component
(c) could removed by the addition of a capacitor. This capacitor charged up as
the envelope increase and discharged as it decreased. It is not able to follow
the rapid variations at carrier frequency so the output closely resembles the
original signal which in contained in the envelope.

You saw that it is the time constant of the combination of this capacitor and
the diode load resistor that is important. If is too small also the output tries
to follow the carrier frequency. Ripple is present and the output voltage is
not as large as it could be. If too large, C does not lose its charge as quickly
as the falling slope of the envelope; you saw that this produced diagonal peak
clipping.

The input to the detector is derived from a tuned circuit so the detector input
impedance should be as high as possible in order to avoid damping of the
tuned circuit. As the forward resistance of a germanium diode is low it is
used to neglect it.

It can then be shown that the input impedance approximate to R. With no

capacitor it appears as 2R. Thus the higher the value of R, the more efficiency
you calculated. However there may be other constrains on the values of R
and C, for example, the value of R must be appropriate to the input of the
next stage. High detector efficiency is not absolutely essential as the signal
can easily be amplified later. This may introduce other problems which we
will discuss in a moment. In television receiver, R is fairly small. The
resultant high damping is usually acceptable because of the wide bandwith
required.

Returning to the problem of diagonal peak clipping, you should have found that
it depended upon the signal frequency p and the depth of modulation m.
Mathematics show how to predict the required time constant knowing m and

- 163 -
the maximum signal frequency p. The formula is


  
 = RC = 
∙

If you substitute your measured values into this formula, you should find good
agreement between theory and pratice. Although this formula shows that for
100% modulation (m = 1) 0 practice shows that the distortion is not excessive
if  < 1/p. The effect of diagonal peak clipping is worst at

the instant when the rate of change of the modulation envelope is greatest, i.e
at high values of m and p. Fortunately in sound broadcasting these seldom
occur together.

The d.c component present in the detector output is used for AVC as will be
discussed later but it must be separated from the signal otherwise it would
upset the bias level on subsequent signal amplifier stages. This is done with a
d.c filter but this means that the detector is now required to supply current
and, if the d.c and a.c resistances are unequal, negative peak clipping may
occur. This was discussed fully in the body of the assignment but one or two
important conclusions emerge.

This distortion is introduced at high values of m, but you should see that it
does not depend upon the frequency of the modulating signal p. We derived
an equation linking the maximum permissible value of m with the component
values if negative peak clipping is to be avoided, and demonstrated a method
of achieving this.

The resultant lowering of detector efficiency is not too important, since as

previously discussed the signal can be further amplified. In these days of
transistors and integrated circuits and additional amplifier stage is very much
easier to use than it was with thermionic valves. It is also possible to use a
separate diode and apply a bias but this is seldom used as it causes other
problems.

Practical 8.3 was a qualitative practical, designed to demonstrate that by a

suitable choice of input frequencies, modulators may also be demodulators.
These are quite simply two different applications of the principle of frequency

- 164 -
changing, or heterodyning, which was discussed earlier. Here we used our
balanced modulator module, but most types of modulator also lend themselves
to be used as demodulators, including the ring-bridge and non-linear modulator
as studied in Assignment 7. However, mathematics of the non-linear
demodulator shows that second harmonic of the signal frequency is introduced,
proportional to m2(i.e t component 2pm2). Such detectors were satisfactory in
the early days when low values of m were used but with the high values of
m in use today. The distortion is undesirable and non-linear demonstrators
are seldom used.

This method is called synchronous demodulation since the demodulating signal

must be synchronous with the signal is not necessary for this sort
demodulation. In practice the carrier in the transmitted signal is often reduced
in amplitude before transmission, to save on transmitter power (suppresses
carrier transmission). At the receiver the carrier is selected and amplified
separately from the signal and is used in the synchronous demodulator to
demodulate the complete incoming signal.

For speech exact synchronism between carrier and demodulating signal does
not render the speech unintelligible, however where very low frequency signals
or even d.c signals are present in the modulating waveform as telemetering
systems, then exact synchronism and phase relationship are necessary.

With simple diode detection the full carrier must be present, and you saw the
effects of over-modulation and loss of carrier on the output waveform. Thus
the choice of transmission system and detector must be made together. Diode
detection requires a full AM wave and the carrier must be trnsmitted with the
consequent power wastage at the transmitter. This system is economical if a
single transmitter serves many, inexpensive receivers, for example in
medium-wave broadcasting. On the other hand, if there is only a single
transmitter and receiver, it may be important to conserve power at the
transmitter (especially if battery operated) at the expense of a more complex
receiver. Then one of the systems of synchronous demodulation is used.
Examples occur in private and military radio systems and in line telephony. In
the latter, with multi-channel operation, a single carriers are regenerated at
the receiving end. In this way they can all be synchronized to a single master
reference standard.

- 165 -
The other practical in the assignment covered automatic gain (or volume)
control -AGC or AVC. This could be used with any demodulation system but it
is a simple matter with diode detection since the d.c component in the detector
output gives a direct measure of the strength of the received signal. This can
then be processed into a form suitable for controlling the gain of an amplifier
stage ahead of the detector. The effect can be improved by increasing the
number of stages under control; two or three is usual.

The operation of this system has already been described, together with the
reason for trying delayed AVC. You should be able to see from your graph a
distinct 'knee' in the curve where the delayed AVC begins to operate. You
should also notice that the slope of the curves determine the improvement
provided by AVC, delayed AVC being the most effective. Your graph with no
AVC at all should clearly show limiting and the improvement that AVC provides
- notably the ability to provide a reasonable output even from very large
inputs.

The choice of component values in the a.f filter is important. The resultant d.c
for AVC should be quite smooth, otherwise the ripple may cause the AVC to
operate within one cycle of modulation. Addition of an extra capacitor to
reduce the ripple should be carried out with care, as this may slow down the
AVC.

- 166 -
LAB.SHEET 9
- SUPERHET RADIO

Object
To demonstrate an application of the principles of modulation and detection as
already investigated - reception of broadcast radio using the superhet principle.

Equipment required
● Power Console ED-2900P
+15 volts d.c
-15 volts d.c

● Oscilloscope, double beam, 5MHZ calibrated

● Digital Frequency Counter, 1MHZ

● Modules Type 2950A, B, C and D

● Tuned Circuit Assembly Type 2950H
● Aerial Assembly Type 2950K

Approximate Time required

Two hours

Prerequisite Assignments
1, 2, 4, 6 and 8

Theory Assumed
Heterodyning(Frequency Changing)

- 167 -
Preliminary Procedure

1. Attach the console ED-2900P.

2. Connect the power supply unit(s) to the mains supply line.

DO NOT SWITCH ON YET.

3. Turn the variable d.c control knob to minimum.

4. If you are using any of modules 2950A, B, C, D or E, plug them into the
sockets on the Console in alphabetical order. If any change is requires,
it will be indicates in the assignment. The act of plugging in the
modules connects them to the power supply in the correct manner. All
sockets are identical.

5. If you are using either of the modules 2950F or G, or the Assemblies

2950H, J, K or L, these do not require any power and may be stood
where convenient or as directed in the assignment.

6. This completes the preliminary procedure.

Introduction

In previous assignments you have studied the functions of all the various
modules. We will now show you how to combine them as they would be in a
typical application, a broadcast receiver. Because you are familiar with the
functions of the various blocks this will be presented as a series of Laboratory
Notes and you will be expected to make some decisions for yourself.

From Assignment 8

We want to recall two very important points from Assignment 8. Firstly, in

order to study diode detection you first generated an AM wave by modulating
a 465kHz carrier wave , with a signal p from your funtion generator. We
suggested you could also try a signal from a tape recoder. The output,
composed of carrier and sidebands, passed directly to the diode detector which
recovered the information. We used a carrier frequency of 465kHz because our

- 168 -
tuned circuits have been designed to operate at this frequency - you will soon
learn why.

Consider for a moment what would happen if we wanted to receive a radio

transmission, for example in the medium wave broadcast band, in which carrier
frequencies between about 500kHz and 1650kHz are used. The signal level
picked up by our aerial may only be a few micro-volts or even less and this
must be amplified to a few volts for the detector to operate properly. The
tuned circuit must be sufficiently selective to pick out the wanted signal
frequency from the thousands of transmitted frequencies prevent in the
crowded air space. These requirements suggest several cascaded tuned
amplifiers to achive sufficient selectivity and gain, all simultaneously variable
between about 550kHz and 1650kHz. Ganged tuning capacitors have to be
used. Amplification becomes more difficult at higher frequencies. This
assignment is called a tuned radio frequency or TRF receiver. We cannot
demonstrate one of these as our tuned circuits cannot cover the frequency
range. However, the main disadvantage is the possibility of unwanted
oscillation. With several common frequency amplifier stages it only needs a tiny
signal to be fed back by radiation or capacity coupling, from bad wiring or
layout or the like and, as you know, if this signal is in such phase as to
sustain the output, the circuit will oscillate.

There are thus several advantages to be derived from carrying out much of th
RF (radio frequency) amplification at a fixed frequency. The frequency can be
chosen so that amplification is easy to achieve the tuned circuit can be set
once and for all during manufacture;ganging and other tuning difficulties
disappear; screening is less critical. This last point, together with the division
of the amplification between two frequencies, which means that less gain is
needed at each frequency, much reduces the chance of oscillation and make for
better stability.

This brings us to the second point we want to recall. You have learnt that
by suitable choice of input frequencies a modulator circuit can also be used as
a demodulator and a frequency changer. Thus we have a means of fulfilling the
requirement above;translation of the RF to another frequency. This is called
the INTERMEDIATE FREQUENCY or IF as it is usually intermediate between
the carrier and the modulation signal p. Another name for a frequency

- 169 -
changing circuit is heterodyne. The complete receiver is called a
superheterodyne, or simply a superhet.

Fig 9.1

A block diagram of the system is shown in fig 9.1. The aerial collects the
wanted carrier and all the unwanted signals and passed them to the RF stage.
This is tuned to the required carrier frequency. The output of this stage
passes to the frequency changer, or mixer, where it is mixed with a locally
generated signal from the local oscillator(LO).

If the chosen intermediate frequency (IF) is say f kHz, then you should be able
to see that the required local oscillator frequency y =  ±f. When heterodyned
with carrier  , either of these frequencies will produce the required IF
frequency. Any sidebands present on the carrier(f ± p) will be similarly
heterodyned to (f ± p) and the sidebands are then amplified in the fixed
frequency IF amplifier before being passed to a simple diode detector which
recovers the signal, p. This then passes to an audio amplifier with manual gain
control, thence to the speaker. You will notice that we are using two further
assemblies in the kit, the Ferrite Aerial assembly 2950K and Audio Unit2950L.

Ferrite Aerial

A more detaied mimic of the Ferrite Aerial module2950K is shown in fig 9.2.

- 170 -
The ferrite aerial is the RF stage and is quite simply a tuned circuit, tuneable
between about 600Hz to 1600Hz. The ferrite rod has a high permeability and
efficiency at radio frequencies so that the coil wound on it can act as an
aerial. Unless you are situated close to a powerful transmitter the signal
received may not be sufficiently powerful and so a few metres of wire
attached to the hi terminal will greatly improve results.

Fig 9.2

Speaker module

The speaker module may be replaced by a power amplifier and loudspeaker to

increase the sound level.

- 171 -
Fig 9.3

- 172 -
Practical 9.1 To set up the radio

You should now be able to construct the radio yourself from the arrangement
of fig 9.1. However in case of difficulty use the circuit given in fig 9.4. Keep
all the leads as short as possible.

On the mixer (2590D module)

( The module must be balanced correctly
or low gain and distortion will occur)
1. Connect a high impedance 10V d.c
voltmeter across the terminal c and
as in fig 9.5.
2. Earth input b (not through the capa-
citor). Connect the +1V socket to
input a. Adjust balance a control Fig 9.4
until connection or disconnection of
the input cause no change of output.
3. Earth input a.
Connect the +1V socket to input b (not through the capacitor).
Adjust balance b for zero output change as in 2.
4. Repeat above, as the two controls are interdependent.

- 173 -
On the detector stage (2950C module)
1. Use a detector load of 100k and 1nF.

On the IF amplifier (2950B and 2950H modules)

1. Use an emitter resistor RE = 620 decoupled by CE = 25F(2950B
module).
2. Set the scale of the tuned circuit (2950H module)to 465kHz.

On the local oscillator (2950A module)

1. Set the variable attenuator to 0dB.
2. Set the manual frequency control C to the local oscillator frequency
(ykHz) that you have chosen.

With the frequency range of the signal source module available we have to use
the lower LO frequency.

Set the output control to 0dB. Then tune the aerial module tuned circuit until
the station is heard (using the frequency scale as a guide). Finally trim the IF
amplifier, tuning for maximum volume. The volume control is the audio
amplifier control on the detector module.

When you are familiar with the setting-up procedure, try finding other stations
using the method given above. After a little practice, you sould be able to do
it without the frequency meter. Remember that tuning requires the operation of
at least two controls and this is not an easy task.

Notes
1. The 200kHz carrier frequency of BBC Radio 2 (1500m long wave) is well
below the tuning range of our aerial module (RF stage). However, if you
disconnect this and just leave a few metres of wire connected to the input
of the mixer you may pick up enough signal to be able to receive this
station by setting the local oscillator to 665kHz.

2. Tune into a strong station, then deturn the local oscillator very slightly.
This will shift the IF. Note that the station can be brought back into tune
by adjusting the IF in the opposite direction to that in which you detuned
the LO.

- 174 -
3. Using the frequency scale on the aerial module as a guide, adjust the
tuning of the RF stage and note the bandwidth (in terms of kHz) over
which the station can be satisfactorily received. Adjust the tuning on the
IF amplifier and note that the satisfactory bandwidth (in terms of kHz) is
narrower than that of the RF stage.

4. Change the value of detector time constant. Note the reduction in gain with
R too small and the distortion with C two large, especially with large
amplitude high frequency signals. Compute this with what you learnt in
Assignment 8.

5. The output level control of the local oscillator controls the output of the
mixer and acts as a gain control in this case.

6. A trapezoidal display, as shown in Assignment 6, can be achived. This

should show a good linearity and give an idea of the values of modulation
index used.

7. AVC can be demonstrated as indicated in fig 9.4 but, with the modulate
gain available with the 2950, it requires an exceptionally strong signal for
it to be necessary and for if to operate significantly.

Practical consideration
& applications

You have already demonstrated one of the major applications of the principles
of modulation and detection - the reception of broadcast radio transmission.
You have seen how to use a balanced modulator as a frequency changer and
demonstrated the superhet principle. This was first invented during World War
1 but not widely adopted until after 1927. It is now the accepted standard for
radio and television receiver. Similar systems are used inline telephony;the
principles of frequency grouping were discussed in Assignment 6. We have
already explained that ease of design for selectivity and high gain are the main
reasons for this preference. We shall now consider some of the practicalities
and other advantages and disadvantages of this circuit.

- 175 -
The local oscillator frequency may be above or below the carrier frequency by
the correct amount - the IF. In a broadcast band receiver, the RF circuits must
tune from about 550kHz to 1650kHz, a frequency ratio of 3:1. With an IF of
465kHz, if the LO were below the carrier, it would cover a range of 85kHz to
1185kHz, a ratio of almost 14:1. If above the carrier, the requires range is
1015kHz to 2115kHz, a ratio of just over 2:1. This is much easier to achieve
with the usual tuning capacitors than the 14:1 ratio. However we have used
the lower range in this trainer so that the same module can be used for
different functions. Since the frequency is voltage controlled and not capacitor
tuned, this is no disadvantage as it would be in a practical receiver.

You noticed how difficult it was to tune the receiver, requiring the operation of
at least two controls. To achieve the advantages of 'single knob tuning' it is
necessary for the LO tuning capacitor to be ganged with those of the RF
stages. The RF tuned circuits, if more than one is used, are made as nearly
identical as possible. Obviously the LO circuit cannot be the same and thus
cannot use the same value tuning capacitor. It is usual to include pre-set
capacitors in the LO tuned circuit, (as shown in fig 9.6). It can then be
arranged that the LO and RF tunings are exactly correct at three points in the
tuning range (usually 600kHz, 1000kHz, 1600kHz) and not more than a few
tenths of one percent out at the other points. This is called tracking. Variable
capacitors are available with the plates of the oscillator section, specially
shaped to provide the correct frequency variation to angle of rotation.

Ganged radio frequency and load oscillator circuits

Fig 9.6

One of the difficulties of the superhet is the presence of image frequencies.

We can understand this probably better if we assume that the L.O(y) is higher
than the RF( ) as it would be in a broadcast receiver. Then just as mixes

- 176 -
with y to provide an output of the intermediate frequency, f = y - , then so
does an unwanted signal at frequency u such that u - y = f. This particular
frequency is much above the local oscillator frequency as the wanted signal
is below it and is therefore called the image channel or second channel. For
example, with BBC Radio 3 at 647kHz an LO of 182kHz procedures the
required IF of 465kHz. But an unwanted signal at (1112 + 465) = 1577kHz will
also be heterodyned by the mixer to an IF of 465kHz. It is the function of the
RF stage to be sufficiently selective to eliminate the image channel. This is
shown diagramatically in fig 9.7 and from the response curves of the RF tuning
you can see that image signals at 1577kHz are likely to be considerably
attenuated. This problem is less severe if the LO is below the RF.
Where the specification is not of prime important, RF amplification is often
omitted. The RF stage then simply consists, as here, of a single tuned circuit
at the input to the mixer. This must be sufficiently selective to reject the
image channel. Often this is not too difficult as the wanted signal is near and
therefore received at good strength. At higher fre- quencies difference
between the wanted and unwanted signals become small compared with the
bandwidth of the tuned circuit. The provision of more RF tuned circuits is
essential for good results. This requires one more RF amplifiers. This
requirement is also common in what are called com- munications receivers.
These are essentially high specification mutiple band radio receivers.

The requirements of the frequency changer have already been covered. It

usually takes the form of a multiplicative circuit which, you remember, was

also used for modulation and demodulation. This circuit was sometimes called
the first detector of the superhet. Sometimes the mixer and local oscillator are
combined. All frequencies developed in the mixer are discarded except the
difference frequency which is passed to the IF stage.

- 177 -
 Fig 9.7

The superhet principle enables the IF stages to be suitably designed for good
amplification at the fixed frequency. The tuning can be set during manufacture.
Coupled tuned circuits are frequently employed to give the required selectivity
curve, commonly as rectangular as possible with steep sides and a flat top.
This helps in rejecting interference which is on adjacent frequencies. This is
not always removed by the RF stage, since its absolute bandwidth (in term of
Hz) is wider than that of the IF stage. Communications receivers are
sometimes fitted with variable bandwith controls to enable the best compromise
between interference and intel- ligibility to be obtained. For further
improvement is selectivity the process of frequency translation may be
repeated to provide further gain and more tuned circuits of a still lower IF.
Double and triple conversion systems are available.

The choice of intermediate frequency is one of compromise. A low IF has the

advantages of better selectivity (the absolute bandwidth is lower) and greater
amplification and better stability for a given circuit. If it is too low, there is a
greater likelihood of the image channel reaching the If and of unwanted signals
being heterodyned into the audio spectrum and causing whistles. Also the IF
the closer the local oscillator is to the signal

frequency and the greater the possibility of frequency pulling of the LO by the
signal. Clearly the IF should not be one used by a station which might be

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picked up directly by the receiver. Broadcast receivers use an IF of about
465kHz and such a frequency is suitable for signals up to about 7MHz. Up to
about 30MHz, 1.6MHz is more satisfactory IF. Television receivers with their
wide bandwidth need a much higher IF and this is usually about 30MHz.

The remainder of the circuit is quite straight forward. The IF amplifier

provides enough output for distortion-free operation of the detector. The
simple diode detector is quite adequate for full AM waves, as discussed in
Assignment 8. The time constants are chosen for best results as discussed
previously. Then follows a signal amplifier to provide the requires audio (op
video) output. The gain control of this amplifier may be the only one in the
circuit or controls may also be provided on the RF and IF stages. This enables
the operator to adjust each section of the receiver for optimum performance in
accordance with the existing conditions of signal strength and interference.
Automatic gain control is usually also provided. This provides some protection
against amplitude variations but cannot cope with transient interference, to
which AM systems are particularly susceptible.

If the receiver is to be used for the reception of interrupted continuous wave

transmissions (e. g morse code) then another oscillator called a beat frequency
oscillator is necessary. This oscillates at frequency about 1kHz removed from
the IF and its output is mixed with the IF before the detector stage. The result
is a beat frequency of about 1kHz which thus renders the CW audible in the
output.

- 179 -
ASSIGNMENT - 10
- FREQUENCY MODULATION

Object
To study the characteristics of Frequency Modulated Waves.

Equipment required
● Power Console ED-2900P
0-4 volts d.c, variable, metered.
+15 volts d.c
-15 volts d.c

● Oscilloscope, double beam, 5MHz, calibrated

(long persistence or storage)
● Digital Frequency Counter, 1MHz
● Signal Generator

● Modules Type 2950A

● Tuned Circuit Assembly, type 2950H
● Crystal Filter Assembly, type 2950J

Approximate time required

Two hours

Prerequisite Assignments
1, 2, 3 and 6

- 181 -
preliminary Procedure

1. Attach the console ED-2900P.

2. Connect the power supply unit(s) to the mains supply line.

DO NOT SWITCH ON YET.

3. Turn the variable d.c control knob to minimum.

4. If you are using any of modules 2950A, B, C, D or E plug them into the
sockets on the Console in alphabetical order. If any change is re-
quired, it will be indicated in the assignment. The act of plugging in
the module connects them to the power supply in the correct manner.
All sockets are identical.

5. If you are using either of the modules 2950F or G, or the Assemblies

2950H, J or K, these do not require any power and may be stood where
convenient or as directed in the assignment.

6. This completes the preliminary procedure.

Experimental Procedure

We said in the introduction to this manual that an unmodulated wave conveys

no information and that it was necessary to vary one or more properties of
this wave in order for communication to take place. This amplitude modulation
was covered at length in Assignment 6-9 and we now want to consider what
happens if we vary the frequency instead.

Remember when you studied the signal source module in Assinment 1 you
discovered that the frequency could be changed by the application of a d.c
voltage to the control inputs.

Q10.1 What do you think would happen if you fed in an a.c signal to
the control input of the signal source module?

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Practical 10.1 Introduction to Frequency Modulation.

Let us see what happens when you feed an a.c signal to the control input of
the signal source modulation.

Connect up the layout of fig 10.1.

On the 2950A module

Set the manual frequency control C to 465kHz.
Set the variable attenuator to 0dB.
Set the sensitivity control B to maximum (X0.1)
On the oscilloscope
Set the Y1 amplifier to 0.5 volt/division.
Set the Y2 amplifier to 2V volts/division.
Set the timebase to 0.5 second/division.

On the function generator

Select a square-wavw output.
Set the frequency at 0.1 Hz.
Set the signal amplitude to 0v.

On the frequency counter

Set to sample once per second.
Using the frequency counter, set the signal source output to 465kHz
exactly.
Set the function generator output to 10 volts pk-pk.

Q10.2 What is happening to the frequency as shown on your oscilloscope?

What
are the readings in the digital frequency counter?

You should see that the frequency if alternating equally about 465kHz. This is
called the CENTRE FREQUENCY and the amount by which the frequency
changes is called the FREQUENCY DEVIATION.

- 183 -
 Fig 10.1

Exercise 10.1a

What is the frequency deviation (either side of 465kHz) with a signal of 10V
peak-to-peak on control input B, at maximum setting? Read off from your
frequency calibration graph from Assignment 1 the frequencies you would
expect with 0.5V control voltage and compare them with those measured
(remember the x0.1 attenuation setting on the B input makes 10V pk-pk
equivalent to 0.5V control).

Reduce the amplitude of the input square wave to 5V peak to peak, as

measures on the oscilloscope.

Q10.3 What happens to the output signal? What does the frequency counter
read now?

Q10.4 What determines the frequency deviation of the output?

Still using the calibrated oscilloscope graticule. reset the function generator
output to exactly 10V peak-to-peak and still with the 0.1Hz square wave,
adjust the B input sensitivity so that the deviation either side of the centre
frequency of 465kHz is exactly 100kHz (i.e from 365kHz to 565kHz).

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DO NOT ADJUST B CONTROL AGAIN DURING THIS ASSIGNMENT

Now you can set known frequency deviations directly by setting the input
voltage.

i.e 10V pk-pk input Frequency Deviation of 100kHz

5V pk-pk input Frequency Deviation of 50kHz
2V pk-pk input Frequency Deviation of 20kHz

You may find it helpful to use the oscilloscope to make calibration marks on
the output amplitude control of your function generator. (It is easier to do this
calibration at 100kHz). Alternatively use a d.c voltmeter to ead the peaks of
the 0.1Hz square wave.

Reset the input to 10V pk-pk and slowly increase its frequency to 100Hz.

Q10.5 What happens to the displayed signal? Why is the frequency counter
reading meaningless?

Q10.6 What determines the rate of change of the frequency of the output?

Reset the input frequency to 0.1Hz and select a sine wave.

Q10.7 What do you see now on the oscilloscope? What happens if you
increase the input frequency in decades to 100Hz?

Q10.8 Has the amplitude of the output signal changed at all during any
these tests?

Q10.9 How will the power supplied by an unmodulated wave change if it is

frequency modulated?

Deviation Ratio

Let us express these results mathematically. We will use the usual symbols;

 = centre frequency (angular) radians/sec

- 185 -
 p = signal frequency (angular) radians/sec

and we will use D to denote the absolute peak frequency deviation from (you
have found that this depends upon the amplitude of the input signal).

The frequency at any instant is  + D cos pt. Consider a sinusoidal signal of

peak amplitude A and frequency s (we must consider only the fundamental
component of our square wave). The signal can be written as A sin st.

In the equation, st is the total electrical angle converted by the signal up to

tome t. It is directly proportional to s since s is constant. However when FM
is being used the frequency is not constant and we must find the angle bi
integration.

s = (  + D cos pt) . dt

D
∴ s =  t + _ . sin pt
p
(ignoring the constant of integration as a constant angle).

If you no not understand integration, you will have to take this result for
granted.

It means that the frequency modulated wave can now be written as

D
A sin (  t + _ . sin pt)
p

D
The quantity _ is called the DEVIATION RATIO and since D and p are in the
p

same units (radians/sec or kHz) will be simply a number.

Thus the FM signal is A sin(  t + sin pt)

- 186 -
In practice, D is usually fixed in relation to the peak input signal amplitude.

Q10.10 If the peak frequency deviation D is fixed, what happens to the

value of deviation ratio if the frequency of the modulating
signal p increases? Remember this, it is important.
Q10.11 What is the deviation ratio for a transmission which carries a
signal of 15kHz with a frequency deviation of 75kHz?

Another coefficient, the modulation index, is sometimes used, and defined as


m = 


In the above example D = 75kHz and  = 465kHz, so that m = 75/465 = 0.16.

It is of limited use and we shall not use it in this manual.

Effect of a continuously changing signal frequency

At first sight it may seem that as the frequency swings D the bandwidth
required will be 2D(red/sec). This is not the case, since the continuously
changing frequency contains harmonics well outside this bandwidth. Let us
prove this mathematically, then we will demonstrate this to you.

The equation of the FM wave, A sin(  t +  sin pt) is of the form sin (a + b)
and can be expanded by the double angle formulae to

A sin t.cos(  cos pt) + A cos  t.sin(  sin pt)

These complex functions can be expanded using mathematics well beyond the
scope of this book, to give
A { J0() sin t
+ J1() [sin(  + p)t - sin(  - p)t]
+ J2() [sin(  + 2p)t - sin(  - 2p)t]
+ J3() [sin(  + 3p)t - sin(  - 3p)t]
+ J4() [sin(  + 4p)t - sin(  - 4p)t]
+..........}

- 187 -
Where J0(), J1(), J2(), etc are constants whose value depends only upon 
They are called BESSEL FUNCTIONS, and tables are available which give
these values. Examples are given in fig 10.5 of this assignment.

Thus an FM wave is composed of a centre frequency ( /2) and

(theoretically) an infinite number of side frequencies  ± p,  ± 2p,  ±3p etc,
each pair spaced by an amount equal to the modulating signal frequency (p/2
). The amplitudes of all these are determined by the Bessel Coefficients. If
the modulating signal is a band of frequencies, there will be an infinite set of
sidebands.

We are going to show you these side frequencies and you will measure the
values of the Bessel coefficients. Do not be alarmed, however. In practice,
especially for small values of  the Jn() coefficients decrease rather rapidly.
The series converges and the number of side frequencies required for good
transmission becomes finite. For example, when  = 2.

J0(2) = 0.224
J1(2) = 0.577
J2(2) = 0.353
J3(2) = 0.129
J4(2) = 0.034
J5(2) = 0.007

This implies that if the amplitude of the unmodulated wave  is A, then

frequency modulation with a value of  = 2 means that there will be a centre
frequency  of amplitude 0.224A and a pair of side frequencies  ± p of
amplitude 0.557. With AM there would be no more frequencies, but with FM
there will be a second pair of side frequencies  ± 2p of amplitude 0.353A, a
third pair  ± 3p of 0.129A, a fourth pair  ± 4p of 0.34A and so on until the
amplitude becomes negligible. Let us now demonstrate this.

Practical 10.2 Observation of the side frequencies

Let us now see these side frequencies shown on our oscilloscope display. We
will use the same method that was used with the AM signal when we swept

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the signal source to pick out the frequency, components with the crystal
module. In Application Note 3.

Connect up the layout of fig 10.2

Fig 10.2

On the 2950A module

DO NOT disturb the setting of control B.
Leave the manual frequency control C at ≈ 465kHz.
Set the variable attenuator to 0dB.

On the oscilloscope
Set the Y1 amplifier to 0. 5 volt/division.
Set the timebase to 500ms/division.

On the function generator.

Set to a sinewave output.
Set the amplitude control to zero.
Set the frequency to 20kHz.

Adjust the A and C controls on the signal source module so that, with no
modulating signal on input B, you see the 'blip' corresponding to 465khz in the
centre of the screen.

- 189 -
Tune the tuned circuit for maximum output and adjust the trimming capacitor
on the crystal module for a symmetrical display.

Now set the function generator to give a signal input of sinewave at 20kHz.
Slowly increase the amplitude to 2 volts peak-to-peak and you should see the
sets of side frequencies displayed. In order to provide an approximate cali-
bration, adjust the A sensitivity control so that the distance between each 'blip'
is one major division on the oscilloscope graticule. Then (temp- orarily
disconnecting the signal if necessary) trim the tuning control C so that the
centre frequency 'blip' coincides with the centre of the screen. This is a
sensitive operation and you may find it helpful to adjust your X-gain and
X-shift controls on the oscilloscope in addition. As discussed in Assignment 2
the control settings when using this swept display technique will depend upon
the nature of the timebase output from each individual oscilloscope. You may
need to externally attenuate your timebase output down to a suitable level;
alternatively use another function generator for the X axis and sweep. If you
do not adjust these controls again, the X axis will now be calibrated so that 1
division = 20kHz. This means that on a 10- division screen, the frequency is
being swept from 365 to 565kHz.

Q10.12 With 2V pk-pk signal input, what value of frequency deviation D have
you set ? (refer back to the text between questions 10.4 and 10.5
for your answer).

Q10.13 With a signal frequency of 20kHz, what is the value of deviation

ratio?

Q10.14 Sketch what you see on the oscilloscope.

Using your approximate frequency calibration, read off the appro-
ximate bandwidth required to transmit all the significant pairs of side
frequencies. Ignore those of a level comparable to the noise. How
many pairs is this?

Now set Vp to 4 volts pk-pk, leaving p at 20kHz.

Q10.15 How can you increase the value of ?

- 190 -
Q10.16 What are now the values of D and ?

Q10.17 How many pairs of side frequencies are now significant and what is
the bandwidth required to transmit these? Sketch what you see.

Now set Vp to 6 volts pk-pk, leaving p at 20kHz.

Q10.18 What are now the values of D and ?

Q10.19 How many pairs of side frequencies are now significant and what is
the bandwidth required to transmit these?
What bandwidth would have been required if AM had been used?

Keeping Vp = 6 volts, reduce the signal frequency p to 4kHz.

Q10.20 What is now the value of ? What has happend to the trace?

Q10.21 From your approximate calibration, estimate the bandwidth required

to transmit all significant side frequencies. What bandwidth could
have been required if AM had been used?

Repeat the above for Vp = 4 volts and 2 volts pk-pk. Try to sketch these
traces. Summarize your results in a table as in fig 10.3.

Fig 10.3

Without changing the value of D, vary the signal frequency p slowly over the
decade 1 kHz-10kHz. Try this for other values of D.

- 191 -
Q10.22 What happens to the number of side frequencies?
What happens to the bandwidth?

Exercise 10.1b

From the above and your table of results (fig 10.2) what can you say about
the bandwidth requirements as you vary D and p and hence .

You should have noticed that the value of the Bassel coefficients Jn() changes
at every value of because = D/p and D is fixed by the peak signal
amplitude.The low signal frequencies have high values of and hence more
significant side frequencies.

The high signal frequencies have low values of and hence fewer signifivant
side frequencies.

You should have seen that a greater number of significant, but more closely
spaced, side frequencies (low value of p) will occupy about the same bandwidth
as does a fewer number of significant but wider spaced side frequencies (high
value of p). For a given of D, the bandwidth is roughly constant.

Practical 10.3 Measurement of the Bessel Coefficient

It is now time to measure the values of some od the coefficients

With the circuit of fig 10.2 still set up, and without adjusting control B,
disconnect the oscilloscope timebase from input A on the signal source module
and connect instead the 'var d.c' socket to input A. (This will en- able you to
vary the frequency by use of a d.c control voltage). Use your digital frequency
counter to set the frequency of your function generator to exactly 10kHz then
connect the frequency meter to the -6dB output of the signal source module.

With the variable d.c at minimum, turn the signal input to zero and set the
centre frequency to 466.0kHz by use of control C.

Now turn the variable d.c up to its maximum on the 4-volt range and set the

- 192 -
A sensitivity control so that the frequency counter reads about 410kHz.
(Now adjustment of Var d.c will vary the centre frequency freom about 466 to
410kHz and the crystal will pick out each side frequency as it coincides with
the 465.17kHz resonant frequency of the crystal). Disconnect the frequency
counter.

If as = D/p and we have set p tp 10kHz, by varying the signal voltage Vp

between 0 and 10 volts peak-to-peak, we can vary the frequency deviation
between 0 and 100kHz, hence the value of between 0 and 10.

Set the oscilloscope to 1 volt/division and turn var d.c to near zero so that the
centre frequency of 465kHz is displayed. Make sure Vp is set to zero so that
is zero. Trim the tuned circuit for maximum amplitude, then trim var d.c for
maximum amplitude and read the peak-to-peak output voltage from the
oscilloscope. This coresponds to the coefficient J0(c).

Because of the high Q of the crystal circuit you will find it necessary to vary
var d.c very slowly about the maximum a few times so that you are rea- lly
sure you are getting the maximum peak-to-peak output. You may also find it
helpful to turn off the oscilloscope timebase so that only a straight line is
displayed. If a storage oscilloscope is available, you can slowly swing the
frequency a few times about resonance, and the maximum peak-to- peak
voltage will be retained on the screen for easy measurement.
As you have seen for = 0 there are no outputs other then the centre
frequency, that is all other Jn(0) values are zero.

Now set Vp to 1 volt pk-pk, using the calibrated oscilloscope or your own
calibration of the amplitude control, correspnding to = 1. Adjust var d.c for
amximum output at 465kHz and read the peak-to-peak signal freom the
oscilloscope. This corresponds to J0(1).

Now increase var d.c until the first side frequency is picked out and trim for
maximum amplitude. The frequency is 465 - 10 = 455kHz. The peak-to-peak
output from the oscilloscope corresponds to J1(1).

Repeat the above at 10kHz intervals by adjusting var d.c for values of n of 1,
2, 3, 4, 5 corresponding to side frequencies of 455, 445, 435, 425, 415 kHz.

- 193 -
Repeat the above for all values of 
from 1 to 10 by setting Vp on your
function generator. Record your resu-
lts in a table as in fig 10.4. Divide all
your reading by the value you ob-
tained for J0(0) so that they are
normalized to J0(0) = 1 and draw up
another copy of fig 10.4.

Fig 10.4

Exercise 10.2

From the table in fig 10.5 at the end of this Practical, plot graphs of the
Bessel coefficients Jn() against  for each value of n from 0 to 5 and for
value of up to 10. We suggest you polt n = 0,2,4 on one sheet and n=1,3,5
on another sheet. Use linear graph paper. Also plot a graph of Jn() against n
for values of n up to 5 and for values of  of 2.5 and 10.
You will notice from this table that some of the coefficients are negative. This
only means a phase inversion and all side frequencies contribute to the total
wave power (power  component2). Our circuit will not recognise phase
differences, so we must take it for granted that this occurs. A circuit to
demonstrate this would be extremely complicated.

- 194 -
 Fig 10.5

Plot your experimental results on the same sheets of graph paper as the
theoretical values and where it is obvious that the curve has crossed the axis,
negate its coefficient in the table and plot the point accordingly.

It is convenient to proceed directly to Assignment 11 entitled FM detectors. In

this case the setting on the B sensitivity control should not be adjusted.
Values of D can then be set up directly.

Practical consideration
& applications

The first Practical in which a square wave was used, was designed to demo-
nstrate that when a signal p is used to frequency modulate another freque-
ncy , it changes the frequency above and below  by amounts corresponding
to the amplitude of the signal at that instant. It is important to grasp that the
total frequency deviation is proportional to the peak amplitude of the
modulating signal. The frequency of the modulating signal p, determines the
rate at which these frequency variations take place.

The mathematics of frequency modulation is a little cumbersome, but you

should at least appreciate the significance of the term deviation ratio. This was
defined as the ratio of D to p and it has an important bearing on the

- 195 -
magnitude of the side frequencies. We showed you that theoretically there are
an infinite number but as you saw in the second Practical, their amp- litudes
decrease rapidly. Fig 10.6 shows the effect of changing values of and you
should have been able to see this on your oscilloscope.

Fig 10.6

In practice, D is usually fixed in relation to the peak signal amplitude so that

as p increases decreases. This means that low frequency signals have more
significant side frequencies than high frequency signals. As fig 10.7 shows the
greater number of significant but closer spaced frequencies occupy about the
same bandwidth as fewer but wider spaced side frequencies.

You should have been able to see this from examination of your figures in fig
10.2 and it is very useful to remember that the bandwidth is roughly constant
for a given value of D. The changing values of the Bessel coefficients J()
make it very difficult to determine which side frequencies are going to be
significant. However it is found in practice that the number of significant side
frequencies lie between + 1 and + 4 depending on the quality required. An
alternative criterion is that all side frequencies needed for good reproduction
will lie between ±D, except for low values of
. High modulation signal frequencies may product significant side frequencies
outside this range but if speech or music is being transmitted, these will
contain low energy and the quality may not be seriously affected. Look again
at your table 10.2 and see how your figures compare with the above criteria.

For FM broadcasting the frequency deviation is usually fixed at ±75kHz. If the

- 196 -
maximum signal frequency for good quality music transmission is 15kHz, this
gives a minimum  value of 5. If we takes a criterion that  + 1 side
frequencies are going to be significant, there will be six pairs. The bandwidth
will then be 2 x 6 x 15kHz = 180kHz. Following the D criterion, the figure is
150kHz. In practice, receiver bandwidths are usually set at about 200kHz. If
AM had been used to transmit this 15kHz, the bandwidth would have been
30kHz. If  is less than about 0.5, there is only one pair

of significant side frequencies. This gives the same bandwidth as for AM but
small values of  do not permit the full advantage of FM in regard to
elimination of interference to be obtained.

Fig 10.7

In practice 10.3 you measured the values of the Bessel coefficients. You should
have found that the agreement between practical and theorectical values was
quite good. The theoretical values are shown in fig 10.8, for values of n up to
4. From this graph you can see that the coefficient J0() becomes zero at
values of equal to about 2.4, 5.5, 8.6, 11.8, 14.9, etc. At these values of , the
centre frequency does not exist and this is the reason for not calling the
frequency with in J0 coefficient the carrier frequency. In fig 10.6(e) and 10.7(a)
the centre frequency ther is small, since at 15 is close to the zero at 14.9.

- 197 -
Your graph of Jn() against n show you how rapidly the coefficients become
zero as decrease. We have said that the bandwidth required for good quality
FM is of the order of 200kHz. This cannot be provided within the normal
range of broadcast frequencies and thus FM is confined to frequencies about
above 30MHz. Fewer stations are allowed to operate on this range and those
that do can be allocated a generous bandwidth. Broadcast radio in Britain
operates in the range 88-100MHz and provides high quality transmission
modulated up to 15kHz. Communication FM systems, such as those used by the
police and other organizations, commonly use a maximum modulating frequency
of 3kHz (as in telephony) with a deviation of 15kHz. The minimum  is thus 5
and the bandwidth is about 50kHz. These transmissions may occupy
frequencies above or below the broadcast bands.

Fig 10.8

The outstanding advantage of frequency modulation is that it is relatively free

from interference, both from random electrical noise and from adjacent
transmissions. It can be shown that the performance is better for high values
of , but the whole subject is outside the scope of this trainer and you are
referred to your text book. We would mention that in broadcast FM, the high
frequencies are boosted at the transmitter and attenuated at the receiver. This
also attenuates any interference in the high frequency bands. This is known as
pre-emphasis and de-emphasis respectively and is specified by the time

- 198 -
constants of the networks used.

The method we have used to generate frequency modulated signals is by no

means the only one available and indeed is not often used in practice. However
this assignment has been concerned with the study of the FM wave, rather
than the method used to produce it. For details you are again referred to your
textbook. We can say that a common methed is to vary the reactance in tuned
circuit of an LC oscillator. This can also be achived using a variable
capacitance diode.

- 199 -
ASSIGNMENT - 11
- FM DETECTION

Object
To construct and examine the principles of operation of four different FM
detectors.

Equipment required
● Power Console ED-2900P
0-4 volts d.c, variable, metered.
+15 volts d.c
-15 volts d.c

● Oscilloscope, double beam, 5MHz, calibrated.

● Digital Frequency Counter, 1MHz
● Signal Generator

● Modules Type 2950A and C

● Tuned Circuit Assemblies Type 2950H : 2ea
● Crystal Filter Assembly Type 2950J

Approximate Time required

Two hours

Prerequisita Assignments
1, 2, 6, 8 and 10

- 201 -
Preliminary Procedure

1. Attach the console ED-2900P.

2. Connect the power supply unit(s) to the mains supply line.

DO NOT SWITCH ON YET.

3. Turn the variable d.c control knob to minimum.

4. If you are using any of modules 2950A, B, C, D or E, plug them into the
sockets on the Console in alphabetical order. If any change is required,
it will be indicated in the assignment. The act of plugging in the
modules connect manner. All sockets are identical.

5. If you are using either if the modules 2950F or G, or the Assemblies

2950H, J or K, these do not require any power and may be stood where
convenient or as directed in the assignment.

6. This completes the preliminary procedure.

Experimental Procedure

In Assignment 10 you studied the side frequency components of the FM wave

and saw how these were related to a defined parameter, the deviation ratio .
In this assignment we will show you four ways of recovering the original signal
information from the FM wave. This involves converting the frequency
variations into voltage variations and is called detection or demodulation.

To begin with we need to produce an FM wave. If you have proceeded

directly to this assignment from Assignment 10 your equipment will be ready
set up and you can start immediately on Practical 11.1. If not, we will repeat
the setting up instructions.

Setting-up instructions.

Connect up the circuit of fig 11.1

- 202 -
On the 2950A module
Set the variable attenuator to 0dB.
Carefully set the manual frequency control C so that the frequency
counter reads exactly 465kHz.

On the function generator

Set the frequency to 0. 1Hz.
Set the waveform to square wave.
Using the oscilloscope, exactly set the output waveform to 10V pk-pk.

On the 2950A module

Watching the frequency counter adjusting the B sensitivity control so
that the frequency changes either side of 465kHz by 100kHz.
That is, it changes from 365kHz to 565kHz.

Any non symmetry in these figures indicates a non-symmetrical square wave

from your function generator and you should take the average. Now if you do
not adjust the B control again during this assignment you can set known
frequency deviation directly by setting the input voltage, so that

10V pk-pk input Frequency deviation of 100kHz.

5V pk-pk input Frequency deviation of 50kHz.
2V pk-pk input Frequency deviation of 20kHz.

You may find it helpful to use the oscilloscope to make calibration marks on
the output amplitude control of your function generator. Do this at 100Hz.
Alternatively use a d.c voltmeter to read the peaks of the 0.1Hz square wave.

Fig 11.1

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Pracrtical 11.1 The Slope Detector

The slope detector is simply composed of a tuned circuit connected to the

signal source. To investigate the operation of the circuit, connect up the
layout of fig 11.2.
The signal source should have been set up according to the 'Setting-up Inst-
ructions'.

On the oscilloscope
Set the Y1 amplitude to 2 volts/division.
Set the timebase to 2ms/division.

Select a signal input of 1V pk-pk sinewave at p = 200Hz from your function

generator.

Fig 11.2

Q11.1 What value of frequency deviation D have you set up?

What the value of the deviation ratio (D/p)?

Adjust the controls for a stable display. Vary the setting of the tuning
capacitor on the tuned circuit module from end to end.

Q11.2 What happens to the trace, especially at both ends and the
centre frequency of 465kHz?

Let us explain what is happening. Recall the response of the tuned circuit you
studied in Assignment 2.

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Q11.3 How can this circuit give an output proportional to frequency?
Do you expect this output to be linear?
Set the scale on the tuned circuit to 500kHz.

Q11.4 What can you say about the waveform you see?

If the tuned circuit is tuned to a resonant frequency fo just away from the
centre frequency as shown in fig 11.3 the slope of the skirt of the
characteristic will provide an output voltage proportional to the input frequency.
As the frequency varies (FM) am amplitude modulated voltage appears across
the tuned circuit. There are of course two possible regions of operation, one
either side of the peak (If fo and  coincide, the var- iations occur equally on
each side of the peak and second harmonic distortion results). Since the slope
is non-linear, we would not expect a distortion-free output.

We can now detect the AM variations, using a diode detector as we did in

Assignment 8. Add the circuit of fig 10.4 to the output of the tuned circuit of
the set-up of fig 10.2. Increase the oscilloscope sensitivity to 0.5V/div and set
the d.c zero to near the bottom of the screen.

Fig 11.3

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Observe what happens now when you tune the tuned circuit from end to end.
Note the distortion present due to the non-linearity of the tuned circuit. Set
the tuned circuit calibrator to 500kHz. From your graph of the tuned circuit
from Assignment 2. estimate the change in output voltage for a frequency
deviation of 10kHz about 500kHz.

Q11.5 Where is the d.c level on your trace?

Temporarily remove the modulating signal input and note the position of the
d.c level.
Measure the output voltage above and below this d.c level.

Q11.6 How do these two measurements compare?

Are they equal above and below the centre level?

If you have a triangular wave available, select this and note the distortion.
Q11.7 How could you adjust the input and/or the tuning to obtain a more
linear output?

Fig 11.4

Set the oscilloscope to 0.2 volt/division.

Reduce the signal input amplitude to 0.4 volt pk-pk

Q11.8 What is now the frequency deviation D and the deviation ratio ?

Now AC-couple the oscilloscope and adjust the tuning until one peak of the

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displayed output begin to distort, then turn it back slightly.

Q11.9 Where are we now in the resonance curve? Use the calibration
scale to help you. What is the peak-to-peak output voltage?
Can you see any distortion?

We are now operating just to one side of the centre frequency on the steepest
part of the curve, hence we have maximum sensitivity. The use of a small
frequency deviation means we are only using a small portion of the curve,
hence better linearity.

Swing the setting to the other side of the peak and note the same effect on
the opposite skirt. Select a triangle wave and observe the better linearity.

However, as we have said, it is a disadvantage to use small values of  which

can occur with high frequency signals and low values of frequency deviation. In
practice this is not important because if the detector has only a small output
we can amplify it later. This you saw in Assignment 8.

Q11.10 Can you see any other ways to get a better linearity from the
tuned circuit slope?
In case you cannot guess, let us suggest that by deliberately damping the Q
value, we can 'flatten' the curve and make it more linear.

Return Vp to 1 volt pk-pk (D = 10kHz) and set the tuned circuit calibration to
500kHz.

Set the oscilloscope to 50mV/div and change the value of R in the diode
detector to 4.7k . Note the increased ripple due to the reduction of detector
time constant, so increase C to 20nF.

Tune the tuned circuit as before until the peak begins to distort, then turn it
back slightly.

Q11.11 What is the peak-to-peak output voltage? How does this campare
with before? How does the setting of the resonant frequency of
the tuned circuit compare with before?

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Select a triangle wave and note the better linearity. We have achieved good
linearity at high values of D, but the output is small. Connect it into the a.c
amplifier on the detector module and turn up the gain to maximum.

Q11.12 What is the peak-to-peak output? Is it distorted?

Change the frequency within the decade.

Q11.13 Does the output amplitude depend upon the signal frequency p?

Q11.14 Does the output amplitude depend on the input amplitude? What
would happen if, say, interference caused the strength of the
received signal to fade?

Practical 11.2 The Round-Travis Detector

Q11.15 Having done Practical 11.1 can you think of any other ways to
improve the linearity? Hint: Consider the use of two tuned circuits.

To see how we can make use of two tuned circuits connect in series, connect
up the layout of fig 11.5. In order to see what results, we shall use the swept
frequency technique.

Q11.16 Why do you think that the primaries of the tuned circuits are joined
in series and not in parallel?

On the oscilloscope
Set the timebase to 2ms/division.

On the 2950A module

Set the tuned circuit at right angles to each other (this will prevent
interaction).
Set each circuit tuning control to 465kHz.
DO NOT alter the B sensitivity control setting.
Now adjust the manual control C and the B sensitivity control, until the
two circuit responses are displayed. As explained in previous assignment,
these settings will depend on the nature of the timebase output from

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 your particular oscilloscope.
Vary the two tuned circuit controls.

Fig 11.5

Q11.17 What does the operation of each control do to the displayed peak?
Does the operation of one control affect the other tuned circuits?

Now transfer the output of the lower tuned circuit from diode D2 to diode D3
on the detector module, effectively reversing the diode.
Q11.18 What has this done to the traces?

Q11.19 Can you see how to combine these to give a response with a linear
portion?

If we could tune one circuit so that it were resonant just below our centre
frequency  , and the other one just above and add the two responses in
opposite sign, we would have their sum as shown in fig 11.6.

This could be done by using a differential amplifier, or more simply to re- vert
back to our original arrangement and connect the voltages in series opposition.
This method is called a phase sensitive rectifier and detailed descriptions of its

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operation can be found in many electronics textbooks.

Fig 11.6

It is known as the Round-Travis detector and its circuit is shown in fig 11.7.
To form the circuit, connect up the detector module to the rest of the circuit
as shown in fig 11.8. (see overleaf)

Exercise 11.1

Set the tuned circuit scales to opposite ends and sketch what you see. Now
adjust the two tuned circuits togther until they add to give the best
straight line section in the middle. Sketch this. You should note that you can
get either slope of the tuned circuit to add but the tuning is very critical.
When the two frequencies coincide, there should be zero output.

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 Fig 11.7

Q11.20 What centre frequency have you set up, i.e where the linear
center section of the curve cresses the axis?

We must now arrange that the centre of the linear portion of this curve
coincides with our set centre frequency of 465kHz so that deviations of
frequency will occur over the linear portion of this curve and be converted into
linear amplitude variations to recover our original signal. We can conveniently
do this using our crystal as a marker.

Simply connect the crystal between the 0dB output of the signal source and Y2
of your oscilloscope shown in fig 11.8.

On the oscilloscope
Set the Y amplifier to 1V/division .
Reduce the timebase speed to 500ms/division.

Leave everything else on your layout as it is.

Adjust the manual tuning control C to bring the crystal peak to the centre of
the screen. Now 465kHz is exactly in the centre of the screen. Under certain
conditions you may have to temporarily disconnect the tuned circuit input, in
order to see the crystal peak.

Disconnect the crystal and Y and increase the timebase back to 2ms/div and
adjust the tuned circuits for the best linear response with the centre of the

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linear portion of the curve in the centre of the screen. This curve is often
called an S-curve because of its shape. You have just set the centre of this
to 465kHz.

Disconnect the sweep input from the signal source and connect your digital
frequency counter to the -6dB output socket. Use the manual tuning control C
to move the output through the S-curve and observe the oscilloscope. You
should find, if you have done your setting up correctly, that the zero output in
the centre occurs at exactly 465kHz.

Fig 11.8

Q11.21 What is the frequency range from one end of the linear portion of
the S-curve to the other, as measured on the frequency counter?
What do you think is a suitable frequency deviation to use with
this detector?

Reset C at the centre frequency of exactly 465kHz.

Reconnet your function generator to input B of the signal source module.
Select a sinewave input at 200Hz and set the output to 0.8 volts pk-pk cor-
responding to a frequency deviation D of 8kHz. This should ensure you are

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well within the range of the S-curve linear section.

Q11.22 What do you see on your oscilloscope?

Where is its d.c level?
Measure its peak-to-peak voltage.

Temporarily select a triangular wave input and note the absence of distortion.
Increase the output above 1 volt pk-pk and note the distortion as you go over
the bend of the S-curve. Reset it back to 0.8Vpp. Vary the frequency p over
the decade.

Q11.23 Is the distortion or output amplitude dependent on frequency p?

Q11.24 Does the input amplitude affect the output amplitude?

As before the low output is no disadvantage, as we can amplify it. Connect up

the a.c amplifier and note the increased output.

Now you ahve completed your measurements, observe the affect of mistuning
first one, then both tuned circuits. Then try to return both circuits to tune
without going through the tuning procedure.

Q11.25 Is the Round-Travis detector easy or difficult to line up?

Do you consider it a better circuit than the simple tuned circuit
detector?

Practical 11.3 The Foster-Seeley Discriminator

Would it not be much easier to set up a detector circuit if all the tuned
circuits could be tuned to the same centre frequency? The Foster-Seeley
discriminator is such a detector. It is similar in many respects to the
Round-Travis circuit of Practice 11.2 but the method of producing the signals
for the diodes is quite different. It is difficult to deduce from first principles,
therfore we shall explain it to you.

The discriminator depends upon the phase relationship in two couples tuned

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circuits, both tuned to the same resonant frequency. Let us show you this
phase relationship.

Connect up the layout of fig 11.9.

On the 2950H module

Place the two tuned circuits 40mm apart.

On the oscilloscope
Set the amplifiers to 0.5 volt/division.
Set the timebase to 1 s/division.
Set only the Y1 trace on the screen.

Adjust the controls of both tuned circuits until a sinewave appears on the
screen then adjust the oscilloscope controls for a stable display. Finally adjust
each control individually until your display shows maximum amplitude.

Investigation of Phase Relationships in Coupled Tuned Circuits.

Fig 11.9

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Q11.26 Why do you have to adjust both controls, although you are only
looking at the output of one trace? Does the above procedure
identical setting on the two scales?

Now bring in the Y2 trace, connected to the secondary of the second tuned
circuit.

Q11.27 Are the two voltages in phase? Can you estimate their phase differ-
ence? This is best demonstrated on a Lissajou's figures type
of XY display. Switch your Y2 channel to the X axis.

Trim the second tuned circuit until a circle is displayed.

Q11.28 What does this tell you about the phase relationship between the
output of the two tuned circuits?

Q11.29 What effect does the following have on the display?

a) Adjustment of first tuned circuit.
b) Adjustment of second tuned circuit.
c) Variation of the mutual inductance (degree of coupling) by
variation of the physical separation.

Set them up to a position where the largest, symmetrical circle is displayed

and do not touch the tuning or the positions for the moments.

Q11.30 How can we obtain from the second tuned circuit, two voltages,
equal in magnitude but opposite in phase (sign) but both still
in quadrature (90 ) with the first tuned circuit?

To see if you are correct, earth the centre point of the second tuned circuit
and connect Y1 and Y2 to the two outer terminals. Return to timebase
operation at 0.5 s/div. Try not to disturb the tuned circuits while you do this.

- 215 -
Q11.31 What do you see?

Let us now explain why we should

want this. We will need to use
phasor diagrams. Suppose we add
the first voltage Vf to these two
second voltages VS1 and VS2. Fig
11.10 shows what happens. We get
two voltages VD1 and VD2, both
equal in magnitude and out of phase Fig 11.10
by the same amount on either side
of the phasor Vf.

If we use VD1 and VD2 as the inputs to diode detectors D1 and D2 as we did
previously, the d.c voltages produced will be equal but of oppo- site polarity.
If added together, they will cancel each other out and there will be zero
output.

Fig 11.11 shows a suitable method of

obtaining this arrangement and is in
fact the full circuit of the Foster
-Seeley discriminator you will bui- ld
in a moment. First we will descr- ibe
what happens at frequencies away
from resonance. You noticed that when
you mistuned the second tuned cir-
Fig 11.11 cuit, the phase of VS change rel-
ative to Vf.

This is what happens if the frequency deviates, say to maximum deviation in

one direction. The situation is shown in fig 11.12(a), VS1 and VS2 are still the
same size and of opposite sign to each other but no longer in quadrature with
Vf (You recall from Assignment 2 that a tuned circuit becomes relative at
either side of resonance). Thus VD1 and VD2 are no longer equal. The
voltage across R will be greater than that across R2 and point F will be
positive with respect to E.

- 216 -
 Fig 11.12

If the frequency is now deviated in the other directin by the same amount, the
situation of fig 11.12(b) occurs. In this case VD1 is less than VD2, the voltage
across R2 will be less than that across R. Thus point F will be negative with
respect to E. Thus if point E is earthed and the frequency deviated back and
forth, an alternating signal will appear at point F, corresponding to the original
modulating signal p.
You may be able to see from the geometry of the phasor diagrwms of fig
11.12 that there will be a limit to the region of linear operation of this cir-
cuit. The best way examine the response of is to use our swept frequency
technique.

Being careful not to disturb the tuned circuits, connect up the circuit as shown
fig 11.13.

- 217 -
 Module layout for Foster-Seeley Discriminator

Fig 11.13

On the 2950C module

The RC circuits in fig 11.13 are balanced by making R1 = R2 100k and
C1 = C2 = 1nF.

On the oscilloscope
Set the amplifier to 0.5 volt/division.
Set the timebase to 5ms/division.

On the 2950A module

Set the variable attenuator to 0dB.
Do not alter the B sensitivity control.
Adjust the sensitivity A and tuning C controls so that you obtain
a discriminator S-curve similar to one obtained for the Round-Travis
detector.

- 218 -
With your swept S-curve display in the centre of the screen, note the effect
of changing in turn the tuning of the first and second tuned circuits and
varying the separation between them.

Q11.32 Does mistuning the first or second tuned circuit move the
centre frequency?

Q11.33 How is it possible to tell which way off-tune either circuit is?

Q11.34 What happens to the magnitude and bandwidth of the reponse when
the mutual coupling (physical separation) is low or high?

Q11.35 Do you consider this detector easy to set up?

Sketch the traces for the above conditions.

Set up the detector to what you consider its optimum response. Remove the
sweep input from A and connect instead the variable d.c line. Connect your
digital frequency counter to the -6dB output and manually sweep through the
discriminator S-curve s before to note the bandwidth. It may be necessary to
adjust manual tuning control C.

Q11.36 What is this bandwidth? What would be a suitable frequency dev-

iation to use with this circuit? How does this compare with the
Round-Travis detector?

Disconnect the variable d.c and use the manual tuning control C to set the
centre frequency to that where your tuned circuits are lines up. (This is of
cource the opposite way round to a practical situation where you would line up
the tuned circuits onto an existing centre frequency). Connect your function
generator to input B, which should still be set up as before. Select a sinewave
at 200Hz pk-pk and slowly increase the output amplitude up to corresponding
to a frequency deviation of 8kHz. If you have set up the tuned circuits
correctly you should observe no distortion. Try a triangular wave and check
the linearity. Increase the input amplitude further and note the distortion
produced as you go outside the linear range of the S-curve.

- 219 -
Q11.37 Does the output depend at all on the signal frequency p or the
input amplitude? Is the circuit sensitive to amplitude variations?
Test this by adjusting the output amplitude (dB) control on the
signal source module.

Now that you have completed your tests, observe the effect on a signal of
mistuning both tuned circuits and of varying the coupling between them.
Finally, restore them as best you can to their original positions - you will use
them again in the next pratical. Also, do not disconnect your circuit.

Practical 11.4 The Ratio Detector

The Foster-Seeley discriminator coveres in practical 11.3 is sensitive to

amplitude variations. You remember that in fig 11.12 the output was taken
across capacitors C1 and C2. At the centre frequency, the two voltages applied
to the diodes VD1 and VD2 were equal and opposite and no output occured as
shown in the phasor diagram fig 11.10. When the frequency deviated either
way, these became unequal and a positive or negative output resulted.

Suppose an undeviated signal produced 1 volt across each capacitor. The

rectified voltages are in such a sence as to cancle, giving zero output. If the
input frequency deviated, say to cause VD1 to increase and VD2 to decrease,
the voltage across C1 may increase to 1.5volts, that across C2 may decrease
to 0.5 volt. The output is 1 volt, or the difference between the two, with the
earth (point E in fig 11.11) as a reference.

A stronger signal may produce 2 volts across each capacitor when undeviated.
When deviated the same amount as before the voltage across C1 increase in
the same ratio to 3 volts, that across C2 decreases to 1 volt. The output is
then 2 volts. The same frequency deviation has produced a stronger output.
This any amplitude modulation of the signal, by interference, will appear on the
output.

However, in both cases the RATIO of the voltages across C1 and C2 are the
same, namely 3:1. If a circuit could be made to respond to this ratio instead of
the absolute magnitudes, then it would be insensitive to amplitude variations

- 220 -
and have better interference rejection.

Modify the Detector Module in fig 11.14 to form the Ratio Discriminator of fig
11.15. Compared to the Foster-Seeley circuit the most important difference is
that the diodes are in the opposite direction and this is achieved by using
diode D3 instead of D2. Also the link between the junction of the resistors and
capacitors is removed with the capacitor junction now being earthed.

On the 2950A module

Set the variation attenuator to 0dB.
Do not alter the B sensitivity control.
Trim the sensitivity A and manual frequency C controls to observe the
discriminator S-curve on the display (The lining-up procedure is the same
as for Foster-Seeley circuit).

Module layout for Ratio Detector

Fig 11.14

- 221 -
Operation of the Ratio Dectector

Let us explain how the circuit in fig 11.15 works. The method of deriving the
voltages for the diodes is the same as for the Foster-Seeley circuit, and the
phasor diagrams of figs 11.10, 11.12(a) and (b) apply here also.

At the centre frequency (refer to fig 11.10) the diode voltages VD produced by
the in-phase and quadrature components (Vf and VS) are equal and opposite.
Because the diodes are in opposite directions, capacitors C1 and C2 charge in
opposite directions so that point F is positive and point E is negative with
respect to their junction, which is connected to point A. Thus a potential exists
between F and E. Consider now the circuit loop consisting of the second tuned
circuits containing both diodes, R1 and R2. A cir- culating current flows round
this loop, due to VS1 and VS2, producing a voltage across R1 and R2, the top
end of R1 being positive. Since the total voltage across the resistors and the
capacitors must be equal (this is the voltage between F and E) and R1 = R2
and C1 = C2, the voltage at the junction of C1 and C2 must equal that at the
junction of R1 and R2, in this case earthed, and there will thus be zero output.

Consider now what happens if the frequency is deviated. The operation of the
tuned circuits is the same as before and fig 11.12 applies.

Ratio Detector Circuit

Fig 11.15

- 222 -
In fig 11.12 the length of the phasor VD1 added to the length of the phasor
VD2 remains substantially constant. The voltages represented by these two
phasors are rectifies by D1 and D3 in fig 11.15 and the resultant voltages
appears across C1 and C2. By virtue of D3 being in the 'opposite' direction
with respect to D1, these voltages add to provide a substantially constant
voltage across F and E. At the centre frequency represented by fig 11.10 the
voltages across C1 and C2 will be equal so that output from the junction of R1
and R2 will be zero (R1 = R2). If however the frequency deviates within the
linear part of the S-curve then an output will appear at the junction of R1 and
R2 due to the inequality of the voltage across C1 and C2. So far this output
voltage thus produced is (like the Foster Seeley discriminator) also dependent
upon the input signal level to the system and is thus sensitive to amplitude
modulation. This is overcome in the Ratio detector by connecting a large
capacitor (often 8 F or so) C5 across E and F. In the absence of amplitude
modulation this has no effect since this voltage, as we have seen, stays
constant for frequnecy modulation within the linear part of the S curve. If
however there is a sudden signal amplitude increase only very slowly due to
the large value of C5 thus suppressing any audible response to the amplitude
increase.

It will be noticed that the S curve cannot be displayed if C5 is connect since

the action of C5 is, as explained above, to keep this voltage across E F
constant. This voltage can only be expected to remain constant for the
restricted linear range of the S curve characteristic then the action of C5 will
spoil the output.

Remove the sweep signal and apply a signal input of 0.8V pk-pk sinewave at
200Hz as before. Adjust the centre frequencies so that they coincide. Observe
the linearity and magnitude of the output. Try feeding it through the a.c
amplifier. If you are going to study AFC as described in Application Note 5,
now is a good time to move directly to it.

Optional Extra Practical-Closed-circuit FM system

Instead of using a signal generator, try feeding an audio source, e.g

microphone, tape recoder, etc, into the signal input and see if you can recover

- 223 -
the audio by connecting the speaker module to the output of the a.c amplifier.
You may find that the inclusion of a tuned amplifier in your set up gives better
results.

Practical considerations
& applications

Two important characteristics of any FM detector are its ease of alignment and
its linearity of output. You have investigated these during your practicals and
should have found that the Foster-Seeley and Ratio Detector circuit were
superior. The single tuned circuit was quite non-linear and not easy to set up,
as it had to be mistuned. The Round-Travis detector was difficult to set up,
but did give a reasonably linear output. The advantage of Foster-Seeley and
Ratio Detector circuits is that all tuned circuits can be set up at the same
frequency. Setting up is often done in practice using a swept display technique
as you have done here. The instrument for doing this is called a wobbulator.
This produces the characteristic discriminator S-curve with which you should
be quite familiar by now. Since the tuned circuit arrangements are the same
for Foster-Seeley and Ratio Detector circuits, you would expect the same
effects if they are mistuned and this is so. Figs 16, 17 and 18 show different
conditions.

Fig 11.16

- 224 -
 Effect on FM detector response of second tuned circuit

Fig 11.17

Effect on FM detector response curve of varying the mutual

coupling between the two tuned circuits

Fig 11.18

- 225 -
 Block diagram of FM receiver

Fig 11.19

You may not have been able to get quite the effect of fig 11.16 due to
interaction and a little of fig 11.19 may have been evident. However the main
tuning rules are (i) the first tuned circuit affects the symmetry, (ii) the second
tuned circuit affects the centre frequnecy, and (iii) the mutual coupling affects
the linearity and bandwidth.

The operation of all the detectors is fully described in the text. You will see
that all except the Ratio Detector are sensitive to amplitude variations. This
may be caused by interference. However, the great advantage of FM is that
the amplitude conveys no wanted information. Thus the signal can be limited
and the FM information still retained. If therefore the detector is preceded by
an amplitude limiter of some form (e.g a limiting amplifier) any randum AM
variations will be lost and reception will be free from interference. Thus
automatic volume control is not required in FM receivers.

No praticals have been carried out in this assignment on this subject and it is
left as a suggestion for further project work. The limiting amplifier was
exanmined in Assignment 4 and now you can see one of its uses.

The Ratio Detector circuit is inherently self limiting and although considerd
slightly more difficult to set up is often used in inexpensive domestic FM
receivers, as a separate limiter circuit can be dispensed with. The

- 226 -
Foster-Seeley is considered FM tuners where the advantages of a limiting
circuit in eliminating interference outweigh the extra cost involved.

FM receivers invariably use the superheterodyne principle, as discussed for AM

receivers in Laboratory sheet 9. A block diagram is shown in fig 11.19. The r.f
circuits must be designed for high-frequency reception (88-108MHz in Korean
for broadcast FM). The i. f must be high enough so that the circuits can
provise sufficient selectivity against image channel interference (see Lab.sheet
9) and sufficient bandwidth to pass all the significant side frequencies. This
dictates an i.f in the region of 10MHz; 10.7MHz is usually used. The circuit is
very similar to AM receivers until the limiter is reached. Here amplitude
variations are removed and the signal passed to the discriminator. The output
of the discriminator consists of signal frequency p, which is further amplified
and passed to the loud speaker.

Finally, we would mention one non-broadcast application of frequency

modulation - position measurement. If a moving object can be made to vary
the capacitance or inductance of an oscillating tuned circuit, frequency
modulation will result. This FM can be applied to a discriminator and a voltage
obtained proportional to the position of the object. This voltage can be
amplified and measured or recorded, or used as feedback to control the
position of the object. One advantage is that there can be no contact between
the object and the transducer.

- 227 -
APPLICATION NOTE

[1] OSCILLATORS

Assignment - 4 (in the Student's manual of the ED-2950 Communications Kit)

covered the different forms of the amplifier - linear, limiting, tuned, gain
controlled. As with any amplifier if positive feedback is applied, it can be made
to oscillate at a frequency dependent upon the component values in the
feedback loop.

Equipment Required

● Power Console ED-2900P

Power Supply Unit +15 volts d.c
Power Supply Unit -15 volts d.c
+15 volts d.c
-15 volts d.c
● Oscilloscope, 5MHz, calibrated
● Digital Frequency Counter, 1MHz
● module Type ED-2950B
● Tuned Circuit Assembly Type ED-2950H
● Crystal Filter Assembly Type ED-2950J

Practical 1. Effect of positive feedback on an amplifier

A tuned circuit (Fig.1) is a suitable choice and if the output of the tuned
collector amplifier is feedback to the base as shown, oscillations will be
observed. If not, the inputs to the tuned circuit must be reversed to obtain the
correct polarity.

Fig.1

- 229 -
 Fig. 2

Vary the frequency. If you have a triangle wave, try this also.
Increase the signal amplitude until you obtain a trace as in Fig. 2(b).

increase the signal amplitude still further to obtain a trace as in Fig. 2(c)
corresponding to over-modulation. the right-hand side of the display, indicated*
in the diagram, appears like this because the balances modulator responds to
negative inputs. In most modulation system, over-modulation reduces the
carrier to zero during a part if the modulation cycle and that section of the
trace become horizontal.

Reduce the signal back to the point where = 1 and then remove the variable
d.c input to the modulator thus removing the carrier. this corresponds to
suppressed carrier modulation.

Reset the frequency to 300Hz.

Now connect the Y and ground leads on the oscilloscope to the Hi and Lo
outputs of the 2950H module.
Slowly increase the frequency to 1kHz and then to 10kHz.'

Swept Frequency Displays

An output connected from the oscilloscope's timebase and connected to an

input on the signal source module 2960A will produce a varying frequency

output with each timebase sweep. if this output is connected via the crystal

- 230 -
module 2950J to the oscilloscope, it will show a sharp peak displayed at
exactly 465kHz. If we sweep the carrier, the three frequencies present in the
modulator output,  - p,  and  + p will each become 465kHz in sequence.
If the output is passed through a tuned circuit and crystal, the crystal will pick
out each component in turn to form a frequency domain display.

To carry out this practical you will need a long-persistence oscilloscope; a

storage type would be even better.

Practical 2. Use of frequency domain displays

Connect up the circuit of Fig. 3.

On the module 2950A : Set the variable attenuator to 0dB.

Connect input B to the timebase output on the oscilloscope

On the Console : Set the variable d.c t give a 0.5V reading on the
volt-meter.

- 231 -
On the Oscilloscope : Set the timebase to 500s/cm and the amplifier
to 0.5V/cm.

On the Function Generator : Set the output for a 1V p-p, 8kHz sinewave.
To tune the Signal source to the crystal adjust the manual tuning Control C on
the 2950A module until the response peak (produced when the swept carrier
frequency coincides with the crystal frequency) is on the screen. If not, adjust
sensitivity and frequency controls B and C on the 2950A module until this
occurs.

Now connect input a on the 2950D module to the function generator output.
Now adjust the controls so that the distance between each peak is exactly 4cm
and the centre peak is in the middle of the screen. although the frequency of
8Khz was chosen to make this easy to do, while still providing a good display,
it is nevertheless a sensitive operation and you should try to get it correct.
The actual settings will of course depend upon the timebase output from your
oscilloscope. adjust the anti-resonance trimming capacitor on the crystal for
the best display. you should now have a display as in Fig. 4.

As the timebase sweeps the carrier the three frequencies present in the
output,  - p,  and  + p will each become 465kHz in sequence and thus
be picked out by the tuned circuit and crystal. If these are lined up, each
components of frequency will be at the resonant peak at different time. The
bandwidth of the tuned circuit is then of no importance, whatever the value of
p.

- 232 -
Select a square-wave input at 1kHz. Note the spacing between the main peak,
side frequencies and the extra peaks that now appear on the display.

Summary

In the trapezoidal display shown in Fig. 2 you should appreciate that the wider
the picture, the greater the depth of modulation. The triangle is completed
when m becomes unity. You should also have appreciated that the straightness
of the edges of the display is a measure of the linearity of the modulator.
when we passed the output through a tuned circuit, the display became as in
Fig. 5.

From Assignment - 2 you will have appreciated that the phase shift through a
tuned circuit changes rapidly about resonance. The higher the frequency of the
modulating signal the greater the separation of the side frequencies from
resonance and the greater the phase shift. This curves the edges of the
display as in Fig. 5.

Fig. 5. Trapezoidal display showing phase shift of side frequencies

The frequency domain display shown in Fig. 4 clearly shows the side
frequencies. Appreciation of the method is very important and we will use it

- 233 -
again to determine the sideband structure of a frequency modulated wave. You
should take particular note of the width of the crystal peaks compared with the
spacing between the side frequencies, as this confirms the narrow bandwidth of
the crystal. when you reduced the setting of the variable attenuator on the
signal source module, You should have found that the whole output decreased
by the same amount. In practice, this would correspond to a variation in the
strength of the received signal.

Further separation causes noticeable attenuation as well as phase shift,

resulting in an apparent reduction in the value of m. This is known as SIDE
BAND CLIPPING. this is why the carrier frequency should be much higher than
the modulating frequency. If tuned circuits are used their bandwidth increases
with their resonant frequency. The higher the bandwidth the greater the
distortion-free bandwidth can be for the sidebands which contain the signal
information.

In fact the maximum frequency broadcast on medium wave AM transmitters is

4.5kHz. Thus the bandwidth requirement is only 9kHz for DSB transmission.
At carrier frequencies of about 1MHz this is easily achieved. If also means that
carrier frequencies in the medium have band can be separated by 9kHz.

- 234 -
 APPLICATION NOTE

[2] Effect of Bandwidth on Transmission of Speech

Laboratory Sheet 5 (in the Student's Manual of the Communications Trainer

ED-2950) contains experiments to determine the frequency response of passive
and active low-pass filters. there are 5-pole Butterworth filters, therefore
having a roll-off of 30dB/octave and the cut-off frequency of the active filter
is variable by shorting out various resistors. this table (Fig. 1) gives the
resistance and 3dB measured cut-off frequency for each position on the Active
Filter 2950E module.

Fig. 1

The following experiment is concerned to show how the cut-off frequency will
affect the legibility of transmitted speech. The principle of the experiment is
to speak slowly and clearly a number of words into a microphone connected to
a transmission channel through the variable active filter. A listener at the other
end of the channel writes down the word he thinks he has heard and his list is
subsequently compared with the sender's list. The percentage of correct words
is calculated and plotted against cut-off frequency, or more simply, position
number, for each setting of the filter. This is done with words of one and two
syllables.

Suitable lists of words are appended to this Application Note. There is a

separate list at the end of the Application Note for each of the 7-filter
positions, for both one-and two-syllable words, making fourteen lists in all.
this avoids any familiarity with the words which may result in inspired
guesswork. Each list contains 100 words but adequate results

may be obtained by using less than this. If you wish to make up your own

- 235 -
lists, for example in another language, try to avoid picking words that sound
the same but are spelt differently, or allow credit for these when marking. Thy
to avoid repeating words.
Equipment Required

● Power Console ED-2900P

Power Supply Unit 0-4 volts d.c, variable, metered
Power Supply Unit +15 volts d.c
Power Supply Unit -15 volts d.c
● Modules Type ED-2950C and E
● Speaker Assembly Type ED-2950L
● Suitable length intercommunication cable
● Microphone or Tape Recorder

Practical 1. Measuring the effect of cut-off frequency on the Transmission of

Speech Legibility

No definite circuit arrangement has been shown as this will depend upon the
auxiliary equipment available in each case but the following block diagram in
Fig. 2 shows the principle. Use screened leads wherever possible to avoid
hum pick-up.

Fig. 2 Block diagram of experiment to show how the cut-off frequency

affects the legibility of transmitted speech

- 236 -
We found it necessary to give the word number before each word, so a
changeover switch is necessary to provide a straight-through path. The
transmission channel may be a cable link to an amplifier and speaker in
another room but we used a stereo tape recorder, changing from
straight-through to restricted bandwidth by using the two recording level
controls alternately. We found that the person who spoke the words obtained
a much higher score than those who listened to them for the first time, so
we do not recommend self-testing. At least two people are required,
preferably more, so that an average of results can be obtained. A blank
should be inserted if a word is not recognised. Don't stop to think, and don't
guess! If a tape recorder is used, do not rewind the tape and repeat the test,
or unfair results will be obtained. We suggest doing the tests in a random
order of bandwidth, not simply increasing or decreasing the cut-off frequency.

The average results from our testers are shown in tabular and graphical form
in Fig. 3 and Fig. 4. Your results will almost certainly differ numerically, but
the following points should emerge.

The sharp drop in legibility indicates the bandwidth requirement. By referring

to the table of cut-off frequencies you should find that for one-syllable words
this begins at about 1.5kHz and at about 1kHz for two-syllable words.
The actual figures are not critical but you should note the sharp drop that
there also have a lower overall legibility.

If you try the experiment with complete sentences, a surprisingly low

bandwidth can be tolerated. This is so for human listeners, but machines are
not capable of using their experience to 'fill-in' badly received information and
are thus more sensitive to bandwidth restrictions.

Notice that this is smaller than the 3.4kHz cut-off frequency used in speech
telephony. The Greater bandwidth is needed to allow for a more relaxed
communication, especially where the signal is weak or in the presence of
noise.

- 237 -
- 238 -
 APPLICATION NOTE

[3] DISPLAY METHODS OF EXAMINING AM WAVEFORMS

This Application Note contains two further Practicals that can be carried out as
an extension to the Practicals in Assignment - 6.

Trapezoidal Display

This is a display mode used to examine distortion that can occur during the
modulation process. Its advantage is that it does not depend upon the input
modulating signal waveform.
the X deflection is proportional to the signal,
the Y deflection is proportional to the modulated signal.

A linear relationship between X and Y produces a trace whose outline is a

straight line if no distortion exists. This straight line occurs no matter what the
form of the signal. If the outline is not straight then distortion will be present.

Equipment Required

● Power Console ED-2900P

Power Supply Unit 0-4 volts d.c, variable, metered
Power Supply +15 volts d.c
Power Supply -15 volts d.c
● Oscilloscope, double beam, 5MHz calibrated
● Digital Frequency Counter, 1MHz
● Signal Generator
● High Impedance d.c voltmeter
● Modules Type ED-2950A and D
● Tuned Circuit Assembly Type ED-2950H
● Crystal Filter Assembly Type ED-2950J

- 239 -
Practical 1. Use of the trapezoidal display

Connect up the circuit of Fig. 1.

On the module 2950A : Set the frequency control C to the dot = 465kHz
Set the variable attenuator to -4dB

On the Console : Set the Var d.c to give a reading of 0.5V on the voltmeter

On the Function Generator : Set an output of 1V p-p, 300Hz sinewave

On the Oscilloscope : Set both inputs a.c coupled to 0.2V/cm

Fig.1

Display therefore the modulated wave on the Y-channel against the original
signal on the X-channel. You should obtain a TRAPEZOIDAL DISPLAY as in
Fig.2(a). The modulation index m can be calculated from

      
 
      

- 240 -
Fig. 2 Modulation Displays

- 241 -
 APPLICATION NOTE

[4] AMPLITUDE STABILISED OSCILLATOR

Assignment - 8 (in the Students' manual of the ED-2950 Communications

Trainer covers diode detection and shows how Automatic Gain Control can be
accomplished using a d.c voltage derived from the detector, feedback to control
the gain of one of the amplifiers.

If a diode detector is used to derive a d.c voltage proportional to the amplitude

of a signal from an oscillator, this voltage may be feedback to the oscillator to
control the amplitude. The polarity of this voltage must be such that if the
output decreases it must try to increase the amplitude of the oscillations.

Equipment Required

● Power Console ED-2900P

Power Supply Unit +15 volts d.c
Power Supply Unit -15 volts d.c
● Oscilloscope. 5MHz, calibrated
● Module Type ED-2950C and D
● Tuned Circuit Assembly Type ED-2950H

Practical 1. Use of AGC to control the output level of an oscillator

The amplifier module cannot be used as the polarities are incorrect but a
convenient oscillator may be built using the balanced modulator (described in
Assignment - 6) and a tuned circuit as shown in Fig. 1. The link on the
detector module biases the output of the d.c amplifier at -12 volts which starts
and maintains oscillations.

- 242 -
 Fig.1

Initially connect up the circuit but leave the input to the d.c amplifier open.

On the oscilloscope : Set the Y-amplifier to 2 volts/cm.Timebase to 1ms/cm.

On the 2950C module : Set the gain of the d.c amplifier to minimum.

You should now observe oscillations. If not, either reverse the input to the
tuned circuit, or transfer the control voltage from input (a) to input (a) on the
modulator.

Vary the tuning control from end to end and note that as the frequency varies,
the amplitude varies, because of the variation of Q of the tuned circuit, as
studied in Assignment - 2.

Now connect the detector output to its input (shown dotted in Fig. 1).
The amplitude of the oscillations should decrease slightly but there should be
no variation as the frequency is varied by the control.

Increase the gain of the d.c amplifier and note that the amplitude of oscillation
decreases but is still held constant if you change the frequency.

- 243 -
 APPLICATION NOTE

[5] DEMONSTRATION OF AUTOMATIC FREQUENCY CONTROL (AFC)

Assignment - 11 (in the manual for the ED-2950 Communications Trainer

covers FM detectors and demonstrates the typical S-curve, as shown in Fig. 1,
using a swept display technique.

It will be seen from this curve that when the tuning is correct and the center
frequency   coincides with the centre of the S-curve, the frequency
deviations occur equally on either side of the centre, and the output voltage is
symmetrical about zero. If the tuning is incorrect, say at point A on the curve
of Fig. 1, then distortion may result if one peak deviation is in excess of the
'knee' of the curve. However, the output will have a steady d.c level and this
can be used to correct the oscillator frequency to bring the circuit back into
turn. The d.c level can be separated from the output with an a.f filter adjusted
to the correct polarity and level and feedback to the oscillator.

Fig. 1 Discriminator S-curve

Either the Foster Seely or Ratio Detector discriminators described in Practicals

11.3 and 11.4 of the Student's manual are suitable for use in this circuit. The
output should be taken to the a.f filter on the Detector Module, consisting of
R5 and C7 , then directly into the d.c amplifier, the output of which is
feedback to input A of the Signal Source Module. A Block diagram is shown in
Fig. 2. Reference should also be made to the diagrams in Assignment - 11.

- 244 -
 Fig. 2 Block diagram of FM system with AFC

Equipment Required

● Power Console ED-2900P

Power Supply Unit 0-4 volts d.c, variable, metered
Power Supply Unit +15 volts d.c,
Power Supply Unit -15 volts d.c,
● Oscilloscope, double beam, 5MHz calibrated
● Digital Frequency Counter, 1MHz
● Signal Generator
● Modules Type ED-2950A and C
● Tuned Circuit Assemblies Type ED-2950H 2 Sets

Setting-up the circuit of Fig. 3.

On the 2950A module : Set the variable attenuator to 0dB.

Carefully set the manual frequency control C so that the frequency meter
reads exactly 465kHz.

On the function generator : Set the frequency to 0.1Hz

Set the wavwform to square wave
Using the oscilloscope exactly set the output waveform to 10V pk-pk

- 245 -
On the 2950A module :
Watching the frequency meter, adjust the B sensitivity control so that the
frequency changes either side of 465kHz by 100kHz.
That is, it changes from 365kHz to 565kHz.
Do not alter the settinh of the B sensitivity control from now on.
Now connect to the 0dB output the primary of one of the tuned circuits as
shown in Fig. 3.

On the 2950H module :

Place the two tuned circuits 10mm apart.
connect the 2950H/1 secondary outputs to the Y2 and earth inputs on the
oscilloscope.

- 246 -
On the oscilloscope :
Set the amplifiers to 0.5 volt/division.
Set the timebase to 1s/division.
Set only the Y2 trace on the screen.

Adjust the controls of both tuned circuits until a sinewave appears on the
screen then adjust the oscilloscope controls for a stable display. Finally adjust
each control individually until your display shows maximum amplitude.

Now connect the secondary of the 2950H/2 turned circuit to the Y2 and earth
inputs on the oscilloscope and adjust each turned circuit control again to obtain
a maximum amplitude display.

On the oscilloscope :
Set the amplifier to 0.5 volt/division.
Set the timebase to 5ms/division.

On the 2950H/2 module :

Connect the secondary outputs across diodes D1 and D3 on the Detector
Module.

On the 2950A module :

Connect the A input to the oscilloscope timebase output.
Set the variable attenuator to 0dB.
Do not alter the B sensitivity control.
Adjust the sensitivity and manual tuning C controls so that you obtain a
discriminator S-curve.

Practical 1. Effect of Automatic Frequency Control on an FM system.

The ratio detector has now been set up according to the setting-up instru-
ctions. The approximate control settings are shown in Fig. 3. Disconnect the
sweep frequency input to A. Then turn to a center frequency of 465kHz
without the AFC link connected and apply the signal such that operation takes
place just in the centre of the linear range of the discriminator S-curve. Take
the output through the a.c amplifier on the detector module and adjust the
again for a convenient display.

- 247 -
Adjust the manual tuning control C and note that only a slight mistuning causes
the output to become distorted and eventually disappear altogether. return the
tuning to the centre position.

Turn the A sensitivity control on the signal source module to maximum and the
gain of the d.c amplifier to minimum and connect the AFC feedback link. If
your tuning is correct you should see no difference to the output.

Fig.4 Holding and Capture Ranges of AFC

SLOWLY turn the manual tuning control C in one direction and note that the
output first becomes distorted then disappears altogether. The range is much
wider than before. Measure these frequency by temporarily disconnecting all
inputs and outputs and connecting a digital frequency meter instead. Repeat
for the other direction of tuning.

Set the tuning to each end in turn, then slowly bring it back and measure the
frequency at which the output suddenly reappears.

Record these measurements in a table as in Fig. 4. Express these figures as

ranges. They are called the Holding Range and Capture Range respectively.

Return the tuning to the center and slowly increase the gain of the d.c

- 248 -
amplifier until the output just begins to drop. Repeat the above measurements.
Typical results are shown in Fig. 4.

Thus the amount of AFC can be determined by the amplifier gain. If should
normally be arranged that the range is less than the separation between
transmitters or the receiver would lock into the stronger signal and the weaker
one could never be received. The capture ratio is a measure of how far the
receiver can be out of tune before the AFC brings in the desired transmitter.
The holding range is a measure of how far the transmitter (or receiver)
frequency may drift before the transmission is lost. In practice a switch is
often fitted to disable the AFC during turing. If this is set to anywhere within
the capture range, the receive will hold the signal when the AFC is restored.

As described in Assignment - 11, FM receivers are invariably superhete-

rodynes and the control is applied to the local oscillator, usually via a variable
capacitance diode placed across the tuned circuit of the oscillator. The tuning
of the radio frequency stages of the receiver will be slightly incorrect but
since AFC is only used for small corrections, this is of no importance. AFC is
rarely used on AM receivers, since it is fairly easy to tune to maximum
volume, which corresponds to correct tuning.

- 249 -
APPENDIX 1
- THE DECIBEL

When considering the 'loudness' experienced by a listener to a certain level of

sound, it is found that the loudness y varies according to the logarithm of the
sound wave intensity. The difference between two degrees of loudness, say
y1 and y2 can be expressed as :

y2 - y1 = log x2 - log x1 = log2 

where x2 and x1 are the corresponding sound wave intensities.

The logarithmic function is useful as it enables us to express large numbers

easily and lends itself to measurement of ratios. If the above is applied to
power ratios, we call the unit the BEL (after the inventor of the telephone).
This unit is too large for practical purposes, so we use the smaller unit, the
DECIBEL or dB. Ten decibels equal one bel. Thus a power ratio can be
expressed as :
 
Power Ratio = log10  Bels or 10 log10  dB
 
Where p1 and p2 are the two powers. The little subscript10 here shows that
we are using common logarithms (to base 10) as distinct from natural
logarithms (to base e).

If  is less than one, we have a power loss instead of a power gain, and


10 log10  will be negative.


 
This is easily calculated since log  = - log a.

For example suppose we wish to calculate the power gain where p2 = 10
watts and p1 = 5 watts.
 
here  =  = 2 gain = 10 log10 2 = 10 x 0.3010 ≒ + 3dB
 
But if the two powers had been p2 = 5 watts and p1 = 10 watts then
  
 =  gain = 10 log10  = -10 log10 2 ≒ -3dB I.e a loss of 3dB
  
We can thus apply the decibel equally well to gains and losses.
The decibel is fundamentally a unit of power ratio but it can also be applied to
current and voltage ratios.

- 250 -
Imagine a voltage V driving a current I through a resistor R.

Then Power P =   =    and the ratio between two powers   and   is

 




= 
    = 




Expressing this in dB
 

gain = 10  
 
= 10    



= 10   



 
= 20   = 20  
   

It is important to remember the figure '20' in front of this formula, which

applies if you are measuring current or voltage ratios, and use '10' only for
ture power ratios.

What happens if you have two (or more) stage cascaded together, all of which
have a power loss or gain expressed in dB as   ,   ,   and so on. The
input and output power of each are shown in Fig. A 1-1.

Fig. A 1-1

Considering each stage separately

  
  = 10   ;   = 10   ;   = 10  
  
But the total power gain or loss of the system is
   
  = 10   or   10   ∙  ∙ 
   
  
or   = 10   + 10    + 10    so   =   +   +  
     

The dB is a very useful unit as it is logarithmic. This enables gain or loss

figures expressed in dB to be simply added together and also reduces large
numbers to manageable proportions.

- 251 -
The table in Fig. A 1-2 shows a comparison of dB values with power and
voltage ratios. It is a good idea to commit certain numbers to memory, as
they represent important factors.
For instance :

3dB represents a power ratio of (almost) 2, a voltage ratio of  or


6dB represents a power ratio of (almost) 4, a voltage ratio of   or 2.
20dB represents a power ratio of (almost) 100, a voltage ratio of 

or 10

Fig. A 1-2

The term half power point, is for most practical taken as the point where the
attenation is 3dB.

Although the dB strictly refers to ratios, if a standard level of power is

assumed, then any absolute power can be expressed as so many dB above or
below the reference standard. This reference power is chosen as 1mW
(produced by 0.775 volt into a load resistance of 600 ).
The unit is then abbreviated dBm.

Thus a power of 4 watts could be expressed as

 × 
10   = 10 × 3.6021 = +36 dBm (approx.)


- 252 -
Fig. A 1-3 indicates some powers expressed dBm.

Fig. A 1-3

- 253 -
APPENDIX 2
- RELATION OF Q TO BANDWIDTH

In the tuned circuit of Fig. A 2-1, Q =    where  is the current flowing

in the capacitor and  is the net current supplied to the tuned circuit as
shown in the phasor diagram Fig. A 2-2. Figs. A 2-3 and A 2-4 show the
corresponding phasor diagram at the lower and upper half power points
respectively. These are the half power points because the value of  has
increased to   and   and is 2 times as big as  . This increase is caused
by the small and opposite changes in the length of  and  brought about by
the change in frequency.

It may be seen by simple geometry that the requisite change in  is

 
approximately +  and in  is -  .
 
   
I.e  =  +  but Q =  →  =  so  =  + 
   

but since   f. [ for   =  ]
 
   
 =  +   -  =  so  =  and so q = 
   
A formal and more exact proof, using complex numbers, of the above relation-
ship is to be found in most text books on the subject to which the student is
referred.

Fig. A 2-1 Fig. A 2-2

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 Fig. 4

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APPENDIX 3
- AMPLITUDE MODULATION

Double sideband Suppressed Modulation

The carrier wave can be represented by   cos  t (this is in fact the fun-
damental component of a square wave), where  is the carrier frequency.
The signal containing the information is denoted by   cos pt, where  is
the signal frequency. If these two signals are then multiplied the output  
will be :
  
  =  cos  t. cos pt

where K is the constant tactor of the multiplier. To this can be applied the
trigonometrical relationship; 2 cos x cos y = cos (x-y) + cos (x+y)
  
∴   =  [ cos ( -p)t + cos ( +p)t ]

This expression defines the waveform obtained in Assignment - 6 and implies
that the output consists of the sum of two separate signals each of half the
amplitude and of frequency  ±p. In other words there are two SIDE
FREQUENCY, each separated from the carrier frequency by an amount equal to
the modulating frequency. Notice however that the carrier frequency does not
appear in this expression. This is thus known as DOUBLE SIDEBAND
SUPPRESSED CARRIER MODULATION (DSSC) and is the reason such a circuit
is called a BALANCED MODULATOR.

Double Sideband Modulation

The carrier frequency can be made to appear in the output by the addition of
a d.c component to the signal. The signal now becomes   +   cos pt.
The output of the modulator is then

 =  cos  t (   +   cos pt)

This can be rewritten as   = E cos  t [ 1 + m cos pt ]

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where E is the unmodulated carrier amplitude and m is called the
MODULATION INDEX and defined as the ratio of signal amplitude to
unmodulated carrier amplitude. It is often expressed as a percentage.

The preceeding tringonometrical relationship

2 cos x cos y = cos (x-y) + cos (x+y) applied to this equation give
   
  =    cos t  [ cos ( -p)t + cos ( +p)t ]
 
or   = E { cos t + 1/2 m [cos( -p)t + cos ( +p)t] }
       
where m =  ÷  =  ( modulation index )
   

  
and E =  (carrier amplitude)


The first term in the above equation represents the carrier,the ( -p) term is
called the LOWER SIDE FREQUENCY and the ( +p) term is called the UPPER
SIDE FREQUENCY. This full AM wave is often called simply DOUBLE
SIDEBAND MODULATION (DSB).

Signal Bandwidth

We can still use these expressions if the signal is complex, e.g speech or
music. By Fourier Analysis we can consider the complex signal as made up of
a sum of sinewaves. We need then only consider the frequency band between
BANDWIDTH. When we are considering a band of frequencies instead if a
spot frequency, the side frequencies become the UPPER and LOWER
SIDEBAND (USB and LSB).

Notice that the side frequencies are each 1/2m times the amplitude of the
carrier. The phasor diagram of an AM waveform is shown in Fig. A 3-1, with
E as the carrier amplitude and the upper and lower side frequencies having
amplitudes of 1/2 mE. The side frequencies will rotate about the carrier with
an angular velocity of ±p with whole rotating at the carrier angular frequency
.

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 Fig. A 3-1 Phasor Diagram of armature

Power in a DSB waveform

Considering the power dissipated by this full modulated wave, as power is

proportional to the square of the voltage then :


Carrier Power ∝   Each side frequency power ∝ (1/2 mE)2 =  ∙  


  
∴ Total wavepower ∝               
 

 
and the fraction provided by both side frequencies =  
= 
      

I.e For m = 1 (100% modulation) total power = 1 1/2 x carrier power

each side frequency has 1/4 of carrier power or 1/6 of total power &
For m = 1/2 (50% modulation) total power = 1.1/8 x carrier power
each side frequency has 1/16 of carrier power or 1/18 of total power

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