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Teknikit
Analogue Communications
Students Workbook
53001S
Feedback
Feedback Instruments Ltd, Park Road, Crowborough, E. Sussex, TN6 2QR, UK.
Telephone: +44 (0) 1892 653322, Fax: +44 (0) 1892 663719.
email: feedback@fdbk.co.uk website: http://www.fbk.com
Manual: 53001S Ed03 072003 Printed in England by Fl Ltd, Crowborough
Feedback Part No. 116053001S
Notes
ANALOGUE COMMUNICATIONS 
STUDENTS WORKBOOK Preface
53001S i
THE HEALTH AND SAFETY AT WORK ACT 1974
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ANALOGUE COMMUNICATIONS 
STUDENTS WORKBOOK Preface
ii 53001S
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ANALOGUE COMMUNICATIONS 
STUDENTS WORKBOOK Contents
53001S TOC1
TABLE OF CONTENTS
1 Introduction 11
2 Assignments using the Amplitude Modulation Workboard 21
2.1 Amplitude Modulation with Full Carrier Assignment 21
2.1.1 Objectives 21
2.1.2 Practicals 21
2.1.3 Workboard Required 21
2.1.4 Theory 22
2.1.5 Practical 1: A Simple Amplitude Modulator 25
2.1.6 Practical 2: Envelope Detectors 29
2.1.7 Practical 3: Product Detection 212
2.2 Amplitude Modulation with No Carrier Assignment 215
2.2.1 Objectives 215
2.2.2 Practicals 215
2.2.3 Workboard Required 215
2.2.4 Theory 216
2.2.5 Practical 1: Double Sideband Suppressed Carrier 219
2.2.6 Practical 2: Generation of Single Sideband Suppressed Carrier (SSB) 222
2.2.7 Practical 3: Demodulation of SSB 225
3 Assignments using the Frequency Modulation Workboard 31
3.1 Generation of Frequency Modulation Assignment 31
3.1.1 Objectives 31
3.1.2 Practicals 31
3.1.3 Workboard Required 31
3.1.4 Theory 32
3.1.5 Practical 1: Concepts Of Frequency Modulation 35
ANALOGUE COMMUNICATIONS 
STUDENTS WORKBOOK Contents
TOC2 53001S
3.1.6 Practical 2: Generation of FM with a VCO 39
3.1.7 Practical 3: Spectrum of an FM signal with a large Modulation Index 311
3.2 Demodulation of Frequency Modulated Signals Assignment 313
3.2.1 Objectives 313
3.2.2 Practicals 313
3.2.3 Workboard Required 313
3.2.4 Theory 314
3.2.5 Practical 1: Quadrature Detector 316
3.2.6 Practical 2: Phase lock loop detector (PLL) 320
3.3 Limiters and the Effect of Noise on FM Demodulation Assignment 325
3.3.1 Objectives 325
3.3.2 Practicals 325
3.3.3 Workboard Required 325
3.3.4 Theory 326
3.3.5 Practical 1: A Quadrature Detector with Limiter 329
3.3.6 Practical 2: The Effect of Noise on a Quadrature Detector 332
3.3.7 Practical 3: PLL Detector with a Limiter 335
3.3.8 Practical 4: The Effect of Noise on a PLL Detector 337
4 Assignments using the Signal Sources Workboard 41
4.1 Wien Bridge Oscillator Assignment 41
4.1.1 Objectives 41
4.1.2 Practicals 41
4.1.3 Workboard Required 41
4.1.4 Theory 42
4.1.5 Practical 1: Basic Wien Bridge Oscillator 45
4.1.6 Practical 2: Amplitude Stabilisation 48
4.1.7 Practical 3: Changes from Standard 411
4.2 LC Oscillator Assignment 413
4.2.1 Objectives 413
ANALOGUE COMMUNICATIONS 
STUDENTS WORKBOOK Contents
53001S TOC3
4.2.2 Practicals 413
4.2.3 Workboard Required 413
4.2.4 Theory 414
4.2.5 Practical 1: TunedCollector Oscillator 417
4.2.6 Practical 2: Effect of Supply Variations 420
4.3 Crystal Oscillator Assignment 422
4.3.1 Objectives 422
4.3.2 Practicals 422
4.3.3 Workboard Required 422
4.3.4 Theory 423
4.3.5 Practical 1: Fundamental and Overtone Modes 427
4.4 Multivibrator Assignment 430
4.4.1 Objectives 430
4.4.2 Practicals 430
4.4.3 Workboard Required 430
4.4.4 Theory 431
4.4.5 Practical 1: Basic Circuit 435
4.4.6 Questions 438
4.4.7 Practical 2: Effect of Variable Supply 439
4.4.8 Practical 3: Mark/space Ratio Control 441
5 Assignments using the Tuned Circuits and Filters Workboard 51
5.1 Audio LowPass Filters Assignment 51
5.1.1 Objectives 51
5.1.2 Practicals 51
5.1.3 Workboard Required 51
5.1.4 Theory 52
5.1.5 Practical 1: Passive LowPass Filter 59
5.1.6 Practical 2: Passive LowPass Filter, Swept Frequency 512
5.1.7 Practical 3: Active LowPass Filter 514
5.1.8 Practical 4: Active LowPass Filter, Swept Frequency 517
ANALOGUE COMMUNICATIONS 
STUDENTS WORKBOOK Contents
TOC4 53001S
5.2 RF Selectivity Assignment 519
5.2.1 Objectives 519
5.2.2 Practicals 519
5.2.3 Workboard Required 519
5.2.4 Theory 520
5.2.5 Practical 1: L/C Tuned Circuit 524
5.2.6 Practical 2: L/C Tuned Circuit, Swept Frequency 526
5.2.7 Practical 3: L/C Tuned Circuit, Transient Response 528
5.2.8 Practical 4: Crystal Filter 531
5.2.9 Practical 5: Crystal Filter, Swept Frequency 533
5.3 RF BandPass Filters Assignment 535
5.3.1 Objectives 535
5.3.2 Practicals 535
5.3.3 Workboard Required 535
5.3.4 Theory 536
5.3.5 Practical 1: Coupled L/C Circuits 542
5.3.6 Practical 2: Coupled L/C Circuits, Swept Frequency 545
5.3.7 Practical 3: Ceramic Filter 548
5.3.8 Practical 4: Ceramic Filter, Swept Frequency 550
5.4 Tuned Amplifier with Gain Control Assignment 552
5.4.1 Objectives 552
5.4.2 Practicals 552
5.4.3 Workboard Required 552
5.4.4 Theory 553
5.4.5 Practical 1: Gain Control 555
5.4.6 Practical 2: Automatic Gain Control 558
5.4.7 Practical 3: Frequency Response with Automatic Gain Control 561
5.4.8 Practical 4: Decibel Gain 563
Chapter 1
ANALOGUE COMMUNICATIONS 
STUDENTS WORKBOOK Introduction
53001S 11
1 Introduction
This manual provides computerbased assignments which make use of Analogue
Workboards, the Discovery II environment and Analogue Telecommunications 53921
software package to provide an understanding of the fundamental principles on which
complex analogue communication systems are based.
Assignments are divided into practicals whose objectives are clearly defined. Every
practical is designed to contain as much circuit investigation, measurement and
observation as possible. Explanatory text, diagrams and instrumentation are fully
integrated.
Details of hardware and software installation are given in manual 530013 together with
Discovery II environment and product operating instructions.
Chapter 1
ANALOGUE COMMUNICATIONS 
STUDENTS WORKBOOK Introduction
12 53001S
Notes
Chapter 2
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Amplitude Modulation Workboard
53001S 21
2 Assignments using the Amplitude Modulation Workboard
2.1 Amplitude Modulation with Full Carrier Assignment
2.1.1 Objectives
On completion of this assignment you will be familiar with:
Basic amplitude modulation and demodulation,
AM characteristics in the time domain,
AM characteristics in the frequency domain,
Envelope detectors,
Product detectors.
2.1.2 Practicals
Practical exercises are provided as follows:
Practical 1: A simple amplitude modulator
Practical 2: Envelope detectors
Practical 3: Product detection
2.1.3 Workboard Required
Amplitude Modulation Workboard 53130 which comprises the following blocks:
Signal Generation
Modulation
Filters
Demodulation
Chapter 2
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Amplitude Modulation Workboard
22 53001S
2.1.4 Theory
2.1.4.1 Modulation
The equation of a sinusoidal voltage waveform is given by:
v = V
max
.sin(t+)
where:
v is the instantaneous voltage
V
max
is the maximum voltage amplitude
is the angular frequency
is the phase
A steady voltage corresponding to the above equation conveys little information.
To convey information the waveform must be made to vary so that the variations
represent the information. This process is called modulation.
Any of these may be varied to convey information.
2.1.4.2 Amplitude Modulation
Amplitude modulation uses variations in amplitude (V
max
) to convey information. The wave
whose amplitude is being varied is called the carrier wave. The signal doing the variation
is called the modulating signal.
For simplicity, suppose both carrier wave and modulating signal are sinusoidal; ie,
v
c
= V
c
sin
c
t (c denotes carrier)
and
v
m
= V
m
sin
m
t (m denotes modulation)
We want the modulating signal to vary the carrier amplitude, V
c
, so that:
v
c
= (V
c
+ V
m
sin
m
t).sin
c
t
where (V
c
+ V
m
sin
m
t) is the new, varying carrier amplitude.
Chapter 2
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Amplitude Modulation Workboard
53001S 23
Expanding this equation gives:
v
c
= V
c
sin
c
t + V
m
sin
c
t. sin
m
t
which may be rewritten as:
v
c
= V
c
[sin
c
t + m sin
c
t. sin
m
t]
where m = V
m
/V
c
and is called the modulation index.
Now:
sin
c
t.sin
m
t = (1/2) [cos(
c

m
) t  cos(
c
+
m
) t]
so, from the previous equation:
v
c
= V
c
[sin
c
t + m sin
c
t. sin
m
t]
we can express v
c
as:
v
c
= V
c
sin
c
t + (mV
c
/2) [cos(
c

m
) t]  (mV
c
/2) [cos(
c
+
m
) t]
This expression for v
c
has three terms:
1. The original carrier waveform, at frequency
c
, containing no variations and thus
carrying no information.
2. A component at frequency (
c

m
) whose amplitude is proportional to the
modulation index. This is called the Lower Side Frequency.
3. A component at frequency (
c
+
m
) whose amplitude is proportional to the
modulation index. This is called the Upper Side Frequency.
It is the upper and lower side frequencies which carry the information. This is shown by
the fact that only their terms include the modulation index m. Because of this, the
amplitudes of the side frequencies vary in proportion to that of the modulation signal.
2.1.4.3 Sidebands
If the modulating signal is a more complex waveform, for instance an audio voltage from a
speech amplifier, there will be many side frequencies present in the total waveform.
This gives rise to components 2 and 3 in the last equation being bands of frequencies,
known as sidebands.
Hence we have the upper sideband and the lower sideband, together with the carrier.
Chapter 2
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Amplitude Modulation Workboard
24 53001S
2.1.4.4 Experimental Determination of the Modulation Index
This is most easily done by measuring the maximum and minimum values which the
instantaneous amplitude of the carrier reaches. Let us call these x and y.
Taking our previous equation:
v
c
= V
c
[sin
c
t + m sin
c
t. sin
m
t]
and rearranging it yet again, we can express v
c
as:
v
c
= V
c
sin
c
t [1 + m sin
m
t]
so that the instantaneous amplitude of the carrier is:
V
c
[1 + m sin
m
t]
Since sin w
m
t can vary between +1 and 1,
x = V
c
(1 + m) and y = V
c
(1  m)
To get the value of modulation index m from x and y, we eliminate V
c
between these
equations by division, giving:
y /x = (1  m)/(1 + m).
Solving for m gives:
m = (x  y)/(x + y)
Chapter 2
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Amplitude Modulation Workboard
53001S 25
2.1.5 Practical 1: A Simple Amplitude Modulator
This practical introduces the concept of Amplitude Modulation.
You will meet the terms Carrier, Modulation and Modulated Signal and how they are
related in both the time domain and frequency domain.
A simple form of information transfer is Morse code, where the signal at the frequency
selected for transmission is switched off and on in dots and dashes. The transmission
frequency is selected by consideration of the transmission medium and, within reason,
has nothing to do with the information it has to carry. For this reason it is called the carrier
frequency.
In order to carry any information some characteristic of the carrier must be changed, or
modulated with that information; hence the term modulating signal. In the Morse code
example where the carrier is switched off and on, it is the amplitude of the carrier that is
carrying the information.
This is a very crude form of Amplitude Modulation (or AM) because there are only two
states: zero amplitude and full amplitude. In order to carry more complex information such
as speech or television, the amplitude is varied linearly so that the instantaneous carrier
amplitude is proportional to the amplitude of the modulation signal.
Clearly, for a given carrier amplitude there are limits for the size of the modulating signal;
the minimum must give zero carrier, the maximum gives twice the unmodulated carrier
amplitude. If these limits are exceeded, the modulated signal cannot be recovered without
distortion and the carrier is said to be overmodulated.
When the modulating signal is varying the carrier from zero to twice its amplitude, the
carrier is said to be fully, or 100%, modulated. Modulation depth is calculated from the
formula:
(x y) / (x + y)
where x is the maximum instantaneous carrier amplitude and y the minimum. The
resulting fraction is often expressed as a percentage. If the fraction exceeds 1
(modulation depth greater than 100%), then the carrier is said to be overmodulated.
2.1.5.1 Amplitude Modulation in the Time Domain
One of the easiest methods of examining a signal is by using the oscilloscope. This is in
the time domain.
The amplitude variation of the carrier in time with the modulation can be seen easily and
the operation of a system analysed.
Chapter 2
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Amplitude Modulation Workboard
26 53001S
In the practicals we use a sine wave as the modulation source as this makes the
oscilloscope pattern clearest, in a practical system the modulating signal could be very
complex. Also, to make the principles clearer, the modulating frequency is one ninth of the
carrier so that individual carrier cycles show.
Again, in a practical system, the modulating frequency would probably be lower. In order
for the oscilloscope trace to be stable it must be triggered from the modulation source.
2.1.5.2 Amplitude Modulation in the Frequency Domain
The use of a spectrum analyser for examining signals in the frequency domain is a very
powerful tool. However, this facility is not always available, as a spectrum analyser costs
much more than an oscilloscope. As an aid to understanding modulation principles it is
invaluable.
The spectrum analyser shows the modulated signal to have three components:
1. The Carrier, at the same frequency as the carrier source.
2. A lower side frequency at the carrier minus the modulation frequency.
3. An upper side frequency at the carrier plus the modulation frequency.
The amplitude of the carrier is independent of the modulation, while the amplitude
of the side frequencies depends entirely on modulation depth.
In the practical the modulation source is a sine wave, containing only one frequency, and
therefore is represented by a narrow line in the frequency spectrum.
In practice, where the modulating signal is more complex, there would be a range, or
band, of side frequencies above the carrier frequency and a band below it, the upper and
lower sidebands. They extend either side of the carrier to an extent equal to the maximum
modulating frequency.
Chapter 2
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Amplitude Modulation Workboard
53001S 27
2.1.5.3 Procedure
In this practical the hardware is configured as shown. You have available an oscilloscope
and a spectrum analyser. Set the carrier level to maximum. Set modulation level to
zero. Note the signals at all monitoring points. Now increase the modulation level and
observe at monitor point 6.
Increase the modulation level until the carrier amplitude just reaches zero on negative
modulation peaks. This is 100% modulation. Observe the signals at all the monitoring
points both with the oscilloscope and the spectrum analyser at various modulation levels.
Also, with a fixed modulation level try adjusting the carrier level.
You will need to return to the practical and make some measurements in order to answer
the questions.
Chapter 2
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Amplitude Modulation Workboard
28 53001S
2.1.5.4 Questions
1. The 'envelope' of the modulated carrier wave is a curve joining its peaks. The positive
envelope, joining the positive peaks, should follow the shape of the modulating signal
in one polarity and the negative envelope, joining the negative peaks, in the opposite
polarity. What happens to the positive and negative envelopes when overmodulation
occurs?
2. How would you recognise overmodulation on the spectrum analyser display?
3. What is the amplitude of the two sidebands relative to the carrier expressed in dB for
50 percent modulation with a sine wave? (HINT: Use the oscilloscope to set the
modulation level and the spectrum analyser to measure the sidebands)
4. See if you can use the theory mathematics to calculate the value in question three and
compare it with the practical measurement.
Chapter 2
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Amplitude Modulation Workboard
53001S 29
2.1.6 Practical 2: Envelope Detectors
In this practical you will investigate the demodulation of an AM signal with an envelope
detector.
The purpose of any detector or demodulator is to recover the original modulating signal
with the minimum of distortion and interference. The simplest way of dealing with an AM
signal is to use a simple half wave rectifier circuit. If the signal were simply passed
through a diode to a resistive load, the output would be a series of halfcycle pulses at
carrier frequency. So the diode is followed by a filter, typically a capacitor and resistor in
parallel.
The capacitor is charged by the diode almost to the peak value of the carrier cycles and
the output therefore follows the envelope of the amplitude modulation. Hence the term
envelope detector.
The time constant of the RC network is very important because if it is too short the output
will contain a large component at carrier frequency. However, if it is too long it will filter out
a significant amount of the required demodulated output.
In this practical the output of the AM generator that you used in the Simple Amplitude
Modulator practical is fed to an envelope detector.
You can monitor the output and compare it with the original modulation source. The time
constant of the filter following the detector can be adjusted. This filter is often called a
postdetection filter. It also introduces a phase shift between the original signal and the
output.
Chapter 2
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Amplitude Modulation Workboard
210 53001S
2.1.6.1 Procedure
Here the signal from the amplitude modulator from the previous practical is demodulated
using an envelope detector. Confirm that the modulated signal is the same.
Use the oscilloscope to monitor the detector output 16 and adjust the time constant.
Note that a large carrier component is present if the time constant is too short.
Increase the time constant and note that the amplitude of the detected output decreases
and becomes distorted as the filter cannot discharge in time to follow the required output.
Use the spectrum analyser to observe the carrier component amplitude.
Compare the original modulating signal with the detector output in both shape and phase
at various time constants using the oscilloscope.
Chapter 2
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Amplitude Modulation Workboard
53001S 211
2.1.6.2 Questions
1. Is the phase shift caused by the post detection filter a lead or lag?
2. Why do you think that the filter causes a phase shift?
3. How does the ratio of modulating frequency to carrier frequency affect the design of
the detector and the post detection filter?
4. What problems could be caused if the range of modulating frequencies was quite
large?
Chapter 2
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Amplitude Modulation Workboard
212 53001S
2.1.7 Practical 3: Product Detection
In this practical you will investigate an alternative demodulator called a product detector. It
has certain advantages over the simple envelope detector but at the expense of some
complexity.
It is not often used for AM but is the only type of detector that will demodulate the
suppressed carrier amplitude modulations that are investigated in the next assignment.
It is important to appreciate that a product detector will demodulate all forms of
AM.
2.1.7.1 What is a Product Detector?
If the AM signal is mixed with (ie, modulated by) a frequency equal to that of its carrier,
the two sidebands are mixed down to the original modulating frequency and the carrier
appears as a dc level.
The mathematics of the process show that this will only happen if the mixing frequency is
equal not only in frequency to that of the carrier, but also in phase; ie, the two signals are
synchronous. This is why a product detector when used for AM is sometimes called a
synchronous detector. For AM the effect is very similar to a fullwave rectifier rather than
the halfwave of the envelope detector.
The output still needs a postdetection filter to remove the residual ripple, but this time the
ripple is at twice the carrier frequency and is therefore further away from the modulation
and hence easier to remove. In general terms the product detector gives less distortion,
partly because it uses both positive and negative peaks of the carrier.
2.1.7.2 Generating the Mixing Frequency
This is produced by an oscillator which is usually referred to as a Beat Frequency
Oscillator or BFO. This is because if it is not at the same frequency as the carrier the
output of the product detector is a frequency equal to the difference between the two
which is called a beat frequency. (You will be able to see this when you adjust the BFO for
synchronism).
As previously described, it is vital that the BFO be synchronised to the carrier. In practice
this is achieved with a special recovery circuit but here for simplicity a sample of the
carrier is fed directly to the BFO and when the free running frequency of the BFO is near
to that of the carrier it locks into synchronism.
Chapter 2
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Amplitude Modulation Workboard
53001S 213
2.1.7.3 Procedure
Here is a product detector demodulating AM. The oscilloscope shows its input at monitor
point 6, which is the output of the same modulator as before.
Now monitor the BFO output with the oscilloscope and use the BFO frequency control to
lock it to the carrier. This will be indicated by a stationary trace.
Use the oscilloscope to look at the output of the detector before the filter and note the
frequency of the ripple compared with the carrier. Use the spectrum analyser to confirm
this. Examine the output of the filter and compare it with the modulation source.
Monitor the detector output before the filter with the oscilloscope, then unlock the BFO
with the BFO frequency control and observe the result. Repeat whilst observing the
filtered output.
Chapter 2
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Amplitude Modulation Workboard
214 53001S
2.1.7.4 Questions
1. Are the design considerations for a postdetection filter different from those for the
envelope detector?
2. Examine the filtered output, using the spectrum analyser at large size, with the BFO
synchronised. The trace should show three points where the level is above the
background ripple. What do they represent?
3. Again examine the filtered output, using the spectrum analyser at large size. Decrease
the amplitude of the modulation signal as far as possible without the instrument trigger
failing. Then vary the BFO control. How wide is the available range of beat frequency?
Chapter 2
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Amplitude Modulation Workboard
53001S 215
2.2 Amplitude Modulation with No Carrier Assignment
2.2.1 Objectives
On completion of this assignment you will be familiar with:
Amplitude modulation with suppressed carrier,
Double sideband suppressed carrier (DSB) modulation,
Single sideband suppressed carrier (SSB) modulation,
Balanced modulators,
Generating SSB with filters,
Demodulation methods.
2.2.2 Practicals
Practical exercises are provided:
Practical1: Double sideband suppressed carrier
Practical 2: Generation of single sideband carrier (SSB)
Practical 3: Demodulation of SSB
2.2.3 Workboard Required
Amplitude Modulation Workboard 53130 which comprises the following blocks:
Signal Generation
Modulation
Filters
Demodulation
Chapter 2
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Amplitude Modulation Workboard
216 53001S
2.2.4 Theory
2.2.4.1 Double Sideband Suppressed Carrier
In the theory for the Amplitude Modulation with Full Carrier assignment, Practical 1, it was
established that if:
v
c
= V
c
sin
c
t
describes a carrier signal and,
v
m
= V
m
sin
m
t
describes a modulating signal, then the normal AM signal is:
v
c
= (V
c
+ V
m
sin
m
t) .sin
c
t
which may be rewritten as:
v
c
= V
c
sin
c
t + V
m
sin
c
t . sin
m
t
In DSB suppressed carrier modulation, the carrier term V
c
sin
c
t is suppressed, leaving
just:
V
m
sin
m
t.sin
c
t
= (V
m
/2) [cos(
c

m
) t  cos(
c
+
m
) t]
as the modulated signal.
The two cosine terms represent the lower and upper sidebands respectively.
In the case of SSB suppressed carrier modulation, one of these sidebands will also be
suppressed.
2.2.4.2 Demodulating the DSB Signal
In order to change the sideband frequencies back to the original modulating frequency, a
locallygenerated carrier frequency (from the BFO) is used to modulate the DSB signal.
(Remember that modulation for the purpose of frequency changing is traditionally called
mixing).
Suppose that the BFO signal is:
v
o
= V
o
sin(
o
+ )
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The modulation process will produce a signal proportional to:
[V
o
sin(
o
+ )] (V
c
/2) [cos(
c

m
) t  cos(
c
+
m
) t]
or to:
2 sin(
o
+ ) [cos(
c

m
) t  cos(
c
+
m
) t]
This can be divided into two terms:
2 sin(
o
+ ) cos(
c

m
) t ... (1)
and:
 2 sin(
o
+ ) cos(
c
+
m
) t ... (2)
but as
2 sin A cos B = sin(A + B) + sin (A  B)
the first term, (1), becomes:
sin(
o
+ +
c

m
) t + sin(
o
+ 
c
+
m
) t
Since
o
is supposed to be equal to
c
, (
o
+
c

m
) will be a frequency roughly twice
that of the carrier.
This does not contribute to the desired signal. The rest of the expression, which does
contribute, is:
sin(w
o
+ 
c
+
m
) t
If
o
=
c
, then sin(
o
+ 
c
+
m
) t becomes simply
sin( +
m
) t,
which is the original modulating frequency. Similarly the other term, (2), makes a
contribution:
 sin(
o
+ 
c

m
) t
which, for
o
=
c
, becomes:
sin( +
m
) t
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We now have two terms, or components of the output signal, each of the original
modulating frequency. However, there is a problem when we combine them.
The two terms are:
sin( +
m
) t
and
sin( +
m
) t
If the phase is zero, the two terms become identical, so they combine to produce the
signal:
2 sin
m
t
i.e a signal at the original modulating frequency.
Now suppose that the phase now changes through /2 radians (90 degrees).
The two sinusoids would now be radians (180 degrees) apart in phase and would
cancel each other out.
We have assumed that
o
=
c
. If this were not true, the effect would be the same as if
were continually changing, making the two terms alternately reinforce and cancel each
other.
This may be shown mathematically thus:
sin( +
m
) t + sin( +
m
) t = 2 sin
m
t cos
Since cos 0 = 1, the strongest output is obtained for = 0.
With = /2, cos = 0, so no output is obtained.
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2.2.5 Practical 1: Double Sideband Suppressed Carrier
This practical introduces the idea of AM with suppressed carrier. After it you will
understand the following ideas:
Balanced modulators and carrier suppression
The BFO as a carrier insertion oscillator
In the assignment on basic amplitude modulation we saw that the modulated signal
comprises a carrier and two sidebands.
The carrier is of constant amplitude and only the sidebands vary in frequency and
amplitude with the modulation. It is therefore clear that only the sidebands carry the
modulating information while the carrier does nothing except, as we will see, help in the
demodulation process.
The transmission of the carrier takes a large proportion of the total transmitted power, so
if the carrier were removed all the power could be used to transmit the sidebands which,
after all, contain the information.
If the modulation process is carried out by a balanced modulator, the output signal does
not contain the carrier component because it is cancelled out by the balanced nature of
the modulator.
This signal is described as double sideband, suppressed carrier or DSB.
2.2.5.1 Carrier Unbalance
If the modulator were perfectly balanced there would be no carrier in the output.
In practice, due to circuit imperfections, some carrier is always present.
The ratio of the actual carrier to that which would be there in a simple AM system is called
the carrier suppression ratio and is an important parameter in such systems. Normally
the ratio is expressed in dB to make the numbers manageable; 30 dB would be a typical
figure.
To calculate the carrier suppression ratio, you need to know what amplitude of carrier
would have been present, if not suppressed. This is the carrier which would give 100%
modulation by the maximum signal level for which the system is designed.
Since 100% modulation produces side frequencies of half the carrier amplitude, the
unsuppressed carrier amplitude may be taken as twice the allowable amplitude of either
sideband.
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In this practical, a carrier and modulation source are connected to a balanced modulator
to provide a DSB signal which can be examined with the oscilloscope and spectrum
analyser.
2.2.5.2 Procedure
Use the oscilloscope and spectrum analyser to examine the signals at monitor point 4 and
monitor point 5.
Set the carrier balance to midscale. Note that they are the same as for simple AM. Now
examine at monitor point 6 and note the waveshape.
Use the spectrum analyser to observe that there are two sidebands but no carrier.
Adjust the carrier balance; note the effect on carrier amplitude. Adjust modulation level
and carrier level and note the effects.
Note that the output from the envelope detector is not the same as the modulating signal.
Monitor at monitor point 13 and adjust the BFO frequency for a stable trace, so that the
BFO is in phase with the original carrier. Observe that the product detector output is the
same as the modulating signal. Unlock the BFO and observe the result.
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2.2.5.3 Questions
1. Why does AM have a low efficiency when the full carrier is transmitted?
2. How can you tell whether the modulator is balanced when using the oscilloscope?, and
when using the spectrum analyser?
3. Measure the carrier suppression ratio for the system in Practical 1 when set for
maximum modulation and minimum carrier amplitude.
4. Does the term overmodulation have any meaning in a DSB system?
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2.2.6 Practical 2: Generation of Single Sideband Suppressed Carrier (SSB)
This practical introduces the concept of single sideband suppressed carrier or SSB
operation.
In the double sideband suppressed carrier practical it was demonstrated that it was
possible to recover the modulating signal without the presence of a transmitted carrier.
This is achieved by inserting a local carrier at the demodulator. That practical used DSB;
ie, both sidebands were transmitted.
It is obvious that as both sidebands are generated from the same carrier and modulation,
they must contain the same information, and therefore the modulating frequency could be
recovered from only one sideband. This saves further transmitter power.
Another very important advantage is that the bandwidth is half that of simple AM or DSB.
2.2.6.1 Generating SSB
The generator in the practical is a balanced modulator, producing DSB, followed by a
bandpass filter for the required sideband.
There are other methods but this filter method is the simplest to understand and is in
very common usage in communication systems. It may be necessary for the bandpass
filter to have a very good shape factor because, at normal carrier and audio frequencies,
the upper and lower sidebands are quite close in frequency.
Another consideration is that the sideband filter should offer significant attenuation to the
carrier, so that the balanced modulator need not be so accurately balanced. In practice
the balanced modulator might provide 30 db of carrier suppression and the filter a further
10db. The other sideband would normally be about 30 to 40 db down on the wanted one.
In order to achieve this, the SSB filter has several poles and is, in most cases a ceramic
filter or crystal filter. Various filters are commercially available with different
specifications depending on the application.
In the practical we use a high modulating frequency so you can see clearly the
relationship between the various frequency components. This means that the filter
specification can be relaxed and here a single tuned circuit is used. Separate filters are
provided for upper and lower sidebands and the means is provided to monitor the output
of both.
You might be surprised that the output from the SSB filters is simply a sinusoidal signal
but, since we use sinusoidal carrier and modulating frequencies, the sum or difference of
the two must be a single frequency.
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When the modulation is a band of frequencies, the SSB output will also be a band
of frequencies.
2.2.6.2 Upper or Lower Sideband?
An obvious question is which sideband should be transmitted? The answer owes more to
convention than theory!
There is no reason why one sideband gives better results than the other, but general
practice seems to favour the upper sideband.
One convention is that with carrier frequencies below 10 MHz the lower sideband should
be used, but this is not always the case. The result of this is that many pieces of
communication equipment have to be able to deal with both.
To begin the practical, please turn to the next page.
2.2.6.3 Procedure
Use the spectrum analyser and oscilloscope to observe at monitor point 6. Note that the
signal is DSB. Adjust the carrier balance as before.
Monitor at monitor point 8, and at monitor point 9, and note that only one sideband is
present. Note that the carrier suppression is less dependent on the carrier balance than
before the filter.
Use the oscilloscope to observe that the SSB output is a sinusoidal signal. Use the
spectrum analyser to note that the upper sideband frequency is the sum of the carrier and
modulation frequencies and the lower sideband is the difference.
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2.2.6.4 Questions
1. Why is the balance of the modulator less important in a filter method SSB generator
than for a DSB generator?
2. How is the width of the SSB filter related to the maximum and minimum modulating
frequencies?
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2.2.7 Practical 3: Demodulation of SSB
This practical is about the demodulation of SSB.
In the double sideband suppressed carrier practical we saw how DSB is demodulated
using the BFO to reinsert the carrier. In the case of DSB the BFO must be in phase with
the original carrier or the process will not work correctly.
Since SSB is transmitted without a carrier it is not surprising that a similar method is
employed.
The main difference is that, for SSB, the BFO need not be in phase with the carrier. It
does need to be at the same frequency but even a small error in the frequency results
only in a small error in the frequency of the demodulated output.
This means that in noncritical applications, such as speech, a small overall frequency
error does not make the system useless. The effect on speech is to raise or lower the
tone of the voice, which within limits does not reduce intelligibility.
The fact that the BFO need not be locked, greatly simplifies the design of the receiver,
and makes SSB one of the most powerful techniques for transmitting audio frequencies
over radio links with its narrow bandwidth and efficient use of available transmitter
power.
In the practical you can use both upper and lower sidebands and see that with the BFO
set correctly, near to the original carrier frequency, even though the two sidebands are at
different frequencies the demodulated output is the same. You can also see that changing
the BFO frequency causes the demodulated output to change in frequency by a similar
amount.
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2.2.7.1 Procedure
Monitor at monitor point 6, and observe the DSB signal. Move to monitor point 10 and
note the upper sideband signal.
Use the spectrum analyser to confirm that the frequency is that of the upper sideband.
Change to lower sideband (by pressing the button) and repeat.
Now monitor at monitor point 14 and compare the output with the modulation input. Use
either the oscilloscope or analyser to set the BFO frequency to that of the carrier, by
monitoring at monitor point 13.
Note that both sidebands give the same output frequency.
Move the BFO frequency and observe the effect on the output using first one and then
the other sideband.
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2.2.7.2 Questions
1. Why is SSB more efficient than either simple AM or DSB?
2. If the BFO frequency rises, what happens to the frequency: a) of the upper sideband?
b) of the lower sideband?
3. Calculate the bandwidth of the transmitted signal when the modulation frequency band
extends from 500 Hz to 50 kHz for simple AM, DSB and SSB.
4. If a SSB channel has no modulating signal, what is the modulated signal like?
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Notes
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3 Assignments using the Frequency Modulation Workboard
3.1 Generation of Frequency Modulation Assignment
3.1.1 Objectives
On completion of this assignment you will be familiar with:
Frequency modulators,
Modulation index,
Bandwidth,
FM signals in the time domain,
FM signals in the frequency domain.
3.1.2 Practicals
Practical exercises are provided as follows:
Practical 1: Concepts of Modulation
Practical 2: Generation of FM with a VCO
Practical 3: Spectrum of an FM signal with a large Modulation Index
3.1.3 Workboard Required
Frequency Modulation Workboard 53140 which comprises the following blocks:
Signal Generation,
Modulator,
Limiter,
Quadrative Demodulator,
VCO,
Phase Comparator.
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3.1.4 Theory
3.1.4.1 Modulation
The equation of a sinusoidal voltage waveform is given by:
v = V
max
.sin(t+)
where:
v is the instantaneous voltage,
V
max
is the maximum voltage amplitude,
is the angular frequency,
is the phase.
A steady voltage corresponding to the above equation conveys little information.
To convey information the waveform must be made to vary so that the variations
represent the information. This process is called modulation.
Any of these may be varied to convey information.
3.1.4.2 Frequency Modulation
Frequency modulation uses variations in frequency to convey information.
We shall think in terms of the angular frequency . The wave whose frequency is being
varied is called the carrier wave. The signal doing the variation is called the modulating
signal.
For simplicity, suppose both carrier wave and modulating signal are sinusoidal; ie:.
v
c
= V
c
sin
c
t
(c denotes carrier) and
v
m
= V
m
cos
m
t
(m denotes modulation)
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3.1.4.3 What is Frequency?
If the frequency is varying, how do we define it?
We can no longer count the number of cycles over a longish interval to count the cycles
per second. Instead we define frequency as the rate of change of phase.
This is consistent with the simple definition, because at a constant (angular) frequency
radians/second the phase is changing at radians per second, which is /2 cycles per
second.
Since we can only define what the instantaneous frequency is by reference to the phase,
we must look at the phase in order to arrive at an expression for the frequencymodulated
signal.
3.1.4.4 Phase of the FM Signal
For the unmodulated carrier v
c
= V
c
sin
c
t, the phase is:
s =
c
t
We want the modulating signal to vary the carrier frequency,
c
, so that its frequency
takes the form:
=
c
+ D cos
m
t
(where D denotes the peak value of the deviation)
It is related to the amplitude of the modulating signal v
m
by the 'frequency slope' of the
frequency modulator (VCO) say k radians/s per V. The peak value of v
m
produces
deviation D, so:
D = k V
m
The total phase change undergone at time t is found by integrating the angular frequency.
It is
s = !(
c
+ D cos
m
t) dt
=
c
t + (D/
m
) sin
m
t.
(If you are not familiar with integration you will have to take this result on trust).
So the FM signal can be expressed as:
V
c
sin [
c
t + (D/
m
) sin
m
t]
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3.1.4.5 Modulation Index
In the expression for the FM signal:
V
c
sin [
c
t + (D/
m
) sin
m
t]
the coefficient (D/
m
) turns out to be quite important and is given the name modulation
index.
It is often represented by the Greek letter beta, .
So we may write the FM signal as:
v
c
= V
c
sin (
c
t + sin
m
) t
where is the modulation index (D/
m
).
In this expression, the factor sin (
c
t + sin
m
) t (let us call it F) is of the form sin (a + b)
which can be expanded to sin a cos b + cos a sin b.
Applying this expansion to F, we get:
F = sin
c
t cos(sin
m
) t + cos
c
t sin (sin
m
) t
3.1.4.6 FM Sidebands
These complicated functions can be expanded, using mathematics too elaborate to
explain here, into a series of terms like this:
F = J
0
() sin
c
t+ J
1
() [ sin (
c
+
m
)t  sin (
c

m
)t ]
+ J
2
() [ sin (
c
+ 2
m
)t  sin (
c
 2
m
)t ]
+ J
3
() [ sin (
c
+ 3
m
)t  sin (
c
 3
m
)t ]
+ J
4
() [ sin (
c
+ 4
m
)t  sin (
c
 4
m
)t ]
+ ...
where J
0
(), J
1
(), J
2
() etc are constants whose values depend only on . They are
called Bessel Functions.
There is an infinite series of these functions, and so an infinite number of FM
sidebands. But in practice the values of the Bessel functions become very small as the
series goes on. For example, when = 2
J
0
(2) = 0.224
J
1
(2) = 0.577
J
2
(2) = 0.353
J
3
(2) = 0.129
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J
4
(2) = 0.034
J
5
(2) = 0.007
3.1.4.7 A Practical Approximate Rule
Because the higherorder sidebands become very small, in practice the bandwidth of the
FM signal may be restricted to a finite bandwidth.
The practical rule that is used, often called Carsons Rule, is to take the bandwidth
required as:
B = 2 ( F
d
+ F
m
)
where B is the bandwidth, F
d
the deviation and F
m
is the bandwidth of the modulation, all
in the same units.
3.1.5 Practical 1: Concepts Of Frequency Modulation
This practical introduces the idea of frequency modulation. Before you start it is necessary
to appreciate some fundamental concepts.
As in amplitude modulation, a carrier frequency is modulated by the information that is
being sent. In AM it is the amplitude of the carrier that is varied in time with the
modulation, in FM it is the frequency that is varied. The amplitude is constant as we will
see.
When no modulation is being applied the carrier is at its nominal frequency i.e the carrier
frequency. The modulating signal causes the frequency to deviate, i.e. to move above
and below its nominal value. With the greatest possible deviation, the minimum frequency
could be near zero and, assuming the modulating signal to have no d.c component, the
maximum frequency would then be about twice the carrier frequency.
However, this would take a very large amount of frequency spectrum and the bandwidth
would have no relationship to the modulating signal bandwidth. A set limit is normally
made on the amount that the carrier can deviate from its nominal frequency and this is
called the maximum deviation.
Different systems use different values of maximum deviation, depending on a number of
factors some of which are very complex
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3.1.5.1 Bandwidth of an FM Signal
It is important that we can understand and estimate the bandwidth of the transmitted
signal so that the transmission parameters can be chosen to fit into the available
spectrum.
Clearly the bandwidth must be at least equal to twice the deviation, as the carrier actually
moves above and below its nominal frequency by that amount. But it also depends on
how fast the frequency is being changed; ie, on the bandwidth of the modulating signal.
The mathematical analysis of FM is quite involved and it shows that an FM signal has
sidebands far above and below the maximum deviation.
However the power in these sidebands decreases quickly as they become further away
from the carrier and it can be shown that, for practical purposes, a good approximation to
the bandwidth is given by:
B = 2 ( F
d
+ F
m
)
(where B is the bandwidth, F
d
the deviation and F
m
is the bandwidth of the modulation.)
This is sometimes called Carsons Rule, and the bandwidth B can be viewed as
containing the majority of the transmitted power, certainly sufficient for successful
demodulation.
3.1.5.2 Modulation Index
As we have seen, the bandwidth of an FM signal depends on both the deviation and the
modulation bandwidth.
It might be thought that, in order to keep the bandwidth as narrow as possible, all FM
systems should be operated with a very small deviation.
However, there are significant advantages to operating with a wide deviation. The main
one is an apparent improvement in noise performance.
So as you can see, a specific bandwidth can be the result of wide deviation with a low
modulation bandwidth or a narrow deviation with a large modulation bandwidth.
The ratio of deviation to modulation bandwidth is called the modulation index and is an
important parameter in describing a FM system.
Modulation index is given by:
MI = F
d
/ F
m
where MI is the modulation index.
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Modulation index is sometimes represented by the Greek letter beta ().
In the practical a frequency modulator is formed from a voltage controlled oscillator. A
voltage is applied to it from a control on the hardware board and the oscillator output can
be examined on the oscilloscope and spectrum analyser.
Using this configuration the fundamental concept of an oscillator frequency being
changed by an external signal can be understood.
3.1.5.3 Procedure
In this practical the hardware is configured as shown.
You have available an oscilloscope and a spectrum analyser. Using this configuration you
can see how the oscillator frequency can be controlled by an external signal.
Set Carrier level to about half scale.
Use the oscilloscope to observe that when the manual frequency control is moved the
frequency changes. Monitor at monitor point 16 to see the voltage applied to the oscillator
and monitor point 4 to see the output.
Monitoring monitor point 4 use the large oscilloscope calibration to measure the total
frequency range of the oscillator.
Use the spectrum analyser to confirm the frequency range measured on the oscilloscope.
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3.1.5.4 Questions
1. Is it easier to measure the frequency range on the oscilloscope or on the spectrum
analyser?
2. Choose two voltages levels at the control input to the oscillator and measure the
corresponding output frequencies. Hence calculate the 'frequency slope' of the
oscillator in kilohertz per volt.
3. Can you see any amplitude variation over the frequency range? Should there be any?
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3.1.6 Practical 2: Generation of FM with a VCO
In this practical a sine wave signal is used to frequencymodulate a carrier so that you can
investigate the appearance of such signals in both the time and frequency domains.
You can adjust the amount of deviation and hence change the modulation index. Notice
that the appearance of an FM signal on the spectrum analyser is similar to that of an AM
signal when the modulation index is small.
Try to reconcile the explanation of the bandwidth of an FM signal given in the previous
background pages with the observations you make in this practical.
Move to the next page to start the practical.
3.1.6.1 Procedure
In this practical the variable voltage used to control the VCO frequency has been replaced
by a sine wave oscillator. This sine wave now frequencymodulates the carrier.
Set Carrier level to about half scale. Look at the signal at monitor point 4 with the
oscilloscope.
Turn the modulation level up and down and observe the effect.
Notice that the frequency is changing. Note where the output at monitor point 4 has a
higher frequency. Change to monitor point 3 and observe how the instantaneous
frequency depends on the instantaneous value of the modulating signal.
Use the spectrum analyser to examine the sidebands of the signal. Adjust the
modulation level and observe that, at low deviation, only the sidebands at F
c
 F
m
and F
c
+ F
m
are present.
At higher deviation, ie, a larger modulation index, higherorder sidebands appear.
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3.1.6.2 Questions
1. By looking at the spectrum of the modulated signal, can you estimate the frequency of
the modulating signal? (Explain carefully how).
2. Would it be equally easy to estimate the bandwidth of the modulating signal from the
spectrum if the modulating signal were complex, having many frequencies?
3. As the modulation level varies, how constant are: (a) the carrierfrequency component
of the modulated signal? (b) the amplitude of the modulated signal?
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3.1.7 Practical 3: Spectrum of an FM signal with a large Modulation Index
This is a simple practical where the frequency modulator is connected to the spectrum
analyser. The carrier frequency has been reduced to about 5 kHz so, since the maximum
deviation is the same as in the other practicals of this Assignment, the modulation index is
much greater.
As we saw, the bandwidth is:
B = 2 ( F
d
+F
m
)
where B is the bandwidth, F
d
the deviation and F
m
is the bandwidth of the modulation.
So if F
m
is small compared with F
d
, ie, the modulation index is large, then:
B = 2 F
d
On the analyser the spectrum appears to be continuous but in reality it is made up of a
large number of sidebands spaced at 5 kHz intervals from the carrier up to F
d
.
This practical simply shows how when the modulation index is large the bandwidth is
determined almost exclusively by the deviation.
3.1.7.1 Procedure
In this practical the modulation frequency has been set to 5 kHz. This means that the
modulation index can be very high. This enables you to see that under these conditions
the bandwidth of an FM signal is almost equal to twice the deviation.
Set Carrier level to about half scale. Turn the 5 kHz level up and down and observe the
bandwidth changing. Note that the bandwidth is almost proportional to the deviation.
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3.1.7.2 Questions
1. If the modulating frequency is 5 kHz and the deviation is 50 kHz, calculate the
modulation index.
2. Calculate the signal bandwidth using Carson's rule.
3. If a bandpass filter were to be added at the input of an FM detector what factors
determine the bandwidth required?
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3.2 Demodulation of Frequency Modulated Signals Assignment
3.2.1 Objectives
On completion of this assignment you will be familiar with:
Quadrature detectors,
PLL detectors,
Noise.
3.2.2 Practicals
Practical exercises are provided as follows:
Practical 1: A Quadrature Detector with Limiter
Practical 2: The Effect of Noise on a Quadrature Detector
Practical 3: PLL Detector with a Limiter
Practical 4: The Effect of Noise on a PLL Detector
3.2.3 Workboard Required
Frequency Modulation Workboard 53140 which comprises the following blocks:
Signal Generation,
Modulator,
Limiter,
Quadrative Demodulator,
VCO,
Phase Comparator.
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3.2.4 Theory
3.2.4.1 Quadrature Detector
The quadrature detector splits the incoming FM signal into two paths. One path is
connected directly to one input of a phase detector. The other path contains a simple
network which shifts the phase of the signal in proportion to its frequency deviation.
Let us consider the nature of a typical phase shifter ...
Regard this as a simple potential divider, with input at point 1, output at point 2. The upper
arm has impedance
jL + R
and the lower arm
1/(jC)
where w is the angular frequency.
The circuit's transmission factor is
e
2
/e
1
= [1/jwC] / [R + jL + (1/jwC)]
e
2
/e
1
= 1 / [ 1 
2
LC + j CR ]
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It is convenient to express the transmission factor in terms of
o
, the resonant frequency,
at which:
o
L = 1/(
o
C) , or
o
2
LC = 1 ,
and the quality factor Q, given by:
Q =
o
L/R = 1/(
o
CR)
Using these definitions, we can substitute 1/
o
2
for LC and 1/Q
o
for CR. So the
transmission factor:
e
2
/e
1
= 1 / [ 1 
2
LC + jCR ]
becomes:
e
2
/e
1
= 1 / [ 1 
2
/
o
2
+ j /
o
Q ]
The phase of this expression is:
=  arctan [/
o
Q ] / [1  (/
o
)
2
]
which, if we define y = /
o
can be written as:
= arctan [ y / Q (y
2
 1) ]
= arctan [ Q (y
2
 1) / y ]
= (/2)  arctan [ Q (y
2
 1) / y ]
= (/2) + arctan Q [ y  (1/y)]
Replacing y by /
o
once more, this can be written as:
= (/2) + arctan [Q (
2

o
2
)/
o
]
= (/2) + arctan [Q ( 
o
) ( +
o
)/
o
]
= (/2) + arctan [Q d ( 2
o
+ d) ]/[ (
o
+ d)
o
]
where d is defined as 
o
.
Neglecting d in comparison with
o
in:
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= (/2) + arctan [Q d ( 2
o
+ d) ]/[ (
o
+ d)
o
]
we get:
= (/2) + arctan [Q d ( 2
o
) ]/[(
o
)
o
]
= (/2) + arctan [ 2 Q d /
o
]
If dw is sufficiently small, the argument of the arctan is small, and therefore close to the
arctan value in radians. So, still to a good approximation:
= ( /2) + [ 2 Q d /
o
]
It must be noted however, that if Q has a high value, these approximations may become
invalid, for quite moderate values of deviation (d).
This indicates that for low distortion, the value of Q must be kept low. For good sensitivity,
Q should be high. So here is an instance where a degree of compromise in the design is
needed.
Summing up, if the carrier is at its nominal frequency (d = 0), the phase is just
( /2). The change in phase from ( /2) is proportional to the deviation (d ), provided
that 2 Q d /
o
is small.
3.2.5 Practical 1: Quadrature Detector
The purpose of an FM demodulator is to return the modulating signal to baseband. Since,
in FM, the instantaneous frequency is proportional to the modulating signal, all that is
needed is a circuit block which produces a voltage proportional to the input frequency.
This is not quite as simple as it sounds.
A very crude way of achieving this is to feed the signal through a filter with its cutoff near
to the carrier frequency. The signal is then attenuated by an amount depending on its
frequency. The filter output is now AM and can be demodulated by any AM detector. This
type of detector uses the slope of the filter characteristic and therefore is called a Slope
Detector.
Slope detectors are not satisfactory in most applications as they are not very linear and
do not reject noise well.
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3.2.5.1 Quadrature Detectors
Another way is to turn the FM to phase modulation and then use a phase detector. This is
a very common technique and is the basis of the Quadrature Detector.
In this practical you will see how a quadrature detector works. The incoming FM signal is
split into two paths. One path is connected directly to one input of a phase detector. The
other path contains a simple network which shifts the phase of the signal in proportion to
its frequency deviation.
The output of the phase shift network is connected to the other input of the phase
detector. As the frequency varies, both detector inputs vary together in frequency, but one
also shifts in phase relative to the other. It is this varying phase shift that produces the
output from the detector.
A minor complication is that most phase detectors produce their mean output for 90
degrees phase difference between the input signals. This is the required condition when
the FM signal is at its centre frequency, so an additional constant 90 degree phase shift is
added to one of the paths. When unmodulated, the two inputs to the phase detector are
at 90 degrees apart, or in quadrature; hence the name of the detector.
This constant phase shift is usually added by means of a simple inductor. The output of
the phase detector still contains a large component at twice the carrier frequency and the
detector is usually followed by a filter that passes the baseband but not the carrier.
Quadrature detectors are used extensively in domestic FM radios and in a lot of
communications equipment.
In this practical, the same modulator is used as in the Generation of Frequency
Modulation assignment. The modulation is a sine wave so that the signal can be followed
through the circuit.
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3.2.5.2 Procedure
This practical shows a quadrature detector working. Monitor at 9 and observe the FM
signal at different settings of modulation level. Note the two signals at the inputs of the
phase detector 9 and 11 . Set the modulation level to about half scale.
Observe the signal at the phase detector output 12 and then after the postdetection filter
at 14.
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3.2.5.3 Questions
1. Use the large oscilloscope to try and measure the phase shift between the two phase
detector inputs when there is no modulation.
2. What frequencies must the output filter:
(a) pass?
(b) reject?
3. Would your answer to question 2(b) be altered if the phase comparator were imperfect
in some way?
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3.2.6 Practical 2: Phase lock loop detector (PLL)
This practical introduces the phase locked loop (or PLL) demodulator. This type of
detector offers some advantages over the quadrature detector when the signal to noise
ratio is poor.
Before trying to understand how a PLL can demodulate an FM signal it is necessary to
understand what a PLL is. The concept is of an oscillator synchronised in phase to an
external signal source using a feedback loop.
As frequency is the same thing as rate of change of phase, once the phase of the local
oscillator is synchronised to the external signal, the frequencies are automatically made
identical.
A phase locked loop consists of three main blocks:
1. An oscillator, the frequency of which is controlled by an external voltage or current
source. A voltage controlled oscillator or VCO is used in this assignment.
2. A phase detector, which compares the phase of the oscillator with that of the
external signal.
3. A filter, which smoothes the output from the detector to provide the control signal
to the VCO, adjusting its frequency so as to reduce the phase difference.
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3.2.6.1 Operation of a PLL
Imagine an incoming signal at a constant frequency within the range of the VCO.
Its phase is compared with that of the VCO and a voltage produced that alters the VCO
frequency. The phase of the VCO therefore starts changing relative to the incoming
signal, until eventually the phases match. Once they are equal, the control signal goes to
zero and the system settles into equilibrium. Any drift of the VCO will be corrected by the
control voltage which again appears. The two signals are said to be phase locked.
A filter is used in the control loop to keep the system stable and limit the maximum rate of
change of oscillator frequency.
This whole description is a very simplified view, and the parameters that set the filter
characteristics are very complex. An important factor in the design is the time before the
two signals become locked.
Phase locked loops are used extensively in communications systems where it is
necessary to produce a reference oscillator in phase with an incoming signal; also in
special signal sources called frequency synthesizers.
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3.2.6.2 The PLL as an FM Detector
Now that you have appreciated the concept of a PLL, how can it be applied to demodulate
FM?
Suppose that there was a PLL locked onto an incoming carrier which was unmodulated.
The VCO would be at the same frequency as the carrier and the VCO control voltage
would be constant.
If the carrier were to change in frequency the VCO would follow the change by means of a
change in control voltage. So the VCO control voltage varies with the carrier
frequency, and if the carrier were frequency modulated the modulation would appear
superimposed on the VCO control voltage.
When a postdetection filter is added to the simple PLL to remove all the frequency
components above the maximum modulating frequency we now have a PLL FM detector.
In this practical you will see a PLL detector demodulating the same FM signal as before.
The PLL is used when the ability to demodulate in the presence of noise is important. The
distortion produced by this type of detector is determined mainly by the linearity of the
VCO but this is often less important in noisy applications.
FM detection under noisy conditions is investigated in the next assignment.
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3.2.6.3 Procedure
This practical shows a phase lock loop detector working. Monitor at 9 and observe the FM
signal at different settings of modulation level. Examine the two signals at the input of
the phase detector at 9 and the tracking VCO at 11 . Set carrier level to maximum.
Observe the signal at the phase detector output 12 and then after the post detection filter
at 14.
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3.2.6.4 Questions
1. What happens to the demodulated output when you reduce the control from maximum
to half amplitude? Explain your answer in a sentence or two.
2. What happens to the demodulated output when you reduce the control to lower
levels?
3. What is the special problem which occurs for very low signal levels?
4. Why do you think it happens and what do you think could be done about it?
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3.3 Limiters and the Effect of Noise on FM Demodulation Assignment
3.3.1 Objectives
On completion of this assignment you will be familiar with:
Limiters,
Predetection noise,
Postdetection noise.
3.3.2 Practicals
Practical exercises are provided as follows:
Practical 1: A Quadrature Detector with Limiter
Practical 2: The Effect of Noise on a Quadrature Detector
Practical 3: PLL Detector with a Limiter
Practical 4: The Effect of Noise on a PLL Detector
3.3.3 Workboard Required
Frequency Modulation Workboard 53140 which comprises the following blocks:
Signal Generation,
Modulator,
Limiter,
Quadrative Demodulator,
VCO,
Phase Comparator.
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3.3.4 Theory
3.3.4.1 Frequency Modulation
The equation of a sinusoidal voltage waveform is given by:
v = V
max
.sin( t+)
where v is the instantaneous voltage
V
max
is the maximum voltage amplitude
is the angular frequency
is the phase
A steady voltage corresponding to the above equation conveys little information. To
convey information the waveform must be made to vary so that the variations represent
the information. This process is called modulation.
From the above equation, the basic parameters of such a waveform are:
its amplitude, V
max
its frequency, (or f)
its phase,
Any of these may be varied to convey information.
Frequency modulation uses variations in frequency to convey information. We shall
think in terms of the angular frequency w. The wave whose frequency is being varied is
called the carrier wave. The signal doing the variation is called the modulating signal.
For simplicity, suppose both carrier wave and modulating signal are sinusoidal; ie,
v
c
= V
c
sin
c
t
(c denotes carrier) and
v
m
= V
m
cos
m
t
(m denotes modulation)
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3.3.4.2 What is Frequency?
If the frequency is varying, how do we define it? We can no longer count the number of
cycles over a longish interval to count the cycles per second. Instead we define
frequency as the rate of change of phase. This is consistent with the simple definition,
because at a constant (angular) frequency w radians/second the phase is changing at
radians per second, which is /2 cycles per second.
Since we can only define what the instantaneous frequency is by reference to the phase,
we must look at the phase in order to arrive at an expression for the frequencymodulated
signal.
3.3.4.3 Phase of the FM Signal
For the unmodulated carrier v
c
= V
c
sin
c
t, the phase is
s =
c
t
We want the modulating signal to vary the carrier frequency,
c
, so that its frequency
takes the form
=
c
+ D cos
m
t
(where D denotes the peak value of the deviation)
It is related to the amplitude of the modulating signal v
m
by the 'frequency slope' of the
frequency modulator (VCO) say k radians/s per V. The peak value of v
m
produces
deviation D, so
D = k V
m
The total phase change undergone at time t is found by integrating the angular frequency.
It is
s = (
c
+ D cos
m
t) dt
=
c
t + (D/
m
) sin
m
t.
(If you are not familiar with integration you will have to take this result on trust).
So the FM signal can be expressed as:
V
c
sin [
c
t + (D/
m
) sin
m
t] ,
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3.3.4.4 The Modulation Index
In that expression for the FM signal:
V
c
sin [
c
t + (D/
m
) sin
m
t]
the coefficient (D/
m
) turns out to be quite important and is given the name 'modulation
index'. It is often represented by the Greek letter beta, .
So we may write the FM signal as:
v
c
= V
c
sin (
c
t + sin
m
) t
where is the modulation index (D/
m
)
In this expression, the factor sin (
c
t + sin
m
) t (let us call it F) is of the form sin (a + b)
which can be expanded to sin a cos b + cos a sin b.
Applying this expansion to F, we get
F = sin
c
t cos( sin
m
) t + cos
c
t sin ( sin
m
) t
3.3.4.5 FM Sidebands
These complicated functions can be expanded, using mathematics too elaborate to
explain here, into a series of terms like this:
F = J
0
( ) sin
c
t+ J
1
( ) [ sin (
c
+
m
)t  sin (
c

m
)t ]
+ J
2
( ) [ sin (
c
+ 2
m
)t  sin (
c
 2
m
)t ]
+ J
3
( ) [ sin (
c
+ 3
m
)t  sin (
c
 3
m
)t ]
+ J
4
( ) [ sin (
c
+ 4
m
)t  sin (
c
 4
m
)t ]
+ ...
where J
0
( ), J
1
( ), J
2
( ) etc are constants whose values depend only on . They are called
Bessel Functions.
There is an infinite series of these functions, and so an infinite number of FM
sidebands. But in practice the values of the Bessel functions become very small as the
series goes on. For example, when = 2
J
0
(2) = 0.224
J
1
(2) = 0.577
J
2
(2) = 0.353
J
3
(2) = 0.129
J
4
(2) = 0.034
J
5
(2) = 0.007
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3.3.4.6 A Practical Approximation Rule
Because the higherorder sidebands become very small, in practice the bandwidth of
the FM signal may be restricted to a finite bandwidth.
The practical rule that is used, often called Carson's Rule, is to take the bandwidth
required as:
B = 2 ( F
d
+ F
m
)
where B is the bandwidth, F
d
the deviation and F
m
is the bandwidth of the modulation, all
in the same units.
3.3.5 Practical 1: A Quadrature Detector with Limiter
In this practical you will see how a limiter works.
In an FM system the information is carried by variations in carrier frequency. Since the
variations in amplitude carry no information, they can be removed before the signal
arrives at the detector. This is the function performed by a limiter.
A limiter is simply a high gain amplifier that turns the usually sinewave carrier of varying
amplitude into a square wave of constant amplitude. The square wave still contains the
frequency variations that contain the modulation.
The addition of a limiter means that the FM detector has a constant amplitude
signal to deal with which means that its output is only dependent on phase
changes and not changes in amplitude.
This can be shown in the practical by varying the carrier amplitude with no limiter in
operation and finding that the output signal also varies in amplitude. When the limiter is
placed in circuit the output no longer varies in this way.
Of course, the limiter cannot produce a signal from nothing so as the input carrier
amplitude to the limiter falls the noise content increases. Ultimately the signal becomes
unrecognisable because of noise.
The effect of noise on the detector performance is investigated in more detail in the next
Practical.
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3.3.5.1 Procedure
This practical shows the effect of a limiter on a quadrature detector. Start with the limiter
out of circuit. Set modulation level to maximum. Observe the signal at the detector
output 14 while varying the carrier level.
Use the Limiter Button to switch the limiter into circuit. Repeat your observations.
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3.3.5.2 Questions
1. In the absence of a limiter, does the demodulated output depend on the carrier
amplitude?
2. Should it, ideally?
3. With the limiter in use, how does the demodulated output vary with carrier amplitude?
4. How does the waveform of the input to the phase comparator (point 9) differ with and
without the limiter?
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3.3.6 Practical 2: The Effect of Noise on a Quadrature Detector
In the first Practical of this Assignment we saw how a limiter forces the input to the
detector to be of constant amplitude. This has an important effect on how the detector
behaves when the signal is noisy.
3.3.6.1 What is Noise?
Noise is simply an unwanted signal which is mixed up with the required signal. In many
cases it is not a specific frequency but is made up of random combinations of many
frequencies.
Such unwanted noise may be generated internally by circuit elements like amplifiers or
come from the transmission medium such as cables or antennas. A very important
characteristic of a communication system is how well it works in the presence of
noise.
FM systems offer some advantage over AM systems in their noise performance. The
theory behind this is quite complex and will not be dealt with here.
3.3.6.2 Signal/Noise Ratios
One measure of the quality of the received signal applied to the detector is its Signal to
Noise Ratio (SNR). This is simply the ratio of signal power to noise power, usually
expressed in decibels for convenience. A high SNR means that there is much more signal
than noise.
After passing through the detector the demodulated output also has noise on it, and
therefore has a signal to noise ratio. These two ratios are often called predetection SNR
(SNR
i
) and post detection SNR (SNR
o
).
In an AM system these two values are approximately equal, but in an FM system the
SNR
o
can be greater than SNR
i
. How much greater depends on many things, but
especially on the modulation index.
Another contrast between AM and FM is that in an AM channel, SNR
i
is proportional to
SNR
o
but in FM it is not; instead, as SNR
i
is reduced below a certain level, called the
threshold, SNR
o
drops very quickly.
This means that FM systems tend to degenerate very quickly at low SNR. In general, the
greater the modulation index (and therefore the bandwidth) the greater the improvement
in SNR given by the detector. However, this is at the expense of a higher threshold.
As is often the case, selecting the parameters to get the best performance out of the
system is a compromise.
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An important point to note is that when noise is present it causes both amplitude and
frequency variations, and to obtain the best performance on an FM system the amplitude
noise should be removed with a limiter.
In the practical, the FM generator is connected to a quadrature detector. A variable
amount of noise is added to the signal fed to the detector generated by a special circuit
that simulates the type of wideband noise that would come from a telecommunications
system.
A limiter is also provided, which can be switched in and out of use. The purpose of the
practical is to investigate the effect of noise on the demodulated output, and how this is
affected by the limiter.
3.3.6.3 Procedure
This practical shows a quadrature detector demodulating a noisy FM signal. Adjust
Carrier level and Modulation level to examine each signal, leaving them both at
maximum. Set the Noise level to minimum. The limiter should not be in use. Observe at
14 with the oscilloscope and note that the demodulator is working correctly. Increase the
Noise level and observe that the output becomes noisy.
Now switch in the limiter with the Limiter Button and repeat the test. Note that the
detector keeps working at lower SNR when the limiter is in use. Use the other monitoring
points to see how the system is operating.
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3.3.6.4 Questions
1. What does the term SNR stand for and what does it mean?
2. Why do you think that it is more convenient to express it in decibels?
3. Why does the use of the limiter reduce the noise on the output?
4. With the limiter in circuit, observe that reducing the carrier level causes an increase in
the noise on the output. Why is this? (Hint  a look at the limiter output may help).
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3.3.7 Practical 3: PLL Detector with a Limiter
In this practical you will see first how a Phase Locked Loop behaves at different signal
amplitudes. Introducing a limiter will then show an improvement in performance.
If the received signal is large enough, the PLL will lock the local oscillator phase to that of
the received signal. Doubling the signal amplitude will not alter this situation, so will not
affect the output signal. To this extent the PLL removes unwanted amplitude modulation
of the received signal.
If the signal is small and the deviation is large, the phase detector cannot give enough
output to move the VCO over a large enough range to track the deviation. This can be
shown in the practical by reducing the carrier amplitude with no limiter in operation.
Failure to track over the whole range of deviation shows as a distortion of the output
signal. For small enough signals, the PLL fails to lock altogether.
The addition of a limiter means that the phase detector in the PLL has a constant
amplitude signal to deal with. The gain of the phase lock control loop is therefore
maintained as the signal level changes.
3.3.7.1 Procedure
This practical shows the effect of a limiter on the performance of a PLL detector at various
carrier levels. Start with the limiter out (this is the default). Set carrier level and 14 with
the oscilloscope.
Reduce the carrier level and observe the output. Note that the detector loses track of the
signal below a certain level, causing distortion of the detector output. Note also that, when
the modulation level is reduced, the carrier can be reduced further without distortion.
Now use the Limiter Button to switch in the limiter and repeat the tests. Note that the
detector continues to work at much lower carrier levels. Use the other monitoring points to
see how the system is operating.
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3.3.7.2 Questions
1. With the limiter switched out, why does a lower modulation level allow the carrier level
to be reduced further before distortion occurs?
2. Does this also happen when the limiter is switched in?
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3.3.8 Practical 4: The Effect of Noise on a PLL Detector
Noise on the received signal causes both amplitude and phase changes. When a limiter
is placed in the circuit, the amplitude changes are removed from the PLL input.
The principal effect on the PLL is that as the input signal tends to zero amplitude, there
remains an adequate amplitude of signal to drive the phase lock loop. This continues to
track the phase of the noisy received signal effectively, and with minimum error caused by
noise amplitude variations.
Of course, the limiter cannot produce a signal from nothing, so as the carrier amplitude
into the limiter falls, the noise from the limiter increases. This noise is faithfully detected
by the PLL and degrades the output signal. This problem is due to fundamental noise
problems and not due to any failing of the detector itself.
3.3.8.1 Procedure
This practical has a PLL detector demodulating a noisy FM signal. Set the Carrier level
and Modulation level to maximum. Set the Noise level to minimum. The limiter should
not be in use. Observe at 14 that the demodulator is working correctly. Increase the
Noise level and observe that the output becomes noisy. Decrease the signal/noise ratio
(SNR) further by reducing the carrier level until the signal becomes unrecognisable.
Now switch in the limiter using the Limiter Button . Note that the detector keeps working
at lower SNR when the limiter is in use. Use the other monitoring points to see how the
system is operating.
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3.3.8.2 Questions
1. How does limiting the signal improve the signal to noise ratio?
2. How does the performance of the PLL demodulator compare with the phase
comparator demodulator?
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4 Assignments using the Signal Sources Workboard
4.1 Wien Bridge Oscillator Assignment
4.1.1 Objectives
On completion of this assignment you will be familiar with:
Conditions for oscillation of a Wien Bridge Oscillator,
The need for amplitude control of a Wien Bridge Oscillator.
4.1.2 Practicals
Practical exercises are provided as follows:
Practical 1: Basic Wien Bridge Oscillator
Practical 2: Amplitude Stabilisation
Practical 3: Changes from Standard
4.1.3 Workboard Required
Signal Sources Workboard 53110 which comprises the following blocks:
Wein Bridge Oscillator,
LC Pass Filter,
Crystal Oscillator,
Multivibrator.
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4.1.4 Theory
4.1.4.1 Wien Bridge Oscillator
The theory of the Wien Bridge oscillator is usually presented for equal values of R and
C in the series and parallel arms. Here we look at the more general case of arbitrary R
and C values.
Here Z
1
is the impedance of the combination of C
1
and R
1
and Z
2
is the impedance of the
combination of C
2
and R
2
.
It is easier to consider the reciprocal 1/B of the feedback fraction, since this is
(Z
1
+ Z
2
)/Z
2
, which simplifies to:
1/B = 1 + Z
1
/ Z
2
where Z
1
= R
1
+ 1/jC
1
and Z
2
= R
2
/ (1+jC
2
R
2
)
Routine manipulation produces the result:
1/B = [1 
2
C
1
C
2
R
1
R
2
+ j (C
1
R
1
+ C
2
R
2
+ C
1
R
2
)] / jC
1
R
2
1/B = [1 
2
C
1
C
2
R
1
R
2
+ j (C
1
R
1
+ C
2
R
2
+ C
1
R
2
)] / jC
1
R
2
which has zero phase shift when:
1 
2
C
1
C
2
R
1
R
2
= 0
which enables the frequency to be calculated.
Now,
1/B = j (C
1
R
1
+ C
2
R
2
+ C
1
R
2
) / jC
1
R
2
1/B = (C
1
R
1
} + C
2
R
2
+ C
1
R
2
) / C
1
R
2
which is the required value of gain.
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It is not difficult to see that if all the CR terms have equal values, the required gain is 3.
Every oscillator has the requirement that, following some initial disturbance, the behaviour
is modified so as to increase that disturbance, until a useful level of signal has been built
up. This happens at some particular frequency, so that a periodic disturbance is built up.
An oscillator typically comprises an amplifier with feedback around it, as shown. If a
steady oscillation is to be maintained, then the input to the amplifier, must satisfy two
conditions:
it must have exactly the right amplitude to generate the output.
its phase must be correct to generate the output.
If we suppose that the amplifier has a positive gain A and no phase shift, then oscillations
will be maintained if AB=1 where B is the gain of the feedback network.
For sinusoidal oscillation, the condition AB=1 should be satisfied at the one
required frequency, and at no other. This will then cause only the one, required,
frequency to be produced.
The Wien Bridge oscillator is so called because the circuit is based on a frequency
selective form of Wheatstone bridge known as the Wien bridge. The form of oscillator you
will use is in fact a highgain amplifier with a Wien bridge around it, but it is more usual to
consider the circuit as being an amplifier of the requisite low value of gain, working with a
feedback network equivalent to half of the Wien bridge.
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The usual Wien bridge oscillator has equal values of capacitance and of resistance
arranged in the series/parallel arrangement shown.
The feedback factor B may be expressed in terms of R and the reactance X= 1/C of the
capacitors, as follows:
B = (R+jX)/[R+jX + jRX/(R+jX)]
= (R+jX)
2
/[(R+jX)
2
+ jRX)]
The condition for oscillation is AB=1.
Substituting B for the previously obtained expression:
(R+jX)
2
/[(R+jX)
2
+ jRX)]
and crossmultiplying, the condition becomes:
A(R+jX) = R+jX + jRX/(R+jX)
A(R+jX)
2
= (R+jX)
2
+ jRX
(A1) (R+jX)
2
= jRX
(A1) (R
2
X
2
+2jRX) = jRX
(A1) (R
2
X
2
) +(A  3) jRX = 0
The condition:
(A1) (R
2
X
2
) +(A  3) jRX = 0
becomes true if both the real and the imaginary parts of the lefthand side are zero;
ie, if A=3 where A is the required gain of the amplifier, and R
2
= X
2
, so that R = +/ X.
Since X = 1/C, the latter condition gives the frequency of oscillation
thus:
R = 1/C
so that
= 1/CR
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4.1.5 Practical 1: Basic Wien Bridge Oscillator
In this practical an operational amplifier is provided with a simple negative feedback
network which allows its gain to be adjusted within narrow limits.
A second, positive, feedback network incorporates the Wien halfbridge, comprising
series and parallelconnected R and C
The positive feedback has zero phase shift at just one frequency. It is shown in the
Theory that both zero phase shift and a particular value of gain are needed to
maintain oscillation.
You will see that excessive gain causes oscillation to build up until limited by the available
output swing from the amplifier.
You will need to be careful when adjusting the gain control in order to see oscillations
building up and dying down.
Notice that the frequency meter may give high or erratic readings before falling to 0Hz as
the amplitude is decreased. This is typical of frequency meter behaviour when the signal
falls to a level it cannot distinguish from noise.
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4.1.5.1 Procedure
You have available an oscilloscope and a frequency meter. The frequency meter is
always connected to the oscillator output.
R5 is adjustable to control the gain of the amplifier.
Start with the gain control at maximum. Look at the waveforms at both monitoring points,
while varying the gain. Observe how a very slight change of gain can stop or start
oscillations.
You will need to return to the practical and make some measurements in order to answer
the questions.
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4.1.5.2 Questions
1. What determines the amplitude of oscillator output when the gain slightly exceeds that
required for oscillation?
2. Do you think that this oscillator could be designed to work satisfactorily, taking into
account production tolerances and other component variations?
3. With the circuit adjusted for a nearly sinusoidal oscillation, what is the frequency :
a) as measured by the frequency meter?
b) measured another way?
4. Given that C
1
= C
2
= 1nF and that R
1
= R
2
= 15k,what does theory say the frequency
should be? Compare with the answers to question 3.
5. Why is the waveform at test point 2 not the same as at test point 3 (ignoring the
amplitude difference)?
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4.1.6 Practical 2: Amplitude Stabilisation
In the Basic Wien Bridge Oscillator Practical you saw how if the gain is very slightly below
the value required for oscillation, the oscillator stops working. On the other hand, with only
slightly excessive gain, the oscillation builds up until the amplifier distorts. How do we
ensure a stable sinusoidal signal?
In practice, oscillations present in linear systems will either increase steadily or decrease
steadily. An exact gain value cannot be achieved due to component tolerances and
component variations with temperature and age.
What we have to do is to find some kind of nonlinearity which will allow the gain to
vary with signal strength, but not distort the signal too much.
This problem is common to all oscillators based on linear amplification. The solution
adopted in this assignment is usable on such oscillators of many different kinds.
The diagram shows the negative feedback path modified by the inclusion of another path
containing a pair of backtoback diodes. These do not conduct significantly until the
voltage across them is about 0.6  0.7 V. But once current has started to flow, the voltage
drop will increase only slightly.
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Imagine that, with the diodes NOT conducting, the negative feedback sets the gain
slightly higher than is needed for oscillation. Oscillations will start to build up.
Once the amplitude has reached some 0.7 V the diodes will start to conduct, increasing
the negative feedback. More negative feedback will be applied, tending to reduce the
gain. There will be an amplitude for which the gain is just right to maintain
oscillation. Any variation from this amplitude will change the gain, tending to keep the
amplitude constant.
4.1.6.1 Procedure
In this practical the hardware is as in the Basic Wien Bridge Oscillator Practical, except
that an additional negative feedback path has been introduced, including the backtoback
diodes.
Start with the gain control at maximum. Look at the waveforms at both monitoring points,
while varying the gain. Observe carefully any difference between the effect of the gain
control in this Practical compared with the Basic Wien Bridge Oscillator Practical.
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4.1.6.2 Questions
1. How easy is it to control the amplitude, compared with Practical 1?
2. What limits the ratio over which (R4 + R5) can be varied, given that other components
have their nominal values? Explain your answer.
3. What happens when (R4 + R5) goes outside the proper range of values?
4. Set R5 in the middle of the range giving stable sinusoidal oscillation.
Measure the signal amplitudes at the input and output of the amplifier.
Calculate the amplifier gain and compare with the theoretical value.
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4.1.7 Practical 3: Changes from Standard
It is convenient, from a manufacturer's point of view, to have the same values R and C in
the series and parallelconnected parts of the Wien halfbridge. This is not necessary for
oscillation however.
This Practical is provided simply to show that a Wien bridge oscillator can have differing R
and C values, for which the standard theory does not apply. See the Wien Bridge
Oscillator Theory.
4.1.7.1 Procedure
In this practical the hardware is as in the Amplitude Stabilisation practical, except that the
values of R
2
and R
4
have been increased.
Set the gain control to give a steady sinusoidal oscillation, then, using the large
oscilloscope, determine the gain of the amplifier from its '+' input terminal to the output.
You will find that the result is different from the value of 3 given in the Wien Bridge
Oscillator Theory. This is not because the Theory is wrong, but because the circuit no
longer conforms to the assumptions of the Wien Bridge Oscillator Theory.
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4.1.7.2 Question
1. What value of amplifier gain did you measure?
2. Was this a change from the value of 3 given for the standard basic theory?
3. Compare the gain and frequency with the theoretical values :
gain = (C
1
R
1
+ C
2
R
2
+ C
1
R
2
) / C
1
R
2
and
frequency = /2 (PI) , where 1 
2
C
1
C
2
R
1
R
2
= 0.
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4.2 LC Oscillator Assignment
There are several detailed kinds of oscillator which use a resonant circuit (inductance and
capacitance together) to set the frequency of oscillation. This assignment deals with one
particular circuit of this kind, which is a typical example.
4.2.1 Objectives
On completion of this assignment you will be familiar with:
The operation of a tunedcollector oscillator,
The factors that determine its amplitude,
The stability of its frequency with supply variations.
4.2.2 Practicals
Practical exercises are provided as follows:
Practical 1: TunedCollector Oscillator
Practical 2: Effect of Supply Variations
4.2.3 Workboard Required
Signal Sources Workboard 53110 which comprises the following blocks:
Wein Bridge Oscillator,
LC Pass Filter,
Crystal Oscillator,
Multivibrator.
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4.2.4 Theory
4.2.4.1 LC Oscillator
The diagram shows a simple transistor amplifier having the parallel tuned circuit, L,C as a
load. The practical inductor has resistance, shown as r.
The transistor will be considered as a simple device whose collector current is dependent
only on the baseemitter voltage with a mutual conductance g. For small ac signals,
i
c
= g v
in
This current flows in the load, whose impedance Z is given by:
1/Z = jC + 1/(r + jL)
which can easily be rearranged to give:
Z = (r + jL) / (1  w
2
LC + jCr)
so that the voltage across the load becomes:
g v
in
(r + j L) / (1 
2
LC + j Cr)
The load voltage,  g v
in
(r + jL) / (1  w
2
LC + jCr), is applied to the impedance (r + jL)
to produce a current in L,
i
L
= g v
in
/ (1 
2
LC + jCr),
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whose phase is 90 degrees when
2
LC = 1.
A secondary coil is now coupled to L, with mutual inductance M. This secondary has
induced in it a voltage jMi
L
; ie:
 jM g v
in
/ (1 
2
LC + jCr)
which, when:
2
LC = 1,
becomes:
 jM g v
in
/ jCr
=  (Mg/Cr) v
in
.
Clearly, if Mg/Cr = 1, the secondary winding can supply the base voltage required to
maintain oscillation.
The minus sign is important as it connected to the start of the winding L, then the base
must be connected to the finish of shows the polarity with which the feedback winding
must be connected. Conventionally winding polarities are identified as start and finish,
with the winding sense taken to be the same for each winding. The value of M would be
taken as positive if increasing positive current into the start of one winding produced a
positive voltage at the start of the other winding.
In this case, if the transistor's collector is the secondary winding. The 'o' symbols in the
diagram conventionally show starts of windings.
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This fairly simple analysis has neglected the effect of base current flowing in the
secondary winding. This is a reasonable approximation when the current gain of the
transistor is large, and assuming also that stray capacitances do not provide significant
additional current paths.
There are numerous other factors in a complete analysis, of which a few instances are
mentioned below.
When the current in the secondary winding is significant, it becomes important to take into
account the effect of the leakage inductance of the secondary winding.
The transistor has various imperfections. The collector current is to some extent
dependent on collectoremitter voltage. There is stray capacitance between collector and
base, providing an additional feedback. There are delays due to the semiconduction
process.
One of the more important effects of these various imperfections is that the frequency is
not exactly the resonant frequency of L,C, but is influenced by stray factors which can
be variable.
4.2.4.2 Calculation of Stray Capacitance
The resonant frequency of an LC circuit is given by:
2
LC = 1
therefore:
C = 1 / 4
2
L f
2
Suppose that we start with a nominal tuning capacitance C and stray capacitance C
s
,
giving a total capacitance C + C
s
and frequency f
C
.
If the nominal tuning capacitance C is increased by a factor n, the new total capacitance
will be nC + C
s
and frequency f
nC
.
So the square of the frequency will be altered in the ratio r given by:
r = ( C + C
s
) / ( nC + C
s
) so C + C
s
= r ( nC + C
s
)
Collecting C
s
terms on the left:
C
s
( 1  r ) = C ( nr  1 )
so that the stray capacitance
C
s
= C ( nr  1 ) / ( 1  r ) , where r = ( f
nC
/ f
C
)
2
.
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4.2.5 Practical 1: TunedCollector Oscillator
This practical introduces the LCtuned oscillator, after it you will understand:
1. How feedback is taken to produce the oscillation.
2. How the presence of the tuned circuit sets the frequency and ensures that the
oscillation will be at least approximately sinusoidal.
3. How stray capacitance affects the tuning.
The diagram shows a transistor amplifier having a parallel tuned circuit as a load.
R
1
, C
1
and C
2
are present simply for biasing purposes.
If an ac signal voltage were applied to the base of the transistor, an amplified version of
the signal would appear at the collector. Maximum amplification would occur when the
frequency of the signal equalled the resonant frequency of the tuned circuit.
There is always some minute amount of noise current in any circuit, distributed over all
frequencies. The noise at the resonant frequency of L and C builds up a voltage across L.
The current in L lags that voltage by about 90 degrees.
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A secondary winding is coupled to L, and this has induced in it a voltage which leads the
current by 90 degrees, so it is in phase with the voltage across LC.
The amplifier inverts the signal at the resonant frequency  while the transistor base is
going positive, the collector will go negative. So in order to build up oscillations, the
secondary winding has to be reversed compared with L, the primary, as is shown by
coloured 'o' symbols by the windings.
Note that most of the discussion has been about what happens at the resonant frequency.
At other frequencies, not only is the loop gain reduced, but its phase changes fairly rapidly
with frequency. Since correct phase is necessary to maintain oscillation, no
oscillations can be produced at frequencies away from resonance.
4.2.5.1 Procedure
The hardware includes two switches. One, as shown, switches in a capacitor C' of equal
capacitance to C. The other switch (not shown) reverses the two connections to the
feedback (lefthand) winding.
Observe the effect of each of the switches.
Use the spectrum analyser to measure the frequency of the signals.
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4.2.5.2 Questions
1. Is the polarity of the feedback important, and if so why?
2. Which components in the circuit do you think are the most important in determining
the frequency of oscillation?
3. What two frequencies of oscillation can you measure?
4. What is the effect of doubling the tuning capacitance? Your answer should include a
mention of a numerical factor or ratio, with explanation if possible.
5. Using your last answer, can you calculate the stray capacitance?
6. What is the value of inductance L?
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4.2.6 Practical 2: Effect of Supply Variations
For reliable operation, an oscillator usually has to be provided with a higher loop gain than
is needed to start oscillation. Oscillations therefore build up in amplitude until something
happens to reduce the effective gain, preventing further build up.
Often, varying the supply voltage will greatly affect the oscillatory amplitude.
This practical allows you to study the oscillator's behaviour over a range of supply voltage
between about 5 and 10 V.
The mechanism by which the amplitude is limited in this oscillator is the nonlinearities
due to the cuttingoff of the transistor.
Note :
1. How effective the amplitude control is.
2. How the presence of the LC circuit prevents
4.2.6.1 Procedure
The hardware is now configured as for the previous TunedCollector Oscillator Practical,
(including the switched capacitor, not shown) except that the positive supply voltage is
now variable between approximately +5 V and +10 V.
Make sure that the feedback switch is set to allow oscillation.
Observe the effect of varying the supply voltage on the amplitude and frequency of the
output.
The changing dc conditions can be seen in terms of the dc voltage at the transistor's
emitter.
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4.2.6.2 Questions
1. Some nonlinearity in the circuit is needed to control the amplitude. Which component
provides this?
2. Name the nonlinear behaviour which limits the amplitude.
3. What is the function of the two diodes in the circuit?
4. Can you explain why the amplitude decreases as the dc in the transistor increases?
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4.3 Crystal Oscillator Assignment
4.3.1 Objectives
On completion of this assignment you will be familiar with:
The behaviour of a quartz crystal,
The use of a quartz crystal for stabilising the frequency of an oscillator,
The fundamental and overtone modes of oscillation of a crystal oscillator.
4.3.2 Practicals
Practical exercises are provided as follows:
Practical 1: Fundamental and Overtone Modes
4.3.3 Workboard Required
Signal Sources Workboard 53110 which comprises the following blocks:
Wein Bridge Oscillator,
LC Pass Filter,
Crystal Oscillator,
Multivibrator.
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4.3.4 Theory
4.3.4.1 Quartz Crystal
Crystal oscillators are used when a very stable frequency of oscillation is desired.
There are many circuits for a crystal oscillator. Many such oscillators are simple LC
oscillators with a crystal inserted into the circuit at some point. Alternatively, a crystal
oscillator can be constructed using logic gates, again associated with a crystal in the
circuit.
The crystal effectively only conducts well at one particular frequency. So by putting the
crystal in an oscillator circuit, one ensures that the oscillator can only oscillate at that one
frequency (if at all).
We will see why the crystal has this special frequency for conduction in a moment.
A crystal for frequency control is usually of quartz, a substance which has two useful
properties:
1. It has very little mechanical damping: it 'rings' very easily, like a bar of steel
rather than a bar of lead.
2. It shows the piezoelectric effect. That is, when it is mechanically stressed it
produces an electrical field; when it is electrically stressed it changes shape
mechanically.
Its mechanical 'ringing' (property number 1) produces (owing to property number 2) a
resonant electrical effect. This is made available at terminals connected to electrodes on
the crystal faces.
4.3.4.1.1 Equivalent Circuit of a Crystal
Because the mechanical vibration or 'ringing' of the crystal has electrical effects, there
appears to be an electrical circuit capable of 'ringing' between the terminals. It is called
the 'equivalent circuit'.
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Because the equivalent circuit can resonate or 'ring', it has inductance and capacitance in
it, shown as L and C in the diagram. There is also some inevitable capacitance between
the terminals, shown as C
stray
.
Component values in the equivalent circuit for a typical 1MHz crystal are shown above.
Compare C with the typical capacitance of a tuning capacitor at 1MHz (a few hundred pF).
L and C have high values of reactance (in the region of 63 megohms) so the slightest
departure from resonant frequency causes the crystal's impedance to rise very
rapidly.
Inevitably there is some energy loss, or damping. This has an effect typically like that of
placing one or two thousand ohms of resistance in series with L.
4.3.4.1.2 The Resonances
The important resonance is the series one, when
jL + (1/jC) = 0
or
2
LC = 1
There is a parallel resonance also, which occurs when the
2
LC' = 1
where C' is the value of C and C
stray
in series.
With the values in the diagram, C' will be about 0.1% different from C, so that the parallel
and series resonant frequencies will differ by about 0.05%.
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4.3.4.2 Crystal Oscillator
Consider Fig 1, in which all dconly components have been omitted. Imagine that the
collector current has a component at the resonant frequency of L and C. Because of
resonance, the current in L and the current in C greatly exceed the collector current.
We can 'tap off' a small portion of the capacitor current, the portion still being as great as
the collector current, and feed it into the emitter.
To do this, some capacitance is removed from C and replaced by a capacitor connected
to the emitter.
Fig 2 shows a form of oscillator using this principle, which is especially useful at the higher
frequencies of tens of MHz upward.
The fedback emitter current is phaseshifted by the capacitor's reactance. To make the
phase correct to maintain oscillation, some lagging phase shift must be added.
This lag is provided by the transistor itself. Injection of emitter current causes holes to drift
across the base barrier to the collector, but this takes time.
The circuit in the Practical is slightly different in detail, in order to achieve similar
functioning at two widely different frequencies.
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You can see that the difference from Fig 2 is just that the emitter feedback is derived in a
slightly different way.
Thevenin's Theorem states that any linear twoterminal network can be replaced by an
equivalent circuit comprising a voltage source equal to the unloaded voltage at the
terminals, and a series impedance such that the shortcircuit currents of the original and
equivalent circuits match.
Fig 4 shows how, in the basic circuit of Fig 1, splitting the capacitance C into two series
capacitances, and feeding the emitter from their junction, is equivalent to Fig 2 with the
feedback source voltage reduced.
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4.3.5 Practical 1: Fundamental and Overtone Modes
This Practical shows how the introduction of a suitable quartz crystal into an oscillator can
stabilise its frequency.
The particular form of oscillator chosen is just one example of many kinds of oscillator in
which a crystal can be used. It is an LC tuned oscillator, with variable tuning capacitance.
Quartz is piezoelectric  mechanical vibration produces electrical effects and vice
versa.
Because a crystal is a block of material it can vibrate in several different modes at
different frequencies. Ordinarily a crystal is designed for use in only one of these modes,
with one specific frequency of oscillation.
If the oscillator is so designed that it cannot oscillate at one of the crystal's resonant
frequencies, it may still oscillate at another. This is shown in the Practical by providing an
LC tuning circuit which is tunable over a wide range.
4.3.5.1 Starting the Practical
At the start of the Practical you must turn the tuning control fully counterclockwise (to its
lowest frequency).
This is because the crystal provided is designed to be used in fundamental mode, at
2MHz, and is not very good at starting oscillation at an overtone frequency.
Because the crystal's performance at the overtone frequency is not defined, the oscillator
has been designed to force oscillations even with poor crystal performance. This forceful
(and not very typical) action can produce oscillations at various frequencies, not
necessarily those of the crystal. But it should be clearly seen, with the aid of the Spectrum
Analyser, that the oscillator does 'lock on' to two different crystal frequencies.
A point to notice is the very erratic behaviour when the oscillator tuning is altered just far
enough from the crystal's fundamental to stop the fundamental oscillation. This may be
thought of as a kind of fight to set the frequency, between the crystal and the tuning
circuit.
A more normal oscillator design would simply refuse to oscillate at any frequency except
the intended crystal frequency.
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4.3.5.2 Procedure
The tuning control appears on the circuit diagram as C1 and C2.
Start with the tuning control set fully counterclockwise. Measure the signal frequency
with the spectrum analyser and the large oscilloscope.
Alter the setting of the tuning control gradually and observe what happens.
Note particularly any frequencies to which the circuit seems to 'lock', needing an
appreciable movement of the tuning control before a change in frequency results. This
is most easily seen using the spectrum analyser.
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4.3.5.3 Questions
1. What are the two frequencies at which the oscillator frequency becomes locked to the
crystal?
2. What is the ratio between these two frequencies?
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4.4 Multivibrator Assignment
This assignment is about the simple multivibrator. This is a type of oscillator which uses
the transistors as switches. It produces waveforms which are far from sinusoidal.
4.4.1 Objectives
On completion of this assignment you will be familiar with:
Factors that determine the frequency of oscillation of a multivibrator,
Multivibrator waveforms and how they are affected by circuit design,
The effect of leakage in the transistors baseemitter diodes and how to prevent it,
Variation in the mark/space ratio of a multivibrator,
The spectrum of the multivibrator output signal.
4.4.2 Practicals
Practical exercises are provided as follows:
Practical 1: Basic Circuit
Practical 2: Effects of Variable Supply
Practical 3: Mark/Space Ratio Control
4.4.3 Workboard Required
Signal Sources Workboard 53110 which comprises the following blocks:
Wein Bridge Oscillator,
LC Pass Filter,
Crystal Oscillator,
Multivibrator.
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4.4.4 Theory
4.4.4.1 The Simple Multivibrator
The operation of a multivibrator is considered in the Basic Multivibrator Circuit
Background, which should be read before continuing. As the Background explains, the
multivibrator is characterised by quick transitions from one state to another, followed by
'relaxation' periods in which the charge on the capacitors changes comparatively slowly.
To calculate the time taken over these changes, we have first to identify the paths taken
by the charging and discharging currents.
4.4.4.1.1 Start of Discharge
The diagram shows the situation where the input terminal has just gone from near the
supply potential down to zero volts. The capacitor previously had its lefthand terminal at
V
s
and its righthand terminal at a small positive potential V
f
due to the forward voltage
drop across the conducting baseemitter diode
Before the input transition, the voltage across the capacitor was V
s
 V
f
, or approximately
V
s
if V
f
is neglected.
Bringing the input terminal down to 0V must bring the base of the transistor down to  (V
s

V
f
). There is therefore a voltage across R
b
equal to (2V
s
 V
f
).
The resulting current (2V
s
 V
f
)/R
b
starts to discharge the capacitor.
This current decays exponentially, with a time constant CR
b
.
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The current is therefore given by:
i R
b
= (2V
s
 V
f
) exp(t/CR
b
)
from which the voltage on the transistor base can be calculated as:
V
b
= V
s
 i R
b
V
b
= V
s
 (2V
s
 V
f
) exp(t/CR
b
)
This transistor ceases to be cut off when V
b
has reached the value V
f
;
ie, at t given by:
V
f
= V
s
 (2V
s
 V
f
) exp(t/CR
b
)
exp(t/CR
b
) = (2V
s
 V
f
) / (V
s
 V
f
)
t = CR
b
ln [ (2V
s
 V
f
) / (V
s
 V
f
) ]
It is not too easy to see the meaning of:
t = CR
b
ln [ (2V
s
 V
f
) / (V
s
 V
f
) ]
But since normally V
f
is small compared with V
s
, we can simplify to find an approximate
value for t :
t
1
= CR
b
ln 2
= 0.69 CR
b
4.4.4.1.2 Capacitor Charging
The other state that needs considering is when a transistor has just ceased conducting. At
that time its collector is at approximately 0 V (with respect to the emitter). The circuit is
effectively as shown here.
The capacitor charges through the collector resistor and the other transistor's base
emitter diode. The latter is assumed to produce a constant small voltage drop V
f
.
If we now measure time t' starting from this last transition, the current is given by:
i R
c
= (V
s
 V
f
) exp (t'/CR
c
)
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But Ohm's Law leads us to:
V
c
= V
s
 i R
c
V
c
= V
s
 (V
s
 V
f
) exp (t'/CR
c
)
or neglecting V
f
,
V
c
= V
s
[1  exp (t'/CR
c
)]
4.4.4.1.3 A Check on the Collector Waveform
We must check our previous assumption that the collector rose to nearly +V
s
. To do this
we must substitute t' with the previously calculated time (the approximate value t
1
will do):
t
1
= CR
b
ln 0.5 = 0.69 CR
b
.
So at the end of this halfcycle, the collector voltage has risen to:
V
c
' = V
s
[ 1  exp (0.69 CR
b
/CR
c
) ]
V
c
' = V
s
[ 1  exp (0.69 R
b
/R
c
) ]
Now, R
b
is typically many times R
c
, because the ratio can approach the current gain of the
transistor. It is 68 in the multivibrator on the board for instance.
Putting R
b
/R
c
= 68, we calculate:
V
c
' = V
s
[ 1  exp (0.69 * 68) ]
V
c
' = V
s
[ 1  exp (47) ]
V
c
' = V
s
[ 1  0.0000000000000000000039 ]
Even if we put R
b
/R
c
as small as 5, V
c
' comes very close to V
s
V
c
' = V
s
( 1  0.0067) = 0.993 V
s
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4.4.4.2 The Effect of Diodes on Multivibrator Theory
In the The Simple Multivibrators Theory, the assumption was made that the baseemitter
diode of each transistor had a substantial constant forward volt drop V(f) when
conducting. If similarly we assume here that there is a constant voltage drop V
df
across
the diodes added in series with each emitter, the theory previously presented can still be
applied, with these adjustments:
1. V
s
is replaced by V
s
 V
df
2. Other voltages are unchanged if measured from the emitters. Alternatively, raise
each of the values for V
b
, V
c
by V
df
.
4.4.4.3 Variable Mark/Space Ratios
The multivibrator need not have equal time constants associated with its two transistors. If
the time constants differ then correspondingly different times will be needed to discharge
the two capacitors.
It was shown in the previous Theory that the time to discharge one of the capacitors is:
t = CR
b
ln [ (2V
s
 V
f
) / (V
s
 V
f
) ]
which may for the present be written as t = k CR
b
,
where k = ln [ (2V
s
 V
f
) / (V
s
 V
f
) ]
In the Multivibrator with variable mark/space ratio Practical a control is introduced which
enables the value of R
b
to be increased in one half of the multivibrator and at the same
time decreased in the other half.
The period of the multivibrator is equal to the sum of the times for which one transistor is
cut off and the other is cut off. It is therefore equal to:
k C
1
R
b1
+ k C
2
R
b2
.
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In our particular case the values of C
1
and C
2
are equal, say to C, so the period is T = kC
(R
b1
+ R
b2
).
However, (R
b1
+ R
b2
) = R
1
+ R
2
+ R
var
, which is constant. So the period T is constant. The
frequency, 1/T, is therefore also constant.
4.4.5 Practical 1: Basic Circuit
Consider a pair of amplifiers with feedback around them. If the product of their gains is
positive and greater than one, any small disturbance in the circuit will be amplified,
returning to the starting point greater than before. The process continues until something
stops further amplification from occurring.
If there are no delays in the loop caused by filters or tuned circuits this process will
happen very quickly. What stops the process is usually limiting which takes place in one
or both amplifiers.
Each amplifier stage in our simple multivibrator is an a.c. amplifier as shown above.
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If the signal source voltage goes positive, current flows through C and the emitterbase
diode, charging C. This current dies away when the capacitor is charged, but there is also
current flowing through R, causing the transistor to conduct heavily.
The result is that the collector suddenly falls to a low potential.
Now consider what is happening in the second stage of our twostage amplifier. The
sudden drop in potential at the first stage's collector causes the base of the second stage
transistor to go negative, cutting off the transistor.
This transistor's collector therefore rises to a potential near the supply rail.
Note that this is just the condition needed to operate the first stage in the way described
previously.
The transistor does not remain cut off indefinitely, however.
The resistor R provides a current which alters the charge on the capacitor, raising the
potential of the transistor base. Eventually the base goes positive with respect to the
emitter and the transistor starts to conduct.
This transistor now enters the state previously described for the first one, and it conducts
heavily while the other one cuts off.
The process repeats indefinitely, switching each transistor on in turn, and the circuit
oscillates.
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4.4.5.1 Procedure
The oscilloscope should be used to examine the waveforms at all four available test
points. (They are NOT entirely symmetrical.)
The frequency meter is connected to the collector of the second stage's transistor at test
point 11 . Make a note of the frequency value.
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4.4.6 Questions
1. Which part of which waveform(s) do you think is important in determining the
frequency of oscillation?
2. What difference do you notice between the waveforms at the two collectors (apart
from the phase difference). Can you think what causes it?
3. Does this difference have any significant effect on the frequency of oscillation?
4. Theoretically, the time for one halfperiod of oscillation is approximately 0.69 CR
b
. In
this multivibrator C is 1nF and R
b
is 68k.Calculate the frequency of oscillation and
compare it with the measured value.
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4.4.7 Practical 2: Effect of Variable Supply
In this Practical two new features are introduced:
1. The supply voltage is variable by a control on the board.
2. A pair of diodes is introduced. (The diodes were present in the first Practical of this
assignment but not shown for simplicity.) They can be switched in and out of circuit.
When the collector of one transistor drops from +V
s
to 0 V, the base of the other transistor
is taken to about V
s
. The baseemitter diode is not designed to withstand more than
about 5V reverse bias, so it partially breaks down in a similar way to a Zener diode.
The purpose of the diodes is to prevent reverse current from flowing. This both
protects the transistors and makes the behaviour of the multivibrator more predictable.
This time the circuit is nominally symmetrical. This gives the output spectrum a particular
character which should be noted.
4.4.7.1 Procedure
In this practical the collector resistors are equal. Diodes in series with each emitter
prevent breakdown of the baseemitter diodes of the transistors. The supply voltage can
be varied. Start with it turned up to maximum.
Use the Diode button to switch the diodes in and out of the circuit. Observe that the
frequency changes. Note any frequency changes caused by varying the supply voltage.
Use the largesized Spectrum Analyser to see which are the important harmonics of the
collector waveforms.
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4.4.7.2 Questions
1. Which has the bigger effect on the frequency of oscillation  varying the supply voltage
between 5 V and 10 V, or switching the diodes in and out of the emitter connections?
2. Is the behaviour of the baseemitter diodes like a Zener diode, with a sharp 'knee' in
the characteristic, or more a general tendency to leak reverse current?
3. Which are the important harmonics in the spectrum of the collector waveforms?
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4.4.8 Practical 3: Mark/space Ratio Control
In this Practical a new feature is a variable mark/space ratio control.
This is a potentiometer configured to appear as two resistors in series with the timing
resistors R1, R2. The supply voltage is connected to the slider on the potentiometer so
that while R
b1
is effectively increased, R
b2
is effectively decreased.
As a result of this, altering the control makes one capacitor charge faster and the other
slower. The charge time needed for each of the capacitors is changed, and the
mark/space ratio is altered.
It is shown in the Theory that changing the mark/space ratio in this way will not alter the
period of the output signal.
The term 'mark/space ratio' is a throwback to the early days of telegraphy, when a line
was said to be in the mark condition when a Morse key was pressed and the space
condition when the key was released. Mark and space are just the two possible states of a
twostate waveform.
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4.4.8.1 Procedure
The mark/space ratio can be adjusted by a control provided on the board. Do this while
observing each of the monitor points in turn. Pay particular attention to the slopes of the
waveforms at the transistor bases.
Examine the frequency as the mark/space ratio is varied.
Observe the changes in the pattern of the frequency harmonics as the mark/space ratio
is varied, using the large Spectrum Analyser.
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4.4.8.2 Questions
1. Explain with the aid of sketch waveforms what happens when the mark/space ratio
control is adjusted, and why.
2. Does the frequency vary when the mark/space ratio is adjusted?
3. Would the frequency vary with mark/space ratio if the two capacitors were of different
capacitance? Explain your answer briefly. (Hint  Calculate time constants for the two
cases in which the variable control is set at the end of its travel.)
4. What is special about the spectrum of the collector waveform when the mark/space
ratio is 1:1?
5. If the collector waveform were a true square wave, the fundamental, 3rd and 5th
harmonics should be in the ratio 1:(1/3):(1/5). Compare these with the measured
ratios.
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Notes
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5 Assignments using the Tuned Circuits and Filters Workboard
5.1 Audio LowPass Filters Assignment
5.1.1 Objectives
On completion of this assignment you will be familiar with:
The frequency and transient response of a passive lowpass filter,
A filter's frequency response using a noise source and spectrum analyser,
The frequency and transient response of an active filter,
Matching considerations for passive and active filters.
5.1.2 Practicals
Practical exercises are provided as follows:
Practical 1: Passive LowPass Filter
Practical 2: Passive LowPass Filter, Swept Frequency
Practical 3: Active LowPass Filter
Practical 4: Active LowPass Filter, Swept Frequency
5.1.3 Workboard Required
Tuned Circuits and Filters Workboard 53120 which comprises the following blocks:
Signal Generator,
Low Pass Filter,
Band Pass Filter,
Amplifier with AGC.
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5.1.4 Theory
5.1.4.1 Filters
A filter is a circuit which passes signals of some frequencies freely, but attenuates at other
frequencies. This assignment is about lowpass filters.
5.1.4.1.1 Lowpass Filters
An ideal lowpass filter passes signals of all frequencies below a certain value and
does not pass signals of frequencies above that value.
The frequency at which the signal starts to be attenuated is called the cutoff frequency.
Since the cutoff is not quite sharp, the exact value must be based on some measure of
how much the signal is reduced. This is typically 3 dB below the level in the pass band.
To pass some frequencies and reject others, a filter must contain components that have
impedances that vary with frequency. This means capacitors and/or inductors must be
used, often together with resistors whose impedance (resistance) does not vary with
frequency.
If the filter does not have any active devices (transistors, IC's, etc.) in its circuit, it is called
a Passive Filter.
Filters can be classified as passive or active. The assignment includes one filter of each
class. 'Active' implies that the circuit contains amplifying devices such as
transistors or opamps. 'Passive' means having no such devices.
5.1.4.1.2 Passive Filters
Passive filters consist of networks of inductors and capacitors. They have the advantage
of working without power supplies. Their disadvantages are mainly apparent at low
frequencies, for which both inductors and capacitors need to be large and expensive.
Inductors also tend to have appreciable resistance, making the action of the filter
imperfect.
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5.1.4.1.3 Active Filters
Passive filters consist of networks of inductors and capacitors. They have the advantage
of working without power supplies. Their disadvantages are mainly apparent at low
frequencies, for which both inductors and capacitors need to be large and expensive.
Inductors also tend to have appreciable resistance, making the action of the filter
imperfect.
5.1.4.2 Transmission Line Characteristic Impedance
The concept of characteristic impedance first arose in the theory of transmission lines
for telegraph purposes (they are now used for carrying RF and other signals).
Imagine an infinite transmission line. In practice this means a line so long that a signal
takes so long to get to the far end that it virtually never comes back.
Now suppose that we wish to send a signal of voltage v along the line. (v can be a
function of time, like V
p
sin wt). If that signal is applied at one end of the line, a current i
will flow into the line.
The characteristic impedance, often denoted by Z
0
, is then defined as:
Z
0
= v / i.
Now let us look at the voltage and current at a point P, some way down the line.
We find them to be v
1
and i
1
respectively. The line remaining is still infinite, so the ratio of
voltage to current must again be Z
0
, and we can see that Z
0
is also equal to v
1
/ i
1
.
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This means that we can cut the line at P and connect an impedance Z
0
at point P without
altering the behaviour of the first section of line at all.
The line is now of finite length and has been terminated with Z
0
. It is said to be matched
to Z
0
, the Characteristic Impedance of the line.
The important point to be made about matching the load to the characteristic impedance
of the line is this :
A properly matched load absorbs all the power which arrives at it, whereas a
mismatched load causes 'spare' power to be reflected back towards the signal
source.
5.1.4.2.1 Filter Matching
If a filter is inserted into a line, ideally it should have an impedance equal to the line's
characteristic impedance at both pairs of terminals. If this condition is realised then the
filter causes no reflections.
In practice a resistive load is never a perfect match for a filter at all frequencies, so some
reflection always takes place.
5.1.4.3 Passive LowPass Filters
A filter is a circuit which passes signals in a selected range of frequencies freely (the
'passband'), but attenuates at all other frequencies (the 'stopband').
An ideal LowPass Filter lets through signals lower than a certain frequency without
attenuating them, but does not let through any signal with a frequency above that value.
The frequency above which no signals are passed is called the cutoff frequency.
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A real filter cannot have a response that changes instantly from zero attenuation to infinite
attenuation at a certain frequency. In a practical low pass filter the response slopes off at
higher frequencies, and the cutoff frequency is defined as being that at which the signal
has been attenuated by 3 dB.
To make the filter pass some frequencies and reject others, it must contain components
whose impedances vary with frequency; ie, capacitors and inductors.
If the filter does not have any active, amplifying devices in its circuit it is called a Passive
Filter. Resistors cannot be used in a passive filter because they would introduce
attenuation.
The theory of filters is too complex to explain here. But in simple terms, an elementary
lowpass filter has inductance in series with the signal path and capacitance in parallel
with it.
As the diagram shows, a filter section can be arranged in a pi or a T form. Either form can
be constructed from two identical halfsections.
The cutoff frequency, rad/s, for any of these filters is given by
2
LC = 1.
The 'design impedance', which approximates to the characteristic impedance at low
frequencies, is Z where
Z
2
= L/C
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5.1.4.4 Active LowPass Filters
An active lowpass filter will often have several stages like this having the same circuit,
though usually not identical component values.
The gain G of the amplifier is normally set by a feedback network (not shown for
simplicity). In the mathematics, 1/G occurs many times, so k is defined to be 1/G.
The input to the amplifier carries a voltage k times the output voltage. So, considering the
impedance of the righthand capacitor:
i = k jC ... (1)
Ohm's Law for the righthand resistor gives:
iR = e
x
 k e
o
... (2)
and the sum of the currents in the feedback capacitor and in the lefthand resistor is equal
to i, so:
i = (e
i
 e
x
)/R + (e
o
 e
x
) jC ... (3)
Combining equations (1) and (2), we get:
k e
o
jCR = e
x
 k e
o
e
x
= k e
o
(1 + jCR) ... (4)
Equating i from equations (1) and (3), then multiplying by R:
ke
o
jCR = e
i
 e
x
+ (e
o
 e
x
) jCR
ke
o
jCR = e
i
+ e
o
jCR  e
x
(1+ jCR)
Substituting for e
x
from (4):
ke
o
jCR = e
i
+ e
o
jCR  ke
o
(1+ jCR)
2
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which by routine manipulation, and replacing 1/k by G, becomes:
e
o
[  (
2
C
2
R
2
+ (3  G) jCR + 1] = G e
i
If = 0, and so at low frequencies, the gain is G. At high frequencies the gain
approximates to 1/
2
C
2
R
2
; ie, falling with increasing frequency. At frequencies given by
CR approximately 1, the behaviour of the filter is very dependent on the j term, with G as
a factor. This controls the damping of the response.
A large value of G produces a large response in the CR=1 region. A small value
produces a gradually drooping response. Usually several stages are used, with different
damping factors, to give a response which is maintained well up to the cutoff frequency
and then falls sharply.
5.1.4.4.1 Setting the Amplifier Gain G
To keep the diagrams simple, the gain of the amplifier has been supposed to be G. In
practice it is usual to employ operational amplifiers with very high gain, and with negative
feedback which sets the gain.
Since the gain of the amplifier is very high, for any value of output voltage, the difference
in potential between the two input terminals is negligibly small. So
e
in
= e
out
R
2
/ (R
1
+ R
2
)
The gain with this feedback is therefore
G = (R
1
+ R
2
) / R
2
= 1 + (R
1
/ R
2
)
5.1.4.5 Swept Frequency and Plotting
In testing communications equipment, we very often need to discover the frequency
response of some item. To do this an oscillator is used to provide the test signal, the
frequency of which can be set to each of a range of different values in turn. For each
frequency the output signal is measured. Usually the results of this are presented by
plotting a graph of the response against frequency.
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5.1.4.5.1 Swept Frequency Facility
Much tedious work can be saved if the oscillator frequency is varied automatically and the
output response is simultaneously measured and plotted. A facility of this kind is provided
in this equipment. A switch enables the oscillator to enter swept mode where the
frequency is modulated slowly by a sweep oscillator. The end frequencies for the sweep
can be set by controls on the board.
5.1.4.5.2 Using the Swept Frequency Facility
In the Practical you will be told, or you can find out experimentally, sensible values for the
start and end of the sweep.
To set up the sweep oscillator with these values, set the Sweep switch to ON whilst
viewing the oscilloscope.
Press and hold down the button on the module marked Min. This will force the
sweep oscillator to stay at its minimum value. The bottom frequency of the swept
range can now be set using the usual Frequency control whilst observing the value
of the frequency meter. Release the Min button when this has been done.
To set the top end of the swept frequency range:
Press and hold down the button on the module marked Max. This will force the
sweep oscillator to stay at its maximum value. The top frequency of the swept
range can now be adjusted as an offset from the minimum frequency that has just
been set. To do this, use the Range control on the module whilst observing the
frequency meter. When this has been set, release the Max button.
Due to the fact that the maximum swept frequency is set as an offset from the minimum
swept frequency, it is important to set the minimum and maximum in that order.
When the range has been set, you can start the Plotter.
The Sweep function is only enabled during those Practicals for which it required. For
other Practicals the Sweep switch should always be set to OFF. Failure to do this will
cause problems, usually failure of the oscilloscope to run or to synchronise.
5.1.4.5.3 How the Sweep Facility Works
It is not necessary to read this section to be able to use the Plotter.
When you select Plotter, the direct action of the Frequency control of the lefthand, signal
source, portion of the module is disabled. The frequency is instead modulated over a pre
set range by a further, lowfrequency, oscillator.
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The plotter takes readings and plots points for values of frequency and peak amplitude.
When taking a reading, the signal frequency is frozen and the peak amplitude of the
output signal is measured and stored to give the vertical deflection on the virtual plotter.
At the same time the modulating oscillator's output is also stored to give the horizontal
deflection.
The plotted points will be scattered at different frequencies, so a graph will gradually build
up as more and more points are plotted.
5.1.5 Practical 1: Passive LowPass Filter
This Practical uses a simple passive lowpass filter.
It comprises inductors and capacitors arranged as shown. It is designed to operate
between a 600ohm source and 600ohm load. The impedance presented by the filter is
not entirely resistive, so mismatching occurs near the cutoff frequency.
There are two stages to the Practical. In the first, you will examine the filter's response to
a sine wave at various frequencies. As you will see, this is quite markedly altered by
whether the correct load is present or not.
The second stage is to examine the response to sudden changes in the input signal. A
single such change is called a step function. This function is approximated by a square
wave produced by the audio oscillator.
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You will find that the effect of removing the high frequencies from the input signal is to
cause rounding or smoothing of the output waveform. Unequal transmission of
frequencies within the pass band gives rise to a distorted step function response.
Since altering the matching also alters the frequency response, you will be able to see
that a poor frequency response also worsens the transient response.
5.1.5.1 Procedure
Switch Sweep OFF. Set the audio level to give a sine wave output of maximum
deflection on the oscilloscope. Observe that this signal stays constant in amplitude as its
frequency is changed. Examine and contrast this signal with that at the filter output,
monitor point 13.
Use the Load Button to remove the load from the filter. Repeat your previous
observations. You should find quite noticeable changes.
Switch the audio oscillator to give a square wave, and repeat the steps above.
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5.1.5.2 Questions
1. What happens to the output of the filter as the signal frequency rises with the load
resister in circuit?
2. At what frequency does the output fall by 3 dB from its value at low frequencies?
3. If the oscillator's output is of constant amplitude, why does the input to the filter vary
with frequency?
4. With square wave input, what is the effect of removing the higher frequencies?
5. With square wave input, what is the effect of having no load resistor on the shape of
the waveforms?
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5.1.6 Practical 2: Passive LowPass Filter, Swept Frequency
Read the Swept frequency Background before continuing for information on how to use
the sweep oscillator and plotter.
This Practical is similar to the Passive LowPass Filter Practical, however the frequency
response will be displayed as a graph by an automatic process.
5.1.6.1 Procedure
Set the audio level for maximum sine wave output. Set the Sweep switch to ON.
Set the minimum frequency to the lowest at which the frequency meter and oscilloscope
still work. Set the maximum to the highest possible value.
Use the Load Button to remove the load from the filter. Repeat your previous
observations. You should find quite noticeable changes.
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5.1.6.2 Questions
1. Describe the general shape of the frequency response of the complete filter.
2. Describe how the frequency response changes when the load is disconnected.
3. The effect of disconnecting the load can be described in terms of the signal being
reflected back into the filter from the mismatched output terminals. Where else is
matching important in this circuit?
4. If the filter were perfectly matched to the source and load resistances at the ends, the
input to the filter would be half the source voltage. At what frequencies is this least
true?
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5.1.7 Practical 3: Active LowPass Filter
This Practical uses an active lowpass filter. It comprises two filter stages, each of which
has a circuit as shown although the second stage has different component values.
You can see that the input impedance of this filter is high compared with the 600 ohms
which is typical of a passive filter. The output impedance is low, because the output is
generated by an operational amplifier. It is therefore far easier to match the filter to
associated circuitry than a passive filter.
As with the passive filter, tests will be performed both with a sinusoidal signal to check the
frequency response, and with a slower square wave to check the transient response.
In a filter of this kind the characteristics are almost entirely determined by the
feedback networks around the operational amplifiers. The Practical allows you to
change the value of one resistance in such a network, so that you can see the changes to
the frequency and transient responses which result.
Making that change should enable you to see once more that frequency and transient
responses are related.
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5.1.7.1 Procedure
Switch Sweep OFF. Set the audio level for maximum sinewave output.
Observe that this signal stays constant in amplitude as its frequency is changed.
Compare this with the signal at the filter input, monitor point 4 .
Observe and contrast the signal at the filter output, monitor point 14.
Set the oscillator to give a lowfrequency square wave. Observe the output waveform.
Use the Stage Button to adjust the second stage response. Then observe the effect on
the output waveform. You may need to reduce the amplitude of the square wave to see
the full effect.
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5.1.7.2 Question
1. How important is the matching of the signal source impedance to the active filter?
Compare this with the situation when using the passive filter.
2. How does the frequency response of the active filter compare with the response of the
passive one?
3. The adjustment to the second stage frequency response gave it a response more like
the first stage. Why was this not a good thing?
4. What is the effect of a sharplypeaked frequency response on the transient response
of the filter?
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5.1.8 Practical 4: Active LowPass Filter, Swept Frequency
Read the Swept frequency Background before continuing for information on how to use
the sweep oscillator and plotter.
This Practical is similar to the Active LowPass Filter Practical. Indeed the principal object
of this Practical is to show that the behaviour of any filter made of inductors and
capacitors can be reproduced by a filter made of resistors and capacitors, together with
active devices (amplifiers).
Active filters are especially useful at low frequencies, because inductors having a good
ratio of reactance to resistance at low frequencies are large, heavy and expensive.
Conversely, passive filters are preferred at frequencies above about 50 kHz, because the
phase shifts in amplifiers become difficult to control and reactors for high frequencies are
comparatively inexpensive.
5.1.8.1 Procedure
Set the audio level for maximum sinewave output. Set the Sweep switch to ON.
Adjust the minimum frequency to the lowest value at which the oscilloscope and
frequency meter work. Set the sweep range to maximum. Then select the Plotter to plot
responses measured at points 14 and 12.
Use the Stage Button to alter the second stage of the filter. This will cause a peaky
response, so you will need to reduce the audio oscillator signal by a factor of about 3, to
avoid overloading the instruments. Repeat the previous observations.
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5.1.8.2 Questions
1. Describe the general shape of the frequency response of the complete filter.
2. The filter has two stages. Compare the response of the first stage only with that of the
complete filter.
3. What kind of frequency response would you expect the second stage to have?
4. When the second stage has its component values altered, what happens?
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5.2 RF Selectivity Assignment
5.2.1 Objectives
On completion of this assignment you will be familiar with:
The frequency response of a tuned circuit,
The transient response of a tuned circuit,
The effect of damping on the frequency response of a tuned circuit,
The frequency response of a crystal filter,
The effect of neutralisation on a crystal filter.
5.2.2 Practicals
Practical exercises are provided as follows:
Practical 1: L/C Tuned Circuit
Practical 2: L/C Tuned Circuit, Swept Frequency
Practical 3: L/C Tuned Circuit, Transient Response
Practical 4: Crystal Filter
Practical 5: Crystal Filter, Swept Frequency
5.2.3 Workboard Required
Tuned Circuits and Filters Workboard 53120 which comprises the following blocks:
Signal Generator,
Low Pass Filter,
Band Pass Filter,
Amplifier with AGC.
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5.2.4 Theory
5.2.4.1 Tuned Circuit
The theory of the tuned circuit is most easily derived either for inductance L, resistance r
and capacitance C all in series (where r is a usually a small resistance representing circuit
losses), or for L, C and a usually high resistance R, all in parallel.
We shall look at the parallel case, as it more nearly fits the Practicals.
For this circuit the parallel impedance Z is given by:
1 / Z = (1/j L) + j C + (1/R)
If we write
0
2
LC = 1 then:
1 / Z = j(
0
2
LC / jL) + j
0
C(/
0
) + (1/R)
= j
0
C y + (1/R)
where y = (/
0
)  (
0
/) and, if w is not too far from
0
, is a measure of the fractional
mistuning, since:
y = (
2

0
2
) /
0
= ( 
0
) (1/
0
+ 1/)
= 2 d /
0
approximately.
The expression:
1/Z = j
0
C y + (1/R)
is clearly a minimum of 1/R when y = 0, so Z has its maximum value R when =
0
.
For large mistunings, the term j
0
C y becomes large, while (1/R) stays constant. So R
does not greatly affect the response for large mistunings. But for zero mistuning, the
larger R, the larger is Z.
The selectivity is therefore highest for large R, implying small resistive losses.
It is generally true that low losses give good selectivity, associated with a high
impedance at resonance. This is true whether the losses are due to series
resistance, parallel conductance or other effects.
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5.2.4.2 Crystal Filter
A crystal for use in a filter is usually of quartz, a substance which has two useful
properties:
1. It has very little mechanical damping  it 'rings' very easily, like a bar of steel
rather than a bar of lead.
2. It shows the piezoelectric effect. That is, when it is mechanically stressed it
produces an electrical field and when it is electrically stressed it changes
shape mechanically.
The crystal's mechanical 'ringing' (property number one) produces a resonant electrical
effect (due to property number two). This is made available at terminals connected to
electrodes on the crystal faces.
A Crystal Filter behaves in the say way as a Tuned Filter, but has a much higher Q factor
of the order of 10,000, compared with a Q of about 100 for a Tuned Filter.
5.2.4.2.1 Equivalent Circuit of a Crystal
When the crystal vibrates or 'rings', it appears that there is an electrical circuit connected
between its terminals that is ringing. This is due to the electrical effects of the crystal's
vibration. The circuit is the 'equivalent circuit' of the crystal.
Because the equivalent circuit can resonate or 'ring', it must have inductance and
capacitance in it, shown as L and C in the diagram. There is also some inevitable
capacitance between the terminals, shown as C
stray
.
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The component values in the above diagram are for the equivalent circuit of a typical
1 MHz crystal. Compare C with the typical capacitance (a few hundred pF) of a tuning
capacitor operating at 1 MHz.
L and C have high values of reactance, in the region of 63 megohms, so the slightest
departure from resonant frequency causes the crystal's impedance to rise very
rapidly.
Inevitably there is some energy loss, or damping. This has an effect similar to that of one
or two thousand ohms of resistance placed in series with L.
5.2.4.2.2 The Resonances
The important resonance is the series one, when:
jL + (1/jC) = 0
or:
2
LC = 1
There is a parallel resonance also, which occurs when the:
2
LC' = 1
where C' is the value of C and C
stray
in series.
With the values in the diagram, C' will be about 0.1% different from C, so that the parallel
and series resonant frequencies will differ by about 0.05%.
5.2.4.2.3 Neutralisation
The stray capacitance has two effects. First it is responsible for the parallel resonance
already mentioned, and secondly it causes a degradation of the response on the low
frequency side of the series resonance peak. Neutralisation aims to remove or modify
these effects.
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The diagram shows a neutralising circuit which is slightly easier to understand than the
one in the Practical, although the principle is the same.
Using Thevenin's Theorem, the circuit may be replaced by an equivalent circuit consisting
of a voltage source, providing the opencircuit voltage at the two terminals, and an
impedance equal to the impedance of the original circuit when all voltage sources are
removed from it.
If the transformer provides equal and opposite voltages applied to C
s
and C
n
, and if these
capacitances are equal, they form a balanced bridge, providing no contribution to the
output voltage. The voltage is therefore simply the original signal voltage.
The impedance now is that of the crystal in parallel with 2C
s
.
The circuit is therefore equivalent to the one above on the right.
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5.2.5 Practical 1: L/C Tuned Circuit
This Practical examines the frequency response of a tuned circuit. A sinusoidal oscillator
is used to generate a singlefrequency signal. You will be able to change the value of this
frequency and observe the effect of the tuned circuit on the output.
Initially the oscilloscope is connected to the signal source. Use it to check that the
oscillator provides a constant amplitude as the frequency is varied.
The next stage is to connect the oscilloscope to the output terminal and observe how the
output signal amplitude varies with frequency.
What you should expect to see is a large response when the oscillator frequency and the
resonant frequency of the tuned circuit match one another. When the frequencies do not
match the response will be much smaller.
Both the oscillator frequency and the tuned circuit's resonant frequency can be varied. As
you will see, the effect of altering either of the frequency controls is very similar; it is
mainly the amount of mistuning which affects how much the signal is reduced.
5.2.5.1 Procedure
Switch Sweep OFF.
Check that the RF oscillator delivers a signal of roughly constant amplitude, with variable
frequency. Compare the way the signal at monitor point 10 is affected by changes in
frequency. Observe also how the Capacitance control of the tuned circuit alters the
frequency at which peak output is obtained.
The selectivity of the circuit is altered by applying damping. Click on the Damping Button
to switch in the damping load resistor.
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5.2.5.2 Questions
1. Describe how the response of the tuned circuit varies with frequency.
2. How is this response altered when the 'Frequency' control in the tuned circuit is
varied?
3. When damping is introduced the resonant frequency also shifts much more than
theory would predict. Can you account for this? (Hint: The component which alters the
tuning is NOT shown on the circuit diagram.)
4. When damping is introduced is there more reduction of the peak response, or of the
mistuned response?
5. Is the selectivity increased or decreased by damping?
The output peaks at one frequency.
The peak frequency varies.
The stray capacity of the damping circuit.
Of the peak response.
Decreased.
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5.2.6 Practical 2: L/C Tuned Circuit, Swept Frequency
In this Practical the process of observing the frequency response will be automated.
Instead of controlling the signal frequency manually, you will only need to set up the
resonant frequency of the tuned circuit and the range of frequencies to be swept.
If you have not used the frequency sweeper and plotter before, you should read the
Swept Frequency Background.
You may need to take some care over the RF level setting. If the RF signal is too small, it
will not drive the frequency meter properly. If it is too large, the oscilloscope and plotter
will be overdriven.
5.2.6.1 Procedure
Switch the Sweep OFF.
Set the RF oscillator to approximately 455 kHz. Tune the tuned circuit for maximum
response, reducing the RF signal to avoid clipping on the oscilloscope.
Switch the Sweep ON.
Set the minimum sweep frequency to 440 kHz and the maximum to 470 kHz, and finally
select the Plotter.
Use the large plotter to see the frequency response in detail.
To observe how the response is changed when damping is applied, switch in the damping
load resistor by clicking on the Damping Button.
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5.2.6.2 Questions
1. Using the large plotter, what do you estimate to be the bandwidth of the circuit at a
level 3 dB down from that at resonance?
2. What is the 3 dB bandwidth with added damping?
3. What values of Q apply, with and without the added damping?
4. If C is the circuit capacitance before damping is added, what (in terms of C) is the
stray capacitance associated with the damping resistor and its leads?
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5.2.7 Practical 3: L/C Tuned Circuit, Transient Response
In this Practical the driving signal will be set to a low frequency square wave. This will give
the tuned circuit a periodic kick which should set it oscillating.
This behaviour is called free oscillation. (The use of a sinewave to drive the circuit at
the resonant frequency is called forced oscillation). Free oscillation is like the ringing of a
bell after it has been struck by a hammer.
There are two important characteristics of a free oscillation:
1. The frequency.
2. The rate at which it decays, or dies away.
The frequency is exclusively determined by the resonant circuit; in this case by the
inductance L and the capacitance C (including any stray capacitances in parallel with it).
The rate of decay is expressed in various ways, such as the logarithmic decrement or the
damping factor. It depends on the fraction of the energy stored in the reactances which is
dissipated per cycle of oscillation. Anything in the circuit which causes energy loss
contributes to the damping or decay of oscillations. In the Practical a resistor is provided
which you can switch into circuit to see this effect.
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5.2.7.1 Procedure
Switch the Sweep OFF. Switch the Audio Oscillator to square wave. Turn the main
Frequency control fully counterclockwise.
Observe the step signal which is used to drive the tuned circuit, then look at monitor point
10 to see the tuned circuit 'ringing'. (The large oscilloscope will give a better display).
Increase the frequency of the square wave until the transient starts to be affected by the
dying ripples of the previous one.
Use the large oscilloscope to see in detail how the amplitude is affected by the phase of
the free oscillation at the time the step arrives.
Observe how the response is changed when damping is applied by switching in the
damping load resistor by clicking on the Damping Button.
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5.2.7.2 Questions
1. How does the frequency of the 'ringing' or free oscillation compare with the frequency
of peak response in Practical 2 with the same control settings?
2. What factors affect the frequency of the oscillation?
3. What is the principal effect of damping on the oscillation?
4. In a frequency multiplier circuit the signal which drives the resonant circuit is usually a
series of sharp, narrow pulses. Can you suggest why this is more efficient than square
wave drive?
5. In the frequency multiplier, why would the square wave be especially unsatisfactory
when the multiplying factor is even?
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5.2.8 Practical 4: Crystal Filter
Crystal filters are used when a very narrow band of frequencies are required to be filtered;
ie, for high selectivity. You will therefore find that great care is needed in tuning the signal
source if you are to see the response of the crystal.
5.2.8.1 The Notch in the Response
Stray capacitance in the circuit gives rise to a secondary resonance. You can observe this
by looking at the output with the oscilloscope. When the frequency reaches this
secondary resonance, the 'notch' will be seen as the complete disappearance of the
output signal.
5.2.8.2 Neutralising
Neutralising is an arrangement for cancelling this parallel resonance by using a similar
small capacitance to provide a signal in antiphase. This is switched in using the
Conditions box. When it has been switched in, you can adjust the neutralising capacitor to
see how the parallel resonant frequency is shifted. It can even shift to the other (lower)
side of the series resonance.
5.2.8.3 Procedure
Switch Sweep OFF. Check that the RF input signal is constant with frequency, then
select monitor point 8 . Set the oscillator's fine frequency control to its midposition and
adjust the oscillator frequency to 455 kHz. Use the fine frequency control to explore the
narrow band of frequencies around 455 kHz. Note particularly any frequency at which the
output signal becomes zero, as well as the peak response frequency.
Click on the Neutralising Button to switch in the neutralising capacitor. Adjust its
capacitance with the crystal capacitance control, noting the effects on the sharpness
and the symmetry of the response peak.
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5.2.8.4 Questions
1. Compare the selectivity of the crystal filter with that of the LC tuned circuit.
2. On which side of the peak response is the notch (zero response)?
3. With maximum neutralising capacitance switched in, what happens to the notch
position?
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5.2.9 Practical 5: Crystal Filter, Swept Frequency
This Practical is the same as the Crystal Filter Practical, with the addition that you can plot
the frequency response using the Swept Frequency plotter.
Use the plotter to examine carefully the shape of the response curve with the neutralising
capacitor in circuit and set:
1. To minimum capacitance.
2. To maximum capacitance.
3. To provide a symmetrical response curve.
5.2.9.1 Procedure
Switch Sweep ON. Adjust the minimum frequency of the sweep to 450 kHz and the
maximum frequency to 460 kHz. Select the Plotter. Note the asymmetric shape of the
response curve.
Click on the Neutralising Button to switch in the neutralising capacitor. Adjust its
capacitance using the crystal capacitance control, noting the effects on the sharpness
and the symmetry of the response peak.
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5.2.9.2 Questions
1. Would a crystal filter be useful in a receiver for highfidelity music reception? Explain
your answer.
2. In what circumstances would you find it useful to be able to vary the tuning of the
notch (ie, to alter the frequency of the parallel resonance)?
3. If a receiver with a crystal filter is trying to receive a signal in the midst of a general
spread of background noise, how should the neutralisation of the filter be adjusted?
Explain why.
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5.3 RF BandPass Filters Assignment
5.3.1 Objectives
On completion of this assignment you will be familiar with:
The effect of coupling on the combined response of two tuned circuits,
The response of a typical bandpass ceramic filter.
5.3.2 Practicals
Practical exercises are provided as follows:
Practical 1: Coupled L/C Circuits
Practical 2: Coupled L/C Circuits, Swept Frequency
Practical 3: Ceramic Filter
Practical 4: Ceramic Filter, Swept Frequency
5.3.3 Workboard Required
Tuned Circuits and Filters Workboard 53120 which comprises the following blocks:
Signal Generator,
Low Pass Filter,
Band Pass Filter,
Amplifier with AGC.
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5.3.4 Theory
5.3.4.1 Bandpass RF Circuits
The RF Selectivity Assignment dealt with single resonators  an L/C circuit and a piezo
electric device, the crystal. Each of these tended to select a single frequency for
transmission in preference to all others.
If extreme selectivity is required  if one wants effectively to exclude all frequencies except
one  then several L/C circuits may be used. Often several amplifier stages, each having
a tuned circuit, are used together to increase the selectivity.
What is often wanted is to transmit a band of frequencies, excluding signals outside the
wanted band. In these cases a bandpass filter is required.
This assignment is about two forms of RF bandpass filter.
5.3.4.1.1 Coupled LC Tuned Circuits
Several tuned circuits may be tuned to the same frequency and coupled together
reactively in various ways. Some examples are shown above.
If the coupling is loose (ie, if M or C
1
are small, or if C
2
is large), then the frequency
response is very similar to that of uncoupled circuits such as those in separate amplifier
stages.
If the coupling is increased, however, more energy can flow back and forth between the
two circuits. Each circuit then alters the resonance of the other.
The result of this is that the frequency response 'spreads'. Initially, as the coupling
increases the resonant peak simply broadens. With tighter coupling the peak splits into
two separate peaks, either side of the uncoupled resonant frequency.
5.3.4.1.2 Ceramic Filters
To get a wide passband with a sharp cutoff of frequencies outside the passband, a
large number of reactive elements is necessary. Complex filters made of inductors and
capacitors have limited performance because of losses in the inductors. Increasingly use
is made of filters based on mechanical resonances, together with the piezoelectric effect.
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The quartz crystal is a singleresonance example. But for multiple coupled resonances
the ceramic resonator is used. A plate of ceramic material has electrodes applied to it so
that, like the quartz crystal, its mechanical behaviour can interact with an electric circuit.
With a ceramic resonator quite complex resonances and couplings between resonances,
can be set up in the ceramic plate by using electrodes of complex shapes. Fine tuning of
these resonance effects is possible during manufacture by grinding the plate and/or
electrodes. A cheap and highly effective bandpass filter can result.
5.3.4.2 Coupled Tuned Circuits
For this network of impedances:
Z
in
= [Z
1
(Z
2
+ Z
3
)] / (Z
1
+ Z
2
+ Z
3
)
output = i Z
in
Z
3
/ (Z
1
+ Z
3
)
= i (Z
1
Z
3
) / (Z
1
+ 2 Z
2
+ Z
3
)
so that if Z
1
= Z
3
= Z, say, then:
output = i Z
2
/ 2(Z + Z
2
).
With two identical tuned circuits for Z
1
, Z
3
, we may write for each:
Z = (jL + R) / (1 
2
LC + jCR)
and if Z
2
is a coupling capacitor, and multiplying by 2:
2 Z
2
= 2 / jC
c
.
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Substituting these values into the expression for the output, and using routine
manipulation gives the result:
output = numerator / denominator,
where numerator = (jL + R)
2
jC
c
and denominator = [1 
2
LC + jCR] [1 
2
L(C + C
c
) + j (C + C
c
)R].
The most obvious thing about this result is that the denominator has two factors, each
similar to the denominator in the response of a single tuned circuit. What is more striking
is that although the two tuned circuits are identical, the resonances are not  they
are at different frequencies.
This does not necessarily mean that there are two peaks in the frequency response
curve. If the two resonant frequencies are close enough, the two peaks merge. The
mathematics to show this is quite complicated, and is given in the Critical coupling
Theory.
5.3.4.3 Critical Coupling
We found in the Coupled tuned circuits Theory that for two coupled tuned circuits :
output = numerator / denominator
where numerator = (jL + R)
2
jC
c
denominator = [1 
2
LC' + jC'R] [1 
2
L(C' + C
c
) + j (C' + C
c
)R]
and the original C, discussed in the Coupled Tuned Circuit theory, has here been
replaced by C'.
It will be convenient to change the variables to make the expression simpler and more
symmetrical.
Let a new C in this critically coupled circuit be now set equal to C'  C
c
. This does not
affect the numerator. The denominator can be written as
denominator = [1 
2
L(C  C
c
) + jR(C  C
c
)] [1 
2
L(C + C
c
) + jR(C + C
c
)]
Let us now define:
o
... where
o
2
LC = 1
y = (
o
/)  (/
o
) ... the fractional mistuning, which is approximately
y = 2(
o
 ) /
o
... for small mistunings
k = C
c
/ C ... coupling factor
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d = R /
o
L ... the reciprocal of Q, the quality factor
Straightforward manipulation then leads to the relationship:
den (
o
/)
2
= [y + k(/
o
) + jd(1  k)] [y  k(/
o
) + jd(1 + k)]
which is of the form (A+B)(AB) = A
2
B
2
, where:
A = [ y + jd ]
B = [ k(/
o
)  jdk ] = k [ (/
o
)  jd ]
so it becomes:
den (
o
/ )
2
= (y + jd)
2
 k
2
[( /
o
)  jd]
2
At this point some approximations have to be made to simplify this last result of:
den (
o
/)
2
= (y + jd)
2
 k
2
[(/
o
)  jd]
2
In any normal tuned circuit d will be small, and its square can be neglected in comparison
with 1. We shall see shortly that k will be chosen to be of the same order of magnitude.
If the mistuning is small, /
o
approximates to 1.
If the mistuning is large then y
2
becomes much larger than k
2
, so again no great error will
arise from setting /
0
equal to 1.
With these approximations, we have:
den (
o
/ )
2
= (y + jd)
2
 k
2
... (1)
The principal factor affecting the response is the righthand side of that equation:
(y + jd)
2
 k
2
or
y
2
 k
2
 d
2
+ 2jyd
Its magnitude is the square root of:
(y
2
 k
2
 d
2
)
2
+ 4y
2
d
2
= y
4
+ k
4
+ d
4
 2k
2
y
2
 2d
2
y
2
+ 2k
2
d
2
+4y
2
d
2
= y
4
+ 2y
2
( d
2
 k
2
) + k
4
+ d
4
+ 2k
2
d
2
= y
4
+ 2y
2
( d
2
 k
2
) + k
4
+ d
4
+ 2k
2
d
2
To find maxima and minima, differentiate this expression with respect to y and equate to
0.
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Thus:
4y
3
+ 4y( d
2
 k
2
) = 0.
A first solution is y=0. There are two further solutions given by:
y
2
= k
2
 d
2
which are real if k > d; ie, if k > (1/Q).
It can be shown by a further differentiation that the solution y=0 is a maximum if k < (1/Q).
Otherwise it is a minimum and the further two solutions are maxima.
The condition when k = 1/Q is called Critical Coupling.
It will be useful next to find out how the maximum and minimum values vary as the
coupling factor k is changed.
5.3.4.3.1 Maximum and Minimum Values
Considering magnitudes, in the numerator (jL + R)
2
jC
c
the R will contribute very little,
since (L/R)
2
is approximately Q
2
, typically a very large number.
Remembering that
o
2
LC=1:
InumeratorI = (L)
2
kC / (
o
2
LC)
=
3
L k /
0
2
to a close approximation.
Recalling equation (1):
den (
o
/)
2
= (y + jd)
2
 k
2
... (1)
then, for y
2
= k
2
 d
2
:
den
2
(
o
/)
4
= (k
2
 d
2
)
2
2(k
2
 d
2
)
2
+ (k
2
+ d
2
)
2
which reduces to 4k
2
d
2
:
So:
den = 2 k d ( /
o
)
2
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The amplitude response at the peaks is therefore:
num / den =
3
L k /
o
2
/ 2 k d (/
o
)
2
= L / 2 d
= Q L / 2
since Q = 1/k. Note that this expression is independent of k.
for y=0, a similar process yields the result:
Lk / (d
2
+ k
2
)
Remembering that d is 1/Q, this can be expressed as:
LQ (kQ) / [(1 + (kQ)
2
] ,
where kQ is the ratio of coupling to critical coupling.
This expression reaches a maximum for kQ=1.
The factor w appears in all these results. However, over the range say 440 to 470 kHz (as
is relevant to the Practicals Capacitively coupled L/C circuits and Coupled L/C circuits with
swept frequency, only varies by about 7%.
5.3.4.4 Ceramic Filters
Ceramic filters in general are too complex to be analysed here. As explained in the
Ceramic Filter Background, they are comprised of several coupled resonators.
Their general theoretical background is therefore related:
1. via the piezoelectric effect to that of the crystal filter covered in the RF Selectivity
Assignment.
2. as a system of coupled resonators, to the simpler case of two coupled LC circuits
covered in the Practicals Capacitively coupled L/C circuits and Coupled L/C circuits
with swept frequency of this assignment.
3. as a system of reactors requiring a resistive termination for correct performance, to
the treatment of the passive lowpass filter covered in the Audio LowPass Filters
Assignment.
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5.3.5 Practical 1: Coupled L/C Circuits
The form of coupling used in this Practical is 'topcapacity coupling'. Controls are
provided for you to adjust the tuning of one of the L/C circuits and the magnitude of the
coupling capacitance. You can also adjust the frequency of the RF oscillator.
Starting with the coupling capacitance at minimum, you will need to ensure that the
frequencies of the two L/C circuits and the RF oscillator all match, in order to set the
circuit up properly. Maximum signal output will then be achieved.
Since only one of the L/C circuits is adjustable, you will have to adjust it and the RF
oscillator together to match them to the other, fixedtuned L/C circuit.
During the experiment you should:
1. Vary the RF oscillator frequency to explore the frequency response.
2. Do this again after altering the coupling between the two L/C circuits.
IMPORTANT !
Once the circuit is correctly adjusted in this way, you must be careful not
to alter the tuning of the L/C circuit until you have finished the
Capacitively coupled L/C circuits Practical and started the Coupled L/C
circuits with swept frequency Practical.
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5.3.5.1 Procedure
Set the Sweep switch to OFF. Check that the RF oscillator delivers a signal of roughly
constant amplitude with variable frequency. Set the coupling control fully counter
clockwise and then change to monitor point 11.
Adjust the oscillator's and the filter's frequency controls until a maximum response is
attained at point 11. Note the frequency of this maximum.
Set the coupling control fully counterclockwise and explore the response of the circuit
with variation of the oscillator frequency. Notice how two peak responses appear as the
coupling is increased.
DO NOT DISTURB THE FILTER'S
FREQUENCY CONTROL AFTER THIS.
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5.3.5.2 Questions
1. Why were you asked to adjust both the RF oscillator and the first tuned circuit for
maximum response?
2. Why was minimum coupling specified while doing this adjustment?
3. What was that frequency after you had done the adjustment?
4. With maximum coupling, what were the frequencies of the two peaks?
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5.3.6 Practical 2: Coupled L/C Circuits, Swept Frequency
This Practical should only be attempted after Practical: Capacitively coupled L/C circuits
has been done. This is because it is important to get the two tuned circuits tuned to the
same frequency.
Practical: Capacitively coupled L/C circuits achieves this by getting :
1. The RF oscillator tuned to the fixedtuned circuit.
2. The variablytuned circuit tuned to the oscillator.
Practical: Coupled L/C circuits with swept frequency will apply the sweptfrequency
technique to plot the response curves of the filter.
With the two L/C circuits correctly tuned, the remaining variable is the coupling between
them. This should initially be set to minimum. The resulting response will look like that of
the single tuned circuit, but with steeper sides, showing more selectivity.
As the coupling is increased, the peak will split into two peaks, giving a band of
frequencies for which the signal is transmitted freely. The condition just before this split
occurs is called 'critical coupling'.
In practice the ideal response is flattopped. A good approximation to this is got by using
overcoupled and critically coupled circuit pairs together, so that the criticallycoupled
hump fills in the gap between two separated peaks.
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5.3.6.1 Procedure
The frequency control in the bandpass filter should be set as it was in the Capacitively
coupled L/C circuits Practical. If it is not, repeat practical before continuing.
Set the coupling control fully counterclockwise and switch the Sweep ON.
Set the minimum sweep frequency to 445 kHz and the maximum to 465 kHz.
Finally, select the Plotter.
You can superimpose responses for minimum and maximum coupling simply by changing
the coupling quickly while the plotter is operating. If the two peaks are uneven, SLIGHT
adjustment of the filter frequency control may help.
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5.3.6.2 Questions
1. Overcoupling pushes the sides of the response curve apart. Does it affect the slope
of them much?
2. When the circuits are overcoupled, what is the principal effect of altering the resonant
frequency of one of the tuned circuits?
3. From it can be found that the response at the centre frequency when overcoupled is
down from the peak response by approximately 2 (kQ) / [1 + (kQ)
2
] where kQ is the
ratio of coupling factor to critical coupling. What is the maximum value of kQ you can
obtain with the equipment?
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5.3.7 Practical 3: Ceramic Filter
As was mentioned in the Coupled L/C circuits with swept frequency Practical, to get a
good flattopped response with steep sides requires several resonators, several pairs of
LC circuits being a possibility. This Practical is concerned with an alternative solution to
the problem of providing several resonators.
Just as the quartz crystal uses the piezoelectric effect to translate a mechanical
resonance into an electrical one, so a device which has several coupled mechanical
resonances can, through piezoelectricity, provide a system of several coupled electrical
resonances. The combination of a piezoelectric ceramic plate with suitably shaped
electrodes can provide this.
The electrodes typically have several prongs or fingers, and the size and shape of each
one determines one of the resonances in the system. Their closeness to each other
determines the coupling. The tuning of the various resonances is done by grinding the
electrodes or substrate ceramic during factory processing.
Consequently, there is little you can do to adjust the circuit.
You will find that, as for the lowpass passive filter Assignment, matching of the
terminations is important. Now that you have seen the effect of damping on a resonant (L
C) circuit, you should be able to appreciate that the matching load resistances do provide
necessary damping on resonances which would otherwise give undesirable peaks in the
response near cutoff.
5.3.7.1 Procedure
Switch the Sweep OFF.
The RF oscillator delivers a signal of roughly constant amplitude, with variable frequency,
as in the Capacitively coupled L/C circuits Practical.
Compare the way the signal at monitor point 9 is affected by changes in frequency. Make
a note of two frequencies between which the interesting part of the filter characteristic lies.
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5.3.7.2 Questions
1. What do you think is the most important difference between the ceramic filter and the
L/C bandpass filter of Practicals 1 and 2?
2. Over what range does the phase of the output signal vary between the two cutoff
frequencies? A large phase change with frequency implies an appreciable delay
(strictly, group delay) in the signal. This is typical of complex filters, having large
numbers of reactive elements.
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5.3.8 Practical 4: Ceramic Filter, Swept Frequency
This Practical continues the Ceramic filter Practical by adding the automatic response
plotter.
5.3.8.1 Procedure
Switch the Sweep ON.
Set the minimum sweep frequency to 445 kHz and the maximum to 465 kHz.
Select the Plotter. Use the large display to see the response in more detail.
The behaviour of the circuit is modified greatly when the correctly matching load or source
impedance is not provided. Switch the load off by pressing the Load Button and observe
its effect.
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5.3.8.2 Questions
1. Describe briefly the effects of removing the filter's load on:
a) the shape of the frequency response.
b) the max output amplitude.
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5.4 Tuned Amplifier with Gain Control Assignment
This assignment deals with the control of amplifier gain, both manual and automatic.
5.4.1 Objectives
On completion of this assignment you will be familiar with:
The gain control characteristic of the amplifier stage (under manual gain control),
How automatic gain control can maintain a steady level of output signal,
Why automatic gain control can make difficulties during frequencyresponse
testing.
5.4.2 Practicals
Practical exercises are provided as follows:
Practical 1: Gain Control
Practical 2: Automatic Gain Control
Practical 3: Frequency Response with Automatic Gain Control
Practical 4: Decibel Gain
5.4.3 Workboard Required
Tuned Circuits and Filters Workboard 53120 which comprises the following blocks:
Signal Generator,
Low Pass Filter,
Band Pass Filter,
Amplifier with AGC.
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5.4.4 Theory
5.4.4.1 Tuned Amplifiers
Radio frequency (RF) amplifiers are usually tuned. That is, they include devices which
select a particular frequency or band of frequencies. This is necessary for two main
reasons:
1. The user is normally interested in only one at a time of the enormous number of
radio signals which can be received. The others have to be rejected as unwanted
interference. This aspect is dealt with in the RF Selectivity Assignment.
2. At high frequencies the reactance of stray capacitances becomes small. This
makes it difficult and expensive to achieve large amplifier gains, unless inductive
reactance is used to cancel out the effects of this capacitance. Where a fairly
narrow band of frequencies is acceptable (and often is required), a parallel
combination of inductance and capacitance resonates at a particular frequency to
provide high impedance.
3. The resonant frequency must match the frequency of the signal to be received.
Achieving this match is called 'tuning'.
5.4.4.1.1 Amplifier Gain Control
The amplitude of signals, especially RF signals, can vary over an enormous range. The
signal arriving at an aerial (antenna) terminal may well be a microvolt or so. But if a
transmitter is near by, that signal may rise to several volts amplitude.
The signal required at the detector is typically in the order of 0.2 V to 10 V. To achieve
this, one or more amplifiers process the aerial signal on its way to the detector. To
achieve say 1 V at the detector, with input signals ranging from 1 microvolt to 1 V the gain
has to be varied in the ratio one million to 1.
The control of the gain of an amplifier is therefore very important.
A typical RF amplifier in a radio or radar receiver is therefore tuned and has a very wide
range of gain which can be controlled. This usually requires several amplifying stages to
be gaincontrolled; this assignment will look at just one such stage.
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5.4.4.2 Decibels
The range of signal strengths in communications work is enormous. Signals may vary
from a fraction of a microvolt to many volts, a ratio of many millions to one. In any given
impedance the corresponding power ratio is trillions to one.
In order to reduce the numbers used to manageable proportions, a logarithmic scale is
convenient. This is because the common logarithm of say 1000000 is only 6 and the
logarithm of 1/1000000 is 6. So the range of logarithms from 6 to +6 covers a ratio of ten
to the power 12; ie, 1000000000000.
5.4.4.2.1 Decibel Power Ratio
The bel is a unit of power ratio, simply defined so that the power gain, defined by the
ratio:
G = power
out
/ power
in
may be expressed as log
10
G bel.
In practice the numbers arrived at tend to be rather small, so the unit generally used is the
decibel (dB). The same power ratio G can therefore be expressed as 10 log
10
G dB.
5.4.4.2.2 Decibel Voltage Ratio
In many communication systems a signal travels along a system such as a line which has
a characteristic impedance; any apparatus connected is matched to this same
impedance. In these circumstances (constant impedance), the volt ratio between two
points is simply the square root of the power ratio between them.
It follows that if g
v
is the voltage ratio, then:
G = (g
v
)
2
and the power gain is:
10 log
10
G dB = 10 log
10
(g
v
)
2
dB
= 20 log
10
g
v
dB.
If the resistive impedance at an input point is R
i
, the voltage is v
i
, at the output it is R
o
, the
voltage is v
o
and the power gain is G, then:
G = ( v
i
2
/ R
i
) / ( v
o
2
/ R
o
)
= ( v
i
/ v
o
)
2
/ ( R
o
/ R
i
)
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Electronic engineers on the other hand use circuits in which impedances vary widely.
Voltage is easier to measure than power, so that, much of the time, they think more in
terms of voltage gain than power gain. But the huge ratios of signal strength still have to
be managed.
As a result, the habit has grown up of speaking of voltage ratios expressed in terms of dB,
but WITHOUT considering the impedance ratio. The 'voltage gain in dB' is taken as the
value NOT including the impedance ratio. Thus:
'voltage gain in dB' = 20 log
10
g
v
dB.
It is important to understand when this convention is being used and when a true dB value
based on power is required.
5.4.5 Practical 1: Gain Control
The gain of an RF amplifier is almost invariably controlled by adjusting a dc control
voltage. This is because problems arise with any attempt to use a potentiometer or
equivalent device in the manner of an audio gain control. Some of these problems are:
1. Ineffective control because stray capacitances provide leakage paths.
2. Unwanted alteration to the tuning.
3. Unwanted changes in selectivity.
This assignment will examine some of the ideas associated with that of 'gain' and use a
dc control voltage to adjust the gain in an amplifier. A manual control will enable you to
adjust the dc level which controls the gain. Measurements with the oscilloscope will
enable you to measure the gain for various dc settings.
Output v. control voltage at constant input
You will need graph paper scaled as shown.
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5.4.5.1 Procedure
Switch Sweep OFF.
Set the gain control of the tuned Amplifier fully clockwise.
Adjust the output of the RF oscillator to a suitable level and then adjust its frequency for
maximum response from monitor point 6 .
Observe the oscilloscope and voltmeter while adjusting the gain control.
Use the large oscilloscope to measure the peaktopeak output.
Keeping the RF signal constant, set different values of dc voltage at about 0.1V intervals,
using the gain control. Plot a graph of output signal against the dc voltage.
Use both monitor points 6 and 7 to extend the range of measurements.
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5.4.5.2 Questions
1. What is the input signal voltage, measured at monitor point 4?
2. What is the maximum voltage gain from point 4 to point 6, assuming the onefifth
attenuation from point 6 to point 7 is correct? Express your answer:
a) as a ratio
b) in dB.
3. What is the minimum voltage gain from point 4 to point 6, expressed:
a) as a ratio?
b) in dB?
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5.4.6 Practical 2: Automatic Gain Control
To cope with signals which vary between the two input levels shown in the diagram would
be extremely tiresome if the gain had to be adjusted manually each time a different signal
were received. Also radio signals quite often vary in strength. This may be due to changes
in propagation conditions, motion of the transmitter or receiver or other causes.
Automatic gain control is intended to adjust the gain automatically so that the desired
constant signal level is achieved at the detector input despite variations in the input
signal.
In order to achieve this, a measure of the signal amplitude at the detector is required, in
the form of a dc voltage. An envelope detector has an output containing dc proportional to
the RF amplitude, together with ac corresponding to any amplitude modulation. The latter
is removed by a filter.
The dc voltage is compared against some reference value and, when it exceeds the
reference value, the difference signal is used to decrease the gain.
The reference voltage is often referred to as the 'AGC delay voltage'. The term 'delay'
does not refer to a time delay. The early AGC systems simply fed back the dc component
from the envelope signal detector to control the gain. This meant that even for undesirably
weak signals, some gain reduction took place. 'Delaying' the start of the feedback as the
signal level rose avoided this problem.
In the circuit diagram of the Practical, the dc signals associated with AGC are distinctively
coloured. You will see the envelope detector which generates the AGC signal is back
biassed by about 0.V. This is the delay voltage and the reference against which the signal
level is compared.
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Output/Input showing effect of AGC.
You will need graph paper scaled as shown.
5.4.6.1 Procedure
The sweep should be switched off.
Set the RF level to maximum and the gain of the tuned Amplifier fully clockwise.
Tune the RF oscillator for maximum response from monitor point 7. Then observe how
the output varies with changes of RF oscillator amplitude. Repeat using monitor point 6.
Use button to switch in automatic gain control and repeat the observation.
Use the large oscilloscope to measure peaktopeak voltages.
Set different amplitudes of RF signal, measured at monitor point 4, and measure the
corresponding output at monitor point 6. Plot a graph of output signal against RF input
voltage.
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5.4.6.2 Questions
1. What causes lack of proportionality between output amplitude and input amplitude:
a) with manual gain control?
b) with automatic gain control?
2. How does the gain control voltage vary as the RF input is increased? Explain why:
a) in relation to the required gain adjustment.
b) in relation to how the control voltage is produced.
3. Can you see a connection between the amplitude at which the output RF is
automatically controlled and the 0.7 V marked on the circuit diagram?
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5.4.7 Practical 3: Frequency Response with Automatic Gain Control
Suppose that you are plotting the frequency response of an amplifier which includes AGC.
A constant input voltage is applied at various frequencies and you hope to see the output
vary in a way that shows the frequency response. What happens in fact?
As the output tries to change, the action of the AGC system tries to prevent this variation.
If it is a good AGC system it will virtually prevent any variation at all, once the output has
reached a certain level.
Consequently, when taking a frequency response, the AGC must be prevented from
working. The Practical will show how important this is.
5.4.7.1 Procedure
Switch the sweep ON. Set max RF oscillator amplitude. Set the gain control of the tuned
Amplifier fully clockwise. Adjust the minimum frequency of the sweep to 445 kHz and the
maximum to 465 kHz.
Change to monitor point 6 and adjust the RF oscillator level so that a full height trace
appears on the small oscilloscope at the peak frequency.
Select Plot Response and observe the resulting humped curve showing the selectivity of
the tuned circuit.
Use the Button to switch to Auto gain control. Observe the response now displayed.
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5.4.7.2 Questions
1. Why does the frequency response appear to become flat when AGC is switched in?
2. If the input signal were made small enough, would it be possible to plot the frequency
response? (Explain your answer).
3. Suppose that the amplifier is operating with an input amplitude such that the AGC
circuit holds a constant output amplitude. Does this mean that a signal on another
frequency will not be relatively attenuated by the tuned circuit, as it would be with no
AGC present? (Hint: Consider carefully whether the AGC control voltage acts on the
active device in the amplifier or on the tuned circuit.)
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5.4.8 Practical 4: Decibel Gain
This Practical introduces the concept of using decibels to express the gain of an amplifier.
In it you are asked to make some measurements and then translate the results into
decibels. Read the theory section on how decibels work before attempting this practical
and answering the questions.
5.4.8.1 Procedure
Switch sweep OFF. Set the amplifier gain to maximum. Tune the RF oscillator for
maximum output, adjusting the oscillator output to avoid overloading the oscilloscope.
Change to the large oscilloscope and adjust the RF amplitude so that a full height trace
appears.
Measure the voltages at points 4 and 6. Answer questions 1 to 3.
Select Automatic Gain Control before answering question 4.
Chapter 5
ANALOGUE COMMUNICATIONS  Assignments using the
STUDENTS WORKBOOK Tuned Circuits and Filters Workboard
564 53001S
5.4.8.2 Questions
1. Take the input power to the amplifier to be that appearing in the 1.8k resistor. The
output power appears in the 56 k load. The voltage across the load is the same as at
monitor point 6. What is the power gain, expressed as a ratio?
2. What is it in dB?
3. Using dB as a unit related only to voltage (no account taken of impedance levels),
what is the voltage gain in dB?
4. Complete the Practical before answering the next question. Over what range, in dB,
can the input vary without significant change in the output amplitude?