THE

ACTICAL
HI-Fl
NDBOOK
Gordon J. King
Assoc. Brit. I.R., M.I.P.R.E., M.T.S.
THE PRACTICAL
HI-FI
HANDBOOK
BY

GORDON J. KING
Assoc. Brit. I.R.E., M.I.P.R.E., M.T.S.

THE MACMILLAN COMPANY

NEW YORK
1960
 First published 1959
First published in the United States of America 1959
Reprinted 1959, 1960

© Gordon J. King, 1959

MADE AND PRINTED IN GREAT BRITAIN BY

TAYLOR GARNETT EVANS & CO. LTD,,
WATFORD, HERTFORDSHIRE
 Contents

CHAPTER PAGE

FOREWORD 7
HI-FI FUNDAMENTALS 9
2 VOLTAGE AMPLIFIERS, FEEDBACK AND CONTROL
CIRCUITS 27
3 THE POWER AMPLIFIER 51
4 TRACING AND CLEARING FAULTS IN AMPLIFIERS 73
5 LOUDSPEAKERS AND ENCLOSURES 102
6 DISK RECORDING 127
7 PICK-UPS AND RECORD PLAYING EQUIPMENT 137
8 MICROPHONES AND MIXERS 160
9 THE USE OF TAPE 174
10 STEREOPHONY 196
INDEX 220
 Foreword

THE field of high-fidelity sound reproduction in all its aspects

is a vast one, and one of the most difficult problems in planning a book of
this kind is that of deciding just how far each branch of the subject can be
explored within the space available. A whole volume could, in fact, be written
on the subject of each chapter, but then a good deal of perspective would be
lost. It has been my aim in the present work not only to give a kind of bird's-
eye view of the subject as a whole, but also to provide the reader with a fair
idea of the operation of the various items of equipment, and knowledge
which will enable him to secure the best results from his own and-if he is
a service technician-his customers' equipment.
The book is not concerned solely with servicing matters, though it
contains much information of a servicing nature which, it is hoped, will
assist the reader who is already a radio service technician, but has only just
entered the high-fidelity field. Such information should prove useful to the
many dealers and their engineers who have recently entered, or intend to
enter, this branch of trading and servicing work. It is hoped that the book
will clearly bring out the difference which exists between accepted standards
in trading in radio and television receivers and the retailing and servicing of
high-fidelity equipment. The enthusiast in this field invariably possesses a
keen understanding of high-fidelity matters, which makes it essential that
the knowledge of the dealer and technician should attain the same standard.
Owing to the very nature of "hi-fi", the enthusiast can only get the best
from his equipment by understanding how the various sections work and how
they are integrated into a perfectly matched system; it is also important to
understand the adjustments which are required to maintain optimum per-
formance. In this respect, the book is directed also to the enthusiastic ama-
teur. It explores in a practical manner all the links in the chain, from the
original sound at the microphone to the reproduced sound at the loud-
speaker. It does not embrace frequency-modulation tuners, however, since
these have been described in detail in my book "F.M. Radio Servicing
Handbook'', issued by the same publishers.
7
 FOREWORD
I have included a full chapter on the subject of stereophony, which will
undoubtedly become of major importance in the future. During the prepara-
tion of this book, stereophonic sound reproduction graduated from tape
to disk, and I have thus been able to include the very latest information on
this development.
Owing to its essentially practical presentation, one or two slight ambigui-
ties may be noticed in the text; it is hoped that these will not be held against
me by the purist, but ignored in the interests of simplicity of description.
My thanks are due to many manufacturers of high-fidelity equipment for
their co-operation in placing so much information at my disposal an<l for
supplying photographs of their equipment. I also wish to record my thanks
to my wife Barbara for her tolerance and encouragement during many late
hours spent in the writing of this book, to my colleague Peter Berry for his
splendid work in preparing certain photographs, and to Mr. N. S. Hyslop,
who has prepared excellent drawings from my very rough sketches.
Finally, I am indebted to Mr. L. C. Holmes who has been faced with the
formidable task of editing not only the MS. of this work, but also my pre-
vious books, "F.M. Radio Servicing Handbook" and "Television Servicing
Handbook".

Oxford, G.J. K.
1959.
 CHAPTER 1

Hi-Fi Fundamentals

LE serv1cmg of high-fidelity ("hi-fl") audio equipment

demands of the service technician and amateur enthusiast a more critical
sense of appraisal of audio reproduction than that required in the case of
ordinary sound equipment, where quantity rather than quality is the domi-
nating factor. While the experimenter-enthusiast will be fully aware of this
higher critical faculty, and will generally possess it, the man whose job it is
to service domestic electronic equipment for a living, taking hi-fl equipment
in his stride along with radio and television receivers, may not have such
a sensitive ear. If he is not closely acquainted with the foibles of the modern
hi-fl enthusiast, the service technician may well be excused his doubts and
irritation on encountering the insistence of such an enthusiast that a
high degree of distortion is being produced by his apparently excellent
amplifier!
Accustomed to a standard of reproduction based on years of servicing
radio sets of considerably limited audio fidelity, the technician may feel that
the enthusiast's request for service of equipment which even in its alleged
faulty condition is capable of reproduction of a high order, is far from
warranted. This problem of differing standards can make life exceedingly
difficult for the service technician when he starts to undertake the repair
of hi-fl equipment. To be really successful at the job, the technician must
himself develop "hi-fl" standards of judgment. This is usually automatic,
as anyone dealing with the servicing of hi-fl equipment works in close liaison
with the enthusiasts who operate it.
A technician new to the field quickly becomes initiated, and quickly
realizes that (for instance) where a close-tolerancl! 50k resistor is stipulated
by the maker, replacement cannot be made satisfactorily with a 47k resistor
of mediocre tolerance, as can often be done in less exacting equipment with
little adverse effect.
Hi-fl equipment does not just happen. It is created in the laboratory by
a large number of small points being given a great deal of attention. The net
result is hi-fi. Slight disturbance to one or more of these small points, either
9
 THE PRACTICAL HI-FI HANDBOOK
as the result of alteration in value of a component or unskilled service, will
unbalance the design and possibly cause distortion. To the uninitiated service
technician the distortion may hardly exist, but to the hi-fi perfectionist a
world of difference will be discernible. The technician will have to use
instruments on which to base his judgment of reproduction; listening tests
waste time and lead to frustration.
Essentially, there are three types of hi-fi enthusiast. First, there is the
music lover who wishes to play his favourite records with the minimum of
distortion. This type is Jess technically exacting, since a reasonable quality
of reproduction is sufficient to re-create in his mind the atmosphere of the
concert hall-slight distortion thus goes unnoticed. Then there is the tech-
nical perfectionist whose observations are keenly focused on the various
responses of the equipment. This type may not possess a highly developed
aesthetic interest in music, but he is able to judge with curious accuracy just
how much harmonic distortion there is, how the equipment is handling
transients, whether or not additional damping would go to improve the
overall results and similar technical matters. He obtains great satisfaction
from listening to sounds of large magnitude with little distortion, and when
he says that distortion is present it is most desirable for the technician to
agree with him-until he can prove otherwise, of course!
Finally, there is the type who is a compromise between the other two-
he represents the large majority of hi-fi enthusiasts, who are enthusiasts
because they are not only technically interested in obtaining distortion-free
reproduction, but are also interested in music in itself.
It is as well for the technician new to the hi-fi world to familiarize
himself with these three types of enthusiast; this is equally as important as
having the technical know-how, for anyone actively engaged in the servicing
of hi-fi equipment will soon become aware that he has to be something of
a psychologist as well as a technician-and it is most desirable to know one's
subject. This is because sound is a subjective thing, and since it is this in
which we are ultimately interested, it is most important to learn a little about
it and its effects before moving on to more objective technical matters.

SENSATION OF HEARING
Sound is the stimulus which when applied to the ear gives rise to the
sensation of hearing. It is not wholly true to consider sound as emanating
from any particular source. Sound is essentially a function of the listener's
ear, nervous system and brain. There would be no sound from an explosion,
for example, occurring in a place without an ear, nervous system and brain
to record it, though there would be considerable air disturbance, to say the
least.
The source of any stimulus producing the sensation of hearing is always
10
 HI-Fl FUNDAMENTALS
in some state of vibration. This can be demonstrated by the piano string,
the tuning-fork or, to keep in line with our present theme, the cone or dia-
phragm of a loudspeaker. The vibration may be so slight and so rapid that
it is not visible, or it may equally be so large and relatively slow as to be
easily observed, as in the case of a loud mains hum affecting the cone of a
loudspeaker. It is of little purpose in trying to alleviate the latter condition
by securing the speech coil of the loudspeaker cone to the magnet pole piece
with good-quality glue-a condition which was once observed by the author
when investigating for lack of signals! (However, when questioned, the owner
was true to principle in remarking, "but I got rid of the terrible hum which
was caused by this cone thing vibrating." A true story!)
In the case of an organ pipe and other wind instruments, the source of
the stimulus is a column of air. This can be realized from the considerable
agitation of fine dry sand on a piece of paper when brought over the mouth
of the pipe. The same effect can be observed by placing the sand-laden paper
over the vent of a vented loudspeaker enclosure when the system is fed with
low-frequency signals to which the vent is tuned, or resonated. In fact, it
represents a good method of discovering the vent resonant frequency-
assuming that an audio generator is at hand to feed a variable audio signal
to the loudspeaker-and the free resonance of the loudspeaker cone. In the
latter case, of course, the sand-covered paper is held over the loudspeaker
cone. The reason for the agitation is that the air is moving in and out of the
pipe or vent rapidly, and so sets the paper vibrating.
In many cases the vibration can be felt by placing a finger on the string
or loudspeaker cone. It is surprising how sensitive the finger can be in this
respect; some engineers check for mains hum by lightly placing the finger on
the loudspeaker cone. Air vibration can also be felt. Standing in front of a
large loudspeaker fully loaded to, say, 10 watts of low-frequency signal
readily illustrates this fact.
Any stimulus of sound (in future we shall refer to it as sound in terms
of both cause and effect) may vary in three ways, that is, infrequency, loudness
and quality or timbre. The number of complete vibrations made by a sound-
producing device in one second is called the frequency and determines the
pitch of the resulting note. As an example, the string corresponding to bottom
A in the piano vibrates at 27· 5 c/s.
The loudness of a note or sound is governed by the amplitude of the
vibration which, of course, determines the energy applied to the ear. The
quality or timbre, which distinguishes between notes of the same pitch
sounded by different instruments, results from the presence of harmonics in
the make-up of the sound. For the present, these can be considered as subsi-
diary vibrations whose frequencies are exact multiples of the fundamental
vibration.
11
 THE PRACTICAL HI-FI HANDBOOK
AUDIBLE FREQUENCY RANGE

As the frequency is reduced, the resulting note eventually becomes

resolved into the separate impulses of which it is composed. As the frequency
is increased, however, the note becomes very shrill, and at about 15,000 c/s
it is little more than a hiss. The high-frequency limit of audibility varies
widely with different individuals. The limit is usually higher with young
people, often extending to the region of 20,000 c/s, while with increase in
age the limit may fall to some 9,000 to 10,000 c/s. Some people are highly
conscious of the 10,000 c/s note produced by television receivers, while
others, usually older people, are not at all disturbed. The high-pitched squeak
of a mouse is often inaudible to people in their fifties, but often very discon-
certing to young people.
At this point it should be made clear that a person who is virtually deaf
above, say, 7,000 c/s is still able fully to appreciate music containing harmonic
components extending well above this figure. It is still necessary for hi-fi
equipment employed by such a person to be capable of reproducing all
frequencies to the limit of the audio spectrum (the frequency range of good-
quality equipment usually extends well beyond the accepted audio range,
for technical reasons which will be explained later). Tests have revealed that
distortion-free reproduction of music containing harmonic components up
to some 18,000 c/s gives the sensation of considerable mutilation, when
passed by way of a filter which chops off all frequencies above 7,000 c/s-
not only to a person whose hearing is unimpaired up to 18,000 c/s, but also
to one who is essentially deaf at 7,000 c/s.
The reason for this, as we shall appreciate better later on, is that a large
part of music is composed of steep, rapidly occurring wavefronts (transients),
produced by harmonic components of the fundamental frequencies of the
various instruments. Cutting the higher frequency components has the effect
of spoiling the desirable steepness of the wavefronts as well as reducing
the overall amplitude of the sound. Since transients are responsible for the
"attack" attributable to music, destroying these in a way that impairs the
corresponding accelerations of the wavefronts is obvious equally to persons
with and without extended frequency range.
There is another important characteristic of the human ear which gives
the impression of dissimilar volume to sounds of different pitch. The sensi-
tivity of the ear rises to a maximum in the region of 3,000 c/s, and falls off
at frequencies above and below this range.
THE DECIBEL AND THE PHON
While the ear is considerably sensitive to small changes in pitch of a
sound it is much less sensitive to changes in amplitude (volume). Instead of
following a linear law, the sensitivity of the ear to changes in volume is
12
 HI-FI FUNDAMENTALS
logarithmic. This simply means that the impression a listener receives when
a sound of certain volume is suddenly increased is proportional to the loga-
rithm of the ratio of the energy or power of the two sound levels. The common
logarithm of the ratio of two powers gives their relationship in be/s. In
mathematical form Nb= log10 (P2/Pl). This holds for a decrease in power
as well as for an increase in power, so that when P2 is less than Pl the value
of Nb becomes negative.
The whole range of hearing corresponds to a change comprising 13
bels, that is, starting at a power or intensity near the threshold of hearing
to a point where the intensity begins to be painful. As I 3 bels is too rough
a scale for ordinary use, each bel is divided into IO decibels (db). Thus, the
difference in level between two powers (Pl and P2) in decibels is given by
N db = IO log10 (Pl/Pl). This expression holds for any change of power,
electrical as well as acoustical. Clearly, the ultimate effect of any change of
electrical power in a hi-fl amplifier, for instance, is to produce a change of
acoustical power from the loudspeaker. It is as well to become familiar with
the decibel, as it crops up frequently in audio work.
As an example, suppose an amplifier delivering l watt into a loudspeaker
is adjusted to promote an increase of I watt. The output is now 2 watts.
Although the effect can be realized from the statement that "the power has
doubled", there is little point in saying that "the power has increased by
I watt" unless, of course, it is first clearly indicated that the original power
was l watt. It is much better to say that "the power has increased by 3 db".
Thus, doubling the power is equal to a 3 db increase (3·01 db, to be precise),
and halving the power is equal to a 3 db decrease. In the latter case it is
usually said that a - 3 db power change has occurred.
A change of 2 db, equal to a power of 3 watts being increased to 4·75
watts or decreased to l ·9 watts, for example, is just about discernible by the
average person, while a change in level of I db is hardly perceptible to the ear.
The decibel is also extensively adopted to compare two currents or
voltages. When used in this way it must be ascertained that the resistances
(R) in which the currents(/) and voltages(£) operate are the same. When this
is the case:
N db= IO log10 (/2 2//1 2) or IO log 10 (£22 / £12), these being equal to 20 log 10
(/2//1) and 20 log 10 (£2/£1).
When the resistances are not equal due allowance has to be made:
N db= 20 log10 (/2//1) + IO log10 (R2/Rl) and
N db= 20 log 10 (£2/£1) + IO log10 (R2/RI).
The decibel, as we have already seen, is essentially a unit for measuring
relative powers, so when it is employed to express current and voltage
gains and losses, allowance has to be made for the fact that power varies by
13
 THE PRACTICAL HI-FI HANDBOOK
TABLE I.I
CONVERSION OF DECIBELS TO POWER AND
VOLTAGEiCURRENT RATIOS

Power Voltage Power Voltage

db db II
I Ratio
I Ratio
I Ratio Ratio

1 1·26 I I ·12 15 I 31 ·6 I
I
5·62
2 1·58 1·26 20 100 I 10
3 2·0 1 ·41 30 1000 31 ·6
I to• JOI
4 2·51 1·58 40
5 3-16 I ·78 50
I
I
10• I 316
6 3·98 2·0 60 10• 10•
7 5·01 2·24 70 107 3160
8 6·31 2·51 80 10• 10•
9 7·94 2·82 90 10• 31600
10 10
i 3·16 100 10•• 10•
i I
the square of the change of current or voltage. For example, an increase in
current or voltage by a factor of two results in the power being increased
by a factor of four.
When N db is known, the power, current and voltage ratio can be
found as follows:
P2/Pl = antilog N db/IO, /2//1 = antilog N db/20 and
£2/El = antilog N db/20.
Decibel tables save the toil of making complex calculations, samples
being given in Table I. I and Table I .2. Table I. I gives conversion of decibels
to power and voltage/current ratios. Figures not given in the table may easily
be calculated. For example, if two db figures are added, their corresponding
power or voltage/current ratios must be multiplied. Table 1.2 gives conver-
sion of power ratios to decibels.
The apparent loudness of any tone is related to its pitch or frequency
as well as to its amplitude or intensity. The phon is the unit of loudness level
actually appreciated by the ear, and represents about the limit of difference
in loudness of which the ear is sensible. At a frequency of 1,000 c/s, the
loudness level of a pure tone in phons is equal to the number of decibels
above the reference power, though this does not hold with any other fre-
quency. It is this apparent non-Iinea,r loudness level over the audio spectrum
which has recently encouraged the use of "loudness" controls on hi-fi ampli-
fiers. As we shall see later, they function essentially to increase the bass
response as the volume is reduced.
14
 HI-FI FUNDAMENTALS
TABLE 1.2
CONVERSION OF POWER RATIOS TO DECIBELS

Power [ Power Power Power I

Ratio db Ratio
I db Ratio I db Ratio db

l ·0 0·000 3·3 5·185 5·6 7·482 7·9 8·976

l ·l 0·414 3·4 5-315 5·7 7·559 8·0 9·031
l ·2 0·792 3·5 5·441 5·8 7·634 8·1 9·085
l ·3 l ·139 3·6 5·563 5·9 7·709 8·2 9·138
1·4 1·461 3·7 5·682 6·0 7·782 8·3 9·191
1·5 l ·761 3·8 5·798 6·1 7·835 8·4 9·243
l ·6 2·041 3·9 5·911 6·2 7·924 8·5 9·294
l ·7 2·304 4·0 6·021 6·3 7·993 8·6 9·345
l ·8 2·553 4·1 6·128 6·4 8-062 8·7 9·395
l ·9 2·788 4·2 6·232 6·5 8·129 8·8 9·445
2·0 3·010 4·3 6·335 6·6 8·195 8·9 9·494
2·1 3·222 4·4 6·435 6·7 8·261 9·0 9·542
i
2·2 3·424 4·5 6·532 6·8 8·325 9·1 9·590
2·3 I 3·617 4·6 6·628 6·9 8·388 9·2 9·638
2·4 3·802 4·7 6·721 7·0 8·451 9·3 9·685
2·5 i 3·979 4·8 6·812 7·1 8·513 9·4 9·731
I
2·6 I 4·150 4·9 6·902 7·2 8·573 9·5 9·111
2·7 4·314 5·0 6·990 7·3 ! 8·633 9·6 9·823
2·8 4·472 5·1 7·076 7·4 I 8·692 9·7 9·868
2·9 4·624 5·2 7·160 1·5 8·751 9·8 9·912
3·0 4·771 5·3 7·243 7·6 8·808 9·9 9·956
3·1 4·914 5·4 7·324 1·1 8·865 10·0 10·000
3·2 5·051 5·5 7·404 7·8 8·921

HARMONICS

Most vibrating bodies execute a simple harmonic motion, giving a pure

tone, or the vibration is composed of a combination of simple harmonic
motions, giving rise to overtones, which are usually related in frequency to
the fundamental. The sine wave (Fig. I.I) is representative of simple har-
monic motion, such as that produced by the vibration of a tuning-fork.
It is the presence of overtones or harmonics which is responsible for the

FIG. I.I. Simple harmonic motion, such as

that produced by a tuning fork, can be
represented by a sine wave.
~lP
~
<(
WAVELENGTH

\/VV
f\ f\

15
 THE PRACTICAL HI-FI HANDBOOK
FIG. 1.2. The two sine waves (a) are related
in that one has twice the frequency of the
other. In (b) the waves are compounded to give
a composite wave. Here the sound is no
longer pure, but has a high second-harmonic
content.

r
(a)

A difference in the quality between the

V
lb>
sounds produced by the various instru-
ments of an orchestra. The human
voice is also rich in harmonics, and since
the harmonic content differs between individuals, it is often a simple matter
to pick out a certain person by his voice. This is not always the case when
contact is by way of the telephone, since this instrument is not wholly
responsive to high-order harmonics, its high audio-frequency range being
considerably limited, and causing a change in the quality of a voice. This
effect is aggravated by speaking through a cloth held in front of the micro-
phone mouthpiece.
Hi-fi amplifiers must be capable of responding fully to all high-order
harmonics, and themselves must not be responsible for the introduction of
harmonics which are not present in the original sound.
Harmonics consist of notes having 2, 3, 4, etc., times that of the funda-
mental. The violin, for example, is rich in harmonics at twice and five times
the fundamental note to which the string is tuned. The amplitude of the
harmonic is also important, and is relatively large in the case of a violin.
In Fig. 1.2 (a) two sine waves representative of simple harmonic motion,
one of which has twice the frequency of the other, are given individually,
the higher-frequency one being the second harmonic of the lower-frequency
fundamental. In Fig. 1.2 (b) the sum of the two waveforms is given graphi-
cally, it being obtained by adding the ordinates of the fundamental and
second-harmonic waves. A waveform such as shown in Fig. 1.2 (b), being
obtained at the output of an amplifier as the result of a pure sine wave input
(Fig. I. 1), would indicate most forcibly that the amplifier itself is producing
a very large degree of second-harmonic distortion. Apart from being revealed
on the screen of an oscilloscope, the distortion would be readily detected,
since the ear is capable of recognizing the two sounds, even when they are
compounded to form the wave of Fig. 1.2 (b).

TRANSMISSION OF SOUND
Any sounding body causes the surrounding air to be alternately com-
pressed and rarefied in sympathy with the vibrations. As long as the vibra-
tions occur, a wave of high pressure is followed by a wave of low pressure
16
 HI-FI FUNDAMENTALS
and again by a wave of high pressure, and so on. Compression and rare-
faction waves are thus radiated in all directions from the sounding body at
1,088 feet per second, and an eardrum within range will be caused to vibrate
in exact sympathy.
Air is the chief medium for the transmission of sound waves, as is
clearly revealed by the classic experiment of extracting the air from a bell-jar
in which is placed a sounding electric bell. As the amount of air in the jar
becomes smaller, the sound of the bell gets weaker. To a lesser degree, all
material substances can transmit sound waves. A wood rod, for example, is
sometimes used to detect mechanical noises in a car engine. One end of the
rod is held in contact with the ear while the other end is held in close contact
with the region of the engine being checked for noise. The wood rod serves
to transmit the sound waves in this case.
Sound waves in air are known as longitudinal waves. This term simply
indicates that the particles of the wave-carrying medium travel backwards
and forwards in a path whose direction is the same as that in which the wave
is travelling. Electromagnetic waves, on the other hand, are known as trans-
verse waves, indicating that the particles of the medium travel in paths at
right-angles to the path of the wave as, for instance, the waves upon the
surface of water.
Sound waves cannot directly be represented by a sine curve, since the
particles of the wave-carrying medium remain in a straight line, being com-
pressed and rarefied as we have seen. Nevertheless, it is possible to represent
diagrammatically, to scale, longitudinal waves by means of a sine curve.
The result is similar to the sine wave in Fig. 1.1. Such a wave possesses four
distinct characteristics, which are (I) amplitude, (2) frequency, being the
number of complete cycles per second emanating from the sounding body,
(3) the velocity at which the wave travels from its source, and (4) wavelength,
being the distance between each consecutive peak. The wave will also be
endowed with the shape created by harmonics of the fundamental frequency
(Fig. 1.2 b).
It is important to remember the relation between wavelength, frequency
and velocity which, irrespective of the form of the wave, is expressed as the
velocity being equal to the product of the frequency and wavelength, or
velocity (V) ~ frequency (f) times the wavelength()..). The wavelength can
be found by dividing the velocity by the frequency, i.e.,
), feet = 1,088//
This expression can be useful when investigating for standing waves in the
listening room, as well as for other purposes.
It sometimes happens that the service technician, hi-fi enthusiast or
sound engineer is called upon to supply sound reinforcement in the open air
17
 THE PRACTICAL HI-Fl HANDBOOK

WINO
--- r±HH
DIRECTION -
Fm. 1.3. Showing at (a) how sound
waves are inclined downwards when the
wind is in the direction of the sound, and
at (b) how the waves are given an
la)

- ft\-\\\ opposite tilt when the wind is against the

sound. Similar refraction occurs due to
WINO

-
-
DIRECTION -

(bl
temperature variation of the atmosphere.

-at a iete or garden party, for example-when the question may arise of the
effect of wind upon sound waves from the loudspeakers. When the wind is
fairly strong it is desirable to place the loudspeakers (with due consideration
to the other factors involved) in relation to the listeners so that the sound is
travelling with the wind. This is not because the wind affects the intensity of
sound, though the velocity would be changed.
The reason that the sound is more clearly heard when it is travelling with
the wind than if there were no wind, and vice versa, is that the sound waves
are tilted as the result of increasing wind velocity with increasing altitude.
This effect is illustrated in Fig. 1.3, where at (a) is shown how the waves are
inclined downwards when the wind is in the direction of the sound and,
at (b), the opposite tilt when the wind is against the sound. It must be remem-
bered that the waves always travel at right-angles to their own planes and,
under the influence of wind, their velocity is altered with increasing height.
The velocity of sound waves is also affected by temperature. A tem-
perature rise promotes an increase in velocity, and the effects shown in Fig.
1.3 are often produced from this cause. During a hot summer's afternoon,
for instance, sound waves may be tilted skywards as the result of the air
temperature being greater at lower levels than at higher levels. The converse
effect is often experienced when the lower air layers are at a lower temperature
than the upper air layers. For this reason, distant sounds are often clearly
heard on a cool. still evening, the effect being particularly noticeable over the
surface of water.

BEATS
When one is slowly overtaking a noisy heavy goods vehicle in a car
whose engine is not unduly quiet, a drumming or beating sound may develop
and vary in frequency as the car engine is increased in speed in order to
overtake the other vehicle as quickly as possible. When this effect is first
experienced, one may incorrectly conclude that the back axle is due for
renewal! The disturbance, however, is quite natural, being caused by a beat
tone created as the result of sound waves from the two engines combining,
18
 HI-Fl FUNDAMENTALS
the frequency of the beat being equal to the difference in frequency of the
two sounds involved.
Such beats are sometimes produced in amplifiers, pick-ups and loud-
speakers, and may give rise to spurious tones, referred to as intermodulation
distortion, which may or may not be harmoniously related to the tones from
which they arise. The distortion usually gives considerable harshness to audio
reproduction, as well as to the sound of car engines!
The hi-fi technician will encounter many problems in which resonance
plays a leading part. If an audio oscillator is connected across the terminals
of a hi-fi loudspeaker system, and the oscillator is tuned fully over the audio
spectrum from about 15 c/s to 15 kc/s (15,000 c/s), it will be found that at
various frequencies different objects in the room will start vibrating vig-
orously in sympathy with the sound produced by the loudspeaker. (Let us
hope that the loudspeaker enclosure is not subject to such disturbance.)
When the sound has a frequency equal to the natural frequency of an object,
then the object will vibrate in sympathy with this sound. This process is
called resonance.
Heavy damping of the object, due to its design and firmness, will
greatly reduce the intensity of the resonance. Loudspeaker enclosures are
usually made so as to reduce their natural resonance to the minimum,
though at the time of writing a speaker enclosure is undergoing development
that is designed intentionally to resonate or flex at certain frequencies. The
enclosure panels are designed to resonate at different frequencies as a means
of damping the air column resonance within the enclosure, and so spread
the effectiveness of the damping over a wider frequency range. It is reasoned
that the more conventional method of acoustic damping wastefully converts
sound energy at the resonant frequency into heat.
The usual arrangement, which is often adopted for hi-fl, is to use sand-
filled panels, or panels of concrete, for speaker enclosures. In this way
complete rigidity is secured, and there is little fear of the enclosure walls
flexing, even when the alternating sound pressure within the enclosure is at a
high level.
Resonance effects are at their height in the small, popularly-priced
record-players, often colloquially referred to as "pop boxes"-not a hi-fi
term! Here the loudspeakers (or loudspeaker) are contained within a portable
housing along with the amplifier and record-player-often an auto-unit is
employed. If an audio oscillator is connected across the speaker of one of
these devices, things really start resonating within the box. After the case
itself has ceased to resonate up to 200 c/s, the valves in the amplifier take
over, then the various metal levers of the record unit at about 2,000 c/s,
and so on.
When the instrument is used as intended, the box resonance enhances
19
 THE PRACTICAL HI-FI HANDBOOK
the bass response in a synthetic manner, and when "bop" records are played
the other higher-frequency resonances undoubtedly merge with the general
background effects. There are on the market, however, quite good portable
record-players in which undesirable resonances have been damped as far as
is possible. These instruments, of course, are more expensive than the single-
valve outfits which are produced essentially for the reproduction of popular
music in current demand. Nevertheless, true hi-fi equipment is demanded
for true fidelity reproduction, and portable equipment is then completely out
of the question. Separate units are essential, and pieces of equipment which
are prone to resonance should as far as possible be removed from the loud-
speaker system.
The power of resonance is illustrated by the traditional order "break
step" which is given to a company of soldiers about to cross a bridge. If
the troops' normal rhythmic step happened to coincide with the natural
frequency of the bridge, vibrations of large magnitude would be promoted
and there would exist a definite possibility of the bridge breaking up.
Apart from the resonance effects of objects, air itself can be caused to
vibrate at certain frequencies under controlled conditions. As an example,
tuning-forks are sometimes mounted upon hollow boxes so as to increase
the volume of sound. The normally feeble sound from a tuning-fork is
considerably amplified because the size and shape of the box is arranged so
that the air inside possesses a natural vibration period equal to that of the
fork. Thus, both the vibration of the fork and the vibration of the air, at the
particular tuned frequency in both cases, contribute to the total energy of
sound applied to the ear. Such a box is known as a resonator.
This particular effect must not be mistaken for the increase in volume
which can be obtained by holding the stem of a tuning-fork in close contact
with a table-top or board. In this case, the table-top of board simply serves
as a sounding board; forced vibration is produced by the fork, and as a
consequence the overall vibration is communicated to a much greater quan-
tity of air than when the fork is vibrating unaided.
A well-known resonator is that due to Helmholtz. It was developed some
hundred years ago for the purpose of harmonic analysis of a note, and it is
still used for this purpose. Such resonators consist (in the original) of a brass
spherical shell on which is formed a taper containing a small hole for the
purpose of inserting into the ear. Diametrically opposite is a larger opening
for presenting to the source of sound.
The air in the resonator resonates to one particular frequency-that
to which the resonator is tuned-and when a sound is applied, the resonator
picks out and amplifies only that component of the sound to which it is
tuned. In this way components of a complex note too feeble to be detected
by the ear alone become easily audible and can be checked for relative strength.
20
 HI-FI FUNDAMENTALS
Resonators of this kind are made in sets, the note of each being set to the
required standard. The resonant or resounding frequency is governed by the
volume of air and the area of the pick-up aperture. The frequency is decreased
by increasing the volume of air or by decreasing the area of the aperture.
The phase inverter or reflex loudspeaker enclosure adopts the principles
of the Helmholtz resonator at the low-frequency end of the audio strectrum.

STANDING WAVES
Resonances also occur in the listening room, as the hi-fi service
technician will undoubtedly discover for himself during the process of
investigating for poor results in a customer's home on equipment which has
previously worked with excellent results in the demonstration room! Such
resonances, sometimes referred to as eigentones, are produced by multiple
sound reflection between the opposite walls, and occur at the frequency at which
the distance between the opposite walls is exactly one half-wavelength. This
condition gives rise to standing waves at the critical frequency, whilst also
considerably accentuating the response at the resonant frequency. In effect,
the room serves as a resonator, and the air resounds at the frequency to
which the room happens to be tuned.
Further resonances occur as the result of the other two parallel walls and
the ceiling and floor, and others governed by the dimensions of the diagonals.
The worst conditions occur when the room approximates a cube, with the
speaker situated in the centre of a wall. Apart from the chief low-frequency
resonance or eigentone at a half-wavelength, others, though possibly less
disturbing, present themselves at all harmonics of the basic frequency. Thus,
with the main resonance at, say, 40 c/s, created by a cube-shaped room with
14 ft. sides, additional resonances at 80, 160, 320 c/s and so on will also
result. Reciprocally, it follows that the reproduction will be exaggerated at
frequencies for which the walls are a multiple of half a wavelength apart.

ELECTRICAL REPRESENTATION OF SOUND

In all forms of sound broadcasting, recording and reproduction a means
must always be provided to convert the sound energy into electricity. Such
a conversion device, capable of receiving energy in one form and passing it
on in another form, is known as a transducer. The microphone comes under
this classification.
All microphones possess a thin diaphragm on which the sound pressure
operates, and the resulting vibrations create currents of electricity which
rise and fall in precise sympathy with the sound waves. For example, a sound-
ing tuning-fork held in front of a microphone will give rise to a current
waveform of frequency coinciding with that of the fork (Fig. I ·4). Similarly,
a complex sound wave, composed of a number of tones and harmonic parts,
21
 THE PRACTICAL HI-FI HANDBOOK

~·kf\
a- VI\ f\ r
V V
FIG. 1.4. A sounding tuning fork held in front ofa
microphone will give rise to a current waveform,
as illustrated, of frequency coinciding with that
TIME- of the fork.

will be electrically reproduced with equal accuracy. Within limits governed

by the design and purpose of the microphone, the electrical output will depend
upon the intensity of the sound applied to the instrument. Increasing sound
intensity will result in increasing output, and vice versa. The electrical out-
put will also vary with the frequency of the applied sound, though for high-
quality work the microphone must respond evenly over the whole of the
audio spectrum.
The output of a microphone is conveniently expressed in decibels relative
to a fixed reference level. The reference level chosen is sometimes 0 db =
1 volt (open-circuit) with a sound pressure of 1 dyne per square centimetre.
Thus, a microphone with an output expressed as - 74 db below I volt/dyne/
cm2 would have an open-circuit voltage of approximately 0·0002 volts r.m.s.
The output is sometimes expressed in terms of power for a stated sound
pressure. The RMA rating is defined as the ratio in db relative to 0·001 watt
dyne per square centimetre.
At this point it should be noted that a sound pressure of 0·0002 dyne
per square centimetre corresponds to the limit of audibility of a 1,000 c/s
note. This in turn corresponds to zero phon, and to give the reader some idea
of the loudness scale, a quiet room is rated at 20--30 phons, average conversa-
tion 60 phons, interior of a tube train with the windows open 90 phons,
proximity to an aeroplane engine 120 phons, while 130 phons is approaching
the threshold of feeling or pain.

SOUND REPRODUCTION
To be of practical use, the very small power available at the output of
the microphone must be considerably amplified, and this has to be performed
without alteration of either the character of the electrical waveform, due to
the sound waves, or of the response over the entire audio spectrum. With
regard to the latter consideration, however, poor acoustics of the room in
which the microphone is used (the studio) can sometimes be countered by
the use of a frequency-selective network between the microphone and ampli-
fier input. For example, the exaggerated response at low frequencies due to
a room of small dimensions is sometimes mitigated by the introduction of a
filter network which attenuates the bass frequencies at the microphone, in
relation to the higher frequencies, before the signal is applied to the amplifier.
This process is known as equalizing for room acoustics. Similarly, the
equalizing function may take place somewhere in the amplifier chain.
22
 HI-FI FUNDAMENTALS

+25 TONE

+20
+15 MAX
MAX
+10
+s
d80 FL.AT FL.AT

-s
-10
-IS
MIN
-20
MIN
-25
10 100 1,0 10.000 2QOOO
BASS c/s TREBLE

Fm. 1.5. Curves showing how the bass and treble response of an amplifier can be
altered to suit the acoustics of the listening room.
Most amplifiers are composed of three distinct sections. First there is
the voltage amplifier whose purpose is to step-up the small audio-frequency
(a.f.) voltages occurring in the varying sound input to a workable level. This
section may also contain equalizing networks of suitable form to cater for
the various signals for which the voltage amplifier is going to serve. Next
comes a tone-control section, in which controls are available for adjusting
the degree of amplification of the treble and bass frequencies of the signal,
usually relative to 1,000 c/s.
The idea is illustrated in Fig. 1.5. It will be seen that the bass is continu-
ously variable from -12 db to + 12 db at 40 c/s, and that the treble is
continuously variable from -15 db to + 12 db at IO kc/s. Having such a
control of the response of the amplifier aids considerably in the correction of
impaired room acoustics from the reproducing point of view. The presence
of low-frequency resonances, for instance, can be prevented from over-
emphasizing the bass from the aspect of the listener by applying a suitable
degree of bass cut and, possibly, treble lift. Conversely, some rooms may be
acoustically "dead"; they have a tendency to absorb more of the lower and
higher frequencies and thus seem to require more bass and treble than
average rooms. Tone controls serve to correct such deficiencies of the listening
room and maintain the faithful balance demanded by the hi-fi enthusiast.
Finally, the equalized, amplified and tone-controlled signal is passed
on to the power amplifier, by way of a volume or loudness control, and is
changed from voltage to power for operation of the loudspeaker. Some
equipments have the power amplifier as a unit completely independent of the
23
 THE PRACTICAL HI-FI HANDBOOK
voltage amplifier and tone-control section, while other smaller amplifiers
are complete in themselves. A block diagram of the three sections we have
discussed is given in Fig. J.6.
The loudspeaker is also a transducer, but it operates in the opposite
way to that of the microphone; it receives an electrical representation of the
sound which was applied originally at the microphone, and passes it on in
the form of sound energy. We shall discuss both microphones and loud-
speakers in some detail in later chapters.
We now have a complete picture of the whole chain of events, from the
sound waves to the microphone, from the microphone through the amplifier
to the loudspeaker, and from the loudspeaker to the ear. Let us always bear
in mind that the results heard are a function of the mind of the individual,
and that they are coloured not only by the equipment used for the reproduc-
tion of the sound, but also by the studio and listening-room acoustics.
Although it is impossible to match the acoustics of the ordinary listening room
with those of the concert hall, it is surprising what can be done synthetically
by equalizers and tone controls, not to mention loudspeakers and enclosures!
SMALL MICROPHONE AMPLIFIED MICROPHONE

SOUND V~O~AGE CfJ>1 VOLTAGE

SOUND
WAVES
WAVES

•I EQUAL· VOLTAGE TONE POWER

f
1111111\,1:JI I Z ER AMPLIFIER CONTROL AMPLIFIER 1:1:11m1111
~
~ MICROPHONE LOUDSPEAKER
TUNING
FORK BASS TREBLE

Fm. 1.6. Block diagram of the three main sections of a hi-ft amplifier.

We have so far considered "live" reproduction of sound, that is direct

from the microphone to the loudspeaker. Of course, sound can be "stored"
and used when required. The most popular medium for storing sound in this
way is the gramophone record.
Instead of actuating the cone of a loudspeaker to produce sound waves
coinciding with those at the microphone, the microphone-amplifier set-up
powers a recording head whose purpose is to cause lateral vibration of a
sapphire or diamond cutting tool in sympathy with the sound energy applied
at the microphone The recording head is tracked radially over the recording
blank, while the blank is carried on a heavy turntable which is arranged to
rotate at a perfectly even speed. The recording head is also pivoted in such
a way that the cutting tool is pressing on the surface of the disk. A spiral
groove is thus cut upon the surface of the disk, running from the circumference
to near the centre. The sound vibrations impart upon the groove a wavy
24
 HI-FI FUNDAMENTALS
lateral effect, which is clearly visible on any gramophone record. Sound is
thus "stored".
Playback is accomplished by rotating the disk at the same speed at which
it was recorded, and by the use of a pick-up carrying a stylus having a hemi-
spherical tip, which rests in the V-section groove cut by the cutting tool.
The pick-up is mounted on a tone arm which is free to rotate about a centre
some distance from the centre of the record, and is free to move only in an
arc which is approximately radial with respect to the record.
The pick-up is thus carried across the record under the control of the
groove, while at the same time the stylus is caused to vibrate laterally in
sympathy with the lateral waveform imparted during the recording process.
The lateral vibration of the pick-up stylus gives rise in the pick-up to an e.m.f.
(electromotive force) having the same pattern as the waveform on the record
and, in some cases, in proportion to the velocity of the lateral movement of
the stylus. This e.m.f. is applied to the input of an amplifier, is suitably
equalized, and ends up as sound waves from the loudspeaker-ideally, as
a replica of the sound waves applied to the microphone during the recording
session.

STEREOPHONIC SOUND
A single-channel (often referred to as monaural) reproducer system, i.e.,
one microphone, one amplifier system and one loudspeaker system into
which the single channel is working (more than one microphone and loud-
speaker may well comprise a single sound source from the monaural aspect),
can never give true fidelity of reproduction. Highly satisfactory reproduction
of an orchestra cannot be secured if all the sound is radiated from a hole in
the loudspeaker cabinet. The use of two or three speaker systems does not
help much in this respect when they are all connected to the output of a
common channel.
The "range" of the orchestra can only be realized by the use of two or
more completely independent channels. With a two-channel system, which
is highly suitable for domestic use, there are two loudspeakers each fed
from a separate microphone (or from a separate signal source) through
separate amplifiers. The basic idea is to place the loudspeakers relative to one
another as the microphones are placed in front of the orchestra, or as they
were placed during the recording of the programme.
In this way both ears of the listener are brought into operation in a
selective sense. The orchestra appears to be spread in correct proportion
across the room, between the loudspeakers, and the listener can pick out the
individual instrumentalists as readily as if he were in the concert hall. The
"muddiness" of the monaural system disappears completely, and a third-
dimensional sense of presence is created.
25
 THE PRACTICAL HI-FI HANDBOOK
At present the normal method of stereophonic recording on disks is
what is known as the "45/45 system" where the two stereo channels are
carried in one groove. This system (described in more detail in Chapter 10)
has taken the place of the system where one channel is recorded by the
"hill-and-dale" process, as adopted 80 years ago by Thomas Edison in
connexion with the phonograph. With this, the cutting tool of the recording
head was arranged to oscillate vertically in sympathy with the sound
vibrations, so that the depth of the groove corresponded to the wave pattern
of the sound; hence the term "hill-and-dale". The other channel was recorded
in the same groove laterally, and a special recording head was used to
modulate the groove both laterally and vertically in accordance with the
two-channel signals applied. On playback, a pick-up functioning electrically
opposite to that of the recording head gave two outputs corresponding to
the two recorded channels. The signals were amplified independently, and
were fed to the two loudspeakers to give the effect of stereophony. Hill-and-
dale/lateral and 45/45 stereo recordings differ essentially only in the way in
which the signals in the two channels are phased.

MAGNETIC RECORDING
Wire and tape coated with a magnetic material are also used for record-
ing. The wire or tape (wire is now rarely used) is drawn steadily over the pole
of an electromagnet, the current in which is caused to follow the wave pattern
of the sound. The wire or tape thus becomes magnetized, as each section
passes over the pole piece of the electromagnet, to a degree dependent upon
the electrical representation of the sound applied at the microphone, and a
magnetic wave-pattern is imparted upon the medium.
On playback, the medium is again drawn at the same speed across the
pole piece of an electromagnet which this time is not energized, but which
has induced in it small voltages corresponding to the varying flux in the core
as the result of the magnetized medium. The voltages, representing the
recorded sound signal, are applied to the input of an amplifier and end up as
sound from the loudspeaker. When required, the recording can be easily
erased by passing the medium over a permanent magnet or an electromagnet
which is energized by a pure signal having a frequency above the audio
spectrum (30-50 kc/s).
This system lends itself readily to two-channel operation, it being a
simple matter to record one channel on one half of the tape and the other
channel on the other half by the use of slightly displaced electromagnets for
record and playback.
Sound can also be stored on film, on the principle adopted for the sound-
track on cine film, but a description of this method falls outside the scope of
this book.
26
 CHAPTER 2

Voltage Amplifiers, Feedback

and Control Circuits

THE weak signal voltages at the output of the microphone,

pick-up or tape recorder have to be increased in magnitude, altered in
response to suit the acoustics of the listening room, and finally changed into
power for adequate operation of the loudspeaker. The hi-fi amplifier system
is required to perform these functions without changing the character of the
original signal to any large degree, whilst also maintaining a level response
over the whole of the audio spectrum and catering for a wide range of signal
levels, from the smallest to the largest, without distortion. The dynamic
range of a programme signal may well extend to the region of 60 db, which
means that the signal range in terms of input voltage may extend from
2 microvolts to 2 millivolts, resulting in a few milliwatts of output from
the loudspeaker on the quietest signal or several watts on the loudest
signal.
The first duty of the amplifier is to step-up the signal voltage to a level
suitable for operating the output stage, or power-amplifier section. Bearing
in mind that the power amplifier may require a signal level approaching
I· 5 volts to fall within its driving range and that the programme signal may
only average I millivolt, one can clearly realize the necessity of voltage
amplification of some 70 db.
At this juncture we should note that the term "voltage amplifier"
indicates that the re-creation of the amplified input signal across a com-
paratively high impedance load is the essential function of this section.
Obviously, since an infinite output impedance does not exist in practice, it
must be concerned also with the amplification of power. However, the output
impedance is usually of the order of O· l to I megohm, so voltage is the pre-
dominant factor. We need not consider here when an amplifier ceases to be
concerned with voltage and changes over to power, but simply regard power
amplifiers as those used to drive a loudspeaker.
A typical pentode voltage amplifier is shown in Fig. 2. I. There is nothing
27
 THE PRACTICAL HI-Fl HANDBOOK
....-------H.T.• Fm. 2.1. Pentode voltage-amplifier stage.

about this circuit as itappearsonpaper

to indicate that it is associated with
O•I
hi-fi equipment. In practice, however, it
Sl~NAL---it-t--t-"'Nlfv1r.=.::
will be found that possibly the anode
and grid resistors are of the "low noise"
variety, also the valve (i.e. Mullard
EF 86 or GEC Z729-low-noise types).
Careful choice of circuit parameters
ensures that the stage is operating at
the lowest distortion figure within its
voltage range. Hum problems are also taken care of by the use of carefully
placed wiring, and by having a common "earthing" point so as to avoid
producing common impedances across which hum voltages may develop.
A triode voltage amplifier is shown in Fig. 2.2. In practice, VI is usually
one half of a double-triode valve, the other half being employed also in the
voltage amplifier, with a frequency-selective feedback loop (usually switched)
providing various degrees of equalization. This method is adopted in the Pye
"Proctor" pre-amplifier. Generally speaking, there is little to choose between
a modern pentode and triode. Greater gain is usually possible from the
pentode, though such a stage does not give such freedom from intermodula-
tion distortion at high output voltages as does the triode. Nevertheless, this
is of little moment when the signal to be handled is of very low level.

NOISE AND HUM

Owing to the very low level of the signals to be amplified by the
first stage, a problem peculiar to this section is that of "noise". Noise in
this sense refers to all spurious disturbances which give rise to unwanted and
undesirable signals across the output load
along with the required signal. Since a quiet
, - - - - - H.T.+
orchestral passage may give rise to only
a few microvolts of signal across a
pick-up or the playback head of a tape
VI
recorder, the stage will obviously o-,
SIGNAL ..._1 IOOK
IN --,

FIG, 2.2. Typical triode voltage-amplifier

stage. Vl usually comprises one half of a
double-triode, the other half-section being
used to complete the stage. A frequency-
selective feedback loop may also be used for
the purpose of equalization. FEEDBACK LOOP

28
VOLTAGE AMPLIFIERS, FEEDBACK AND CONTROL CIRCUITS
be susceptible to the pick-up of stray hum signals and noise caused by
mcorrect or faulty components.
Ideally, the only noise voltage which should be present with the signal
across the output load is that attributable to the random behaviour of
electrons in the resistive components and in the valve. This is sometimes
referred to as white noise, and is characterized by the hiss which emanates
from the loudspeaker when the volume control of a very high-gain amplifier
is turned full on. White noise is not confined to any particular frequency,
but is distributed throughout the entire spectrum. If the equipment has peaks
over -its response, the effect of the noise will be considerably emphasized at
the frequencies corresponding to the peaks. Because our ears tend to "peak"
around 3 kc/s, white noise resolves as a hiss, focused on 3 kc/s. An interesting
test is to apply white noise to the input of a hi-fi amplifier by way of switched
filters serving to attenuate progressively the high- and low-frequency com-
ponents of the noise. A filter tuned to around 600 c/s changes the hiss to a
whistle, while a filter tuned to the low-frequency end of the spectrum gives
rise to a roar.
With practice, white-noise tests of this nature permit rapid appraisal of
the performance of hi-fi equipment, particularly if an oscilloscope can be used
in the tests (remembering that the character of white noise is rather like that
of transients, about which more will be said later).
Unfortunately, apart from white noise, there often exist other and more
disturbing spurious signals across the output load of the voltage amplifier.
Hum is a big bugbear in this connexion. Hum poses much more of a problem
in hi-fi equipment than in ordinary domestic radios of limited low-frequency
response. In the first stage, hum is invariably induced into the input circuit
from either stray electromagnetic or electrostatic fields. In most hi-fi ampli-
fiers, the control-grid circuit of the voltage amplifier is reached by way of the
programme-selector switch, which gives the positions "gram", "tape" and
"mic".
Hum caused by stray fields will diminish on backing-off the volume
control, since the control is usually located in the circuit following the output
of the voltage amplifier. Operating the selector switch also gives conclusive
evidence as to whether the hum pick-up is common to all circuits or a short-
coming of one particular channel. If the amplifier is in good order, as can
nearly always be proved by these simple tests, it is safe to assume that the
hum signal is gaining admittance either by way of the leads connecting to
the various sources of programme signal or by way of the external units
themselves.
Electromagnetic induction occurs mainly in circuits of low impedance,
and demands a complete loop into which induction can occur. For example,
a low-impedance magnetic pick-up may enter an electromagnetic hum field
29
 THE PRACTICAL HI-Fl HANDBOOK
as it traverses the turntable. The hum field may emanate from the gram motor
or from a power transformer situated nearby. Whatever the cause, a small
voltage (in terms of microvolts) will circulate the circuit comprising the pick-
up coil and the primary of the matching transformer, but this voltage will
appear at the grid of the valve stepped up in the same ratio as the matching
transformer. Thus, an induced voltage as small as 2 microvolts will rise to
JOO microvolts at the grid with a transformer having a turns ratio of 50:1,
which is a reasonable value for an input transformer.
Electromagnetic induction of a similar nature may well occur in a low-
impedance microphone circuit, in the circuit of the playback head of a tape
recorder, or even at the coupling transformer. The overall loop effect can
be obviated by employing either a tightly twisted pair of conductors or a
coaxial line between the programme signal source and the low-impedance
amplifier input. There is little purpose in using parallel conductors as these
aggravate the loop effect, and a screening over such conductors offers little
or no protection against electromagnetic fields.
The susceptibility of the inductor at the low-impedance signal source
(such as the winding and core associated with a low-impedance magnetic
pick-up) and the coupling transformer at the amplifier end in responding to
electromagnetic fields, particularly those at mains frequency (50 c/s}, can be
reduced or almost eliminated by the use of high-permeability magnetic
shields. The effect of such a shield at low frequency is shown in Fig. 2.3.
Where higher-frequency electromagnetic fields are present, more
elaborate screening is usually called for. With a microphone transformer,
for instance, it is often necessary to house it in a case formed of several
shields, two of the type described and an intermediate one which operates
by inducing into itself a field which opposes the offending field.
Electrostatic induction rarely affects low-impedance circuits, since the
electrostatic charge is quickly dissipated around the low-impedance loop.
However, in the case of a low-impedance circuit isolated from chassis, an
electrostatic noise charge may appear at the grid as the result of the charge
developing between the low-impedance circuit as an entirety and chassis.
This can usually be cleared simply by earthing the low-impedance side of the
circuit, preferably at a centre-tap on the primary of the transformer.

-----------------
... --------------- --------
-,--------------,'
/ MAGNETIC SHIELD_;;, \ - - - - - - --

SPACE INSIDE SHIELD LOW FREQUENCY FIG. 2.3. A magnetic

ELECTROMAGNETIC
CLEAR OF FIELD
FIELD
shield serves to protect
' \ / -------- the transformer or induc-
,,
~
---------'✓
'·- ;
- - - - - - - - - - - - --:... ,,,,. -- - - - - - - - tor from stray electro-
magnetic fields.
30
VOLTAGE AMPLIFIERS, FEEDBACK AND CONTROL CIRCUITS
Long lines between the programme source and the input of the amplifier
should always be avoided if hum is to be kept at the lowest level. If some
reason or other necessitates long leads, then 600-ohm transformers should
be employed either side of the line, that is at the programme source and at the
amplifier. In this way the danger of introducing hum due to different earth
points can be avoided.
High-impedance connecting links should always comprise shielded cable
having a low value of capacitance, bearing in mind that excessive capacitance
at high impedance will impair considerably the high-frequency response of
the programme signal. Television coaxial cable can be used successfully for
this purpose, but if the lead is extra long and is liable to be moved about a
lot-for example, if it is used as a trailing high-impedance microphone cable
-micropbony effects may become troublesome. These resolve as the result
of small changes in resistance of the outer conductor as the cable is moved
and give rise to a "brushing" noise from the loudspeaker. If the cable is
dropped, the effect from the loudspeaker is similar to that obtained by tapping
the microphone.
Enthusiasts living close to a powerful television transmitter often com-
plain of hum caused by the pick-up of the vision signals. In some cases the
sound transmission is also troublesome. This unwanted pick-up gains admit-
tance to the equipment either at the input circuits of the voltage amplifier
or on the negative-feedback loop, by way of the speaker leads. It can be
cured without difficulty by the insertion of a television choke in the speaker
leads, as close as possible to the amplifier, or in the control-grid circuit of
the voltage-amplifier valve, as close as possible to the grid tag. A choke (or
chokes) can easily be made up by forming a self-supporting coil from 18
s.w.g. tinned copper wire, with the wire equal in length to that of a quarter-
wavelength of the frequency of the offending signal (67 inches for Channel 1,
59 inches for Channel 2, 53· 5 inches for Channel 3, 49 inches for Channel 4,
and 45· 25 inches for Channel 5). IT A stations have not yet been reported as
responsible, possibly because of the higher frequency used.
Trouble similar to this was reported on a hospital call system. In this
case, though, the pick-up was from the v.h.f. transmitter employed in ambu-
lances. The trouble was cured by the insertion of chokes and the making good
of poor connexions on the call-system microphone. In the latter respect, the
trouble was being aggravated by rectification (demodulation) of the signal
caused by high-resistance joints.

VALVE MICROPHONY
Valve microphony is another factor which affects the voltage amplifier.
The effects are ringing from the loudspeaker and a definite "ping" when the
valve is tapped with a finger. Essentially, the trouble is caused by vibration
31
 THE PRACTICAL HI-FI HANDBOOK
of the electrode structure, promoting corresponding signal fluctuations across
the anode-load resistor. High-slope triode valves are more susceptible to the
effect than pentodes, particularly older-type triodes. Modern valves are less
prone to the trouble, and circuit techniques help, as in these days it is not
common practice to run the valve for maximum gain. Microphony is aggra-
vated by vibrations from the loudspeaker, particularly when the speaker is
situated in the same cabinet as the amplifier.
After the first amplifier stage, the signal is usually large enough not to
be affected by problems of noise, since then the noise voltage is a very small
ratio of the signal voltage. Apart from signal amplification, the duty of the
first stage is that of securing the highest possible signal-to-noise ratio, and
this is no mean task when it is considered that the applied signal voltage may
well have a magnitude of only a few microvolts on soft passages of music.
As an aid in maintaining a good signal-to-noise ratio, the full signal
voltage is invariably applied to the control grid of the first valve; the volume
control being introduced after initial amplification when the signal is at much
higher level. There are times, however, when the programme signal itself is
at high level; for instance, when a high-output pick-up is used or when an
amplifier is incorporated in a radio tuner or tape recorder. When this is the
case, some form of attenuation is needed between the programme source and
the amplifier input to avoid overloading the first valve.

FEEDBACK

One cannot progress far into hi-fi before coming up against feedback.
There are two kinds of feedback, positive and negative. Positive feedback
means that a portion of the output signal of an amplifier is fed back to the
input in the same phase as the applied signal. The application of positive
feedback results in an increase in gain of the amplifier, and instability and
oscillation when the feedback exceeds a certain degree. Positive feedback is
the modus operandi of oscillator circuits. Negative feedback, on the other
hand, results in degeneration, and is arranged by feeding back a portion of
the output signal in opposite phase to the applied signal.
There are also two modes of feedback, current and voltage. The former

OUTPUT VOLTAGE (Eo)

I
IN
AMPLIFIER
GAIN =A 1--~,-+------ OUT

FIG. 2.4. Block diagram

showing the application
ofnegative feedback to an
amplifier.
32
VOLTAGE AMPLIFIERS, FEEDBACK AND CONTROL CIRCUITS
occurs when the feedback voltage is proportional to the output current, and
the latter when the feedback voltage is proportional to the output voltage.
The most common method of obtaining current feedback is by the use of
an un-bypassed cathode resistor in a valve amplifier. Here, the signal voltage
acros-. the cathode resistor, being in proportion to the current, is reflected
anti-phase into the control-grid circuit.
Now for a few simple expressions.
The gain of an amplifier stage is reduced by the omission of the cathode
capacitor by the factor

+ µ.} +z
+ Rk (I-----
ra
where Rk is the cathode resistor, µ. and ra are the amplification factor and
a.c. resistance of the valve, and Z is the anode coupling impedance. The
expression can be reduced to

I+ g Rk
where g is the mutal conductance of the valve, when ra is large compared
with Z, as is often the case with pentode valves.
The block diagram in Fig. 2.4 represents an amplifier with a negative-
feedback loop. Writing A for the gain of the amplifier without feedback, and
B for the fraction of the output voltage fed back, it can be stated that

Gain with feedback = A

1 +AB
The factor (I + AB) is known as the gain reduction factor, and may be
expressed in decibels.
As an example, in an audio amplifier we may have A = 180 and
B = 1/20, giving AB= 9 and the feedback factor {I + AB)= 10. The gain
with feedback is then 180/10 = 18. In this case, the feedback has reduced the
gain of the amplifier 10 times, which is the same as saying that the amplifier
has 20 db feedback. Clearly, the input signal required with feedback to
secure the original output must be (1 + AB) times the input signal without
feedback. In other words, in the foregoing example, the application of
negative feedback has made it necessary to increase the input signal 10 times
(20 db} in order to obtain the original output signal.
The term "loop gain" often appears in relation to feedback. It refers to
the factor AB, which is the gain that would be indicated by applying the signal
at the grid of the first valve and measuring the signal at the output of the
feedback loop.
Within limits, the application of negative feedback reduces distortion in
the same ratio as it reduces the gain. For example, a small amplifier without
33
 THE PRACTICAL HI-FI HANDBOOK
feedback may well produce something like IO or 11 per cent distortion
(possibly made up of IO per cent in the output stage and I per cent in the
preceding stages). The application of 20 db feedback will not only reduce the
gain of the amplifier by a factor of 10, but also the distortion by the same
factor-in this case, bringing it down to something like l per cent. This is,
indeed, a useful application and one which is practised extensively in all hi-fi
equipment. We shall see later that modern equipment includes a number of
feedback loops each serving a specific purpose.
Positive feedback can be added to improve even further on the distortion
figure. If we again consider the small amplifier mentioned above, and apply,
say, 6 db positive feedback over the stages preceding the output stage, both
the distortion and gain of these stages will be increased by a factor of 2
(positive feedback increases both the gain and distortion, as would be
expected). The distortion in the first two stages will thus rise from I per cent
to 2 per cent.
Without negative feedback, therefore, the overall distortion will now be
in the region of 12 per cent. Now if the fraction of output voltage B fed back
is maintained as in the original example, the negative feedback will be 26 db
instead of the original figure of 20 db. This, of course, is because A has been
doubled, as also has the loop gain (AB). This means that the distortion will
be reduced by a factor of 20 (26 db), which brings it down to something like
0·6 per cent.
If the positive feedback is further increased the negative-feedback loop
gain will rise in proportion and the distortion will reduce accordingly. As with
all things, there is a limit to which positive feedback and negative feedback
can be increased. However, this device lends itself admirably to the use of
triode voltage amplifiers, as distinct from pentodes. With a double-triode
valve, for example, positive feedback may well permit a stage gain of some
2,000 times, thus allowing the use of some 20 db more negative feedback as
compared with a pentode.

FEEDBACK STABILITY

The capacitive and inductive circuits employed in amplifiers can really

disturb the application of feedback. So far we have assumed that the voltage
fed back as negative feedback is exactly 180 deg. out of phase with the input
voltage, and that the voltage fed back as positive feedback is exactly in phase
with the input voltage. Unfortunately, these ideal conditions hold for only a
comparatively small range of frequencies. Series capacitive reactances, such
as coupling capacitors, cause a progressive phase advance, while leakage
inductance in transformers and odd stray capacitances in the circuit promote
a progressive phase delay. The phase change between the input and output
signals occurs progressively, positively and negatively, with frequency either
34
VOLTAGE AMPLIFIERS, FEEDBACK AND CONTROL CIRCUITS
FIG. 2.5. Typical amplitude and phase AMPLITUDE
RESPONSE
characteristics of an audio amplifier. ~o
::E
s:
----+------/
9
w
side of the frequency design centre of ;
the amplifier. The effect is illustrated .. 180
graphically in Fig. 2.5. +120
Such phase change with fre- +60
quency causes the feedback voltage
to vary in phase in relation to the in- - 60
put voltage either side of the optimum -120
for the feedback employed. However, · 180
for negative feedback to change to ,_1~ - -.......--+-----<,oo>-o_,..____1.,...20-

positive feedback its phase must - c/5 -

+- kc/5 -

change through I 80 deg. Whilst this FREQ DESIGN CENTRE

is unlikely to happen within the
range of the audio spectrum, it may take place at a frequency well above
audio (supersonic frequency) or at a very low frequency. At these frequencies,
the response of the amplifier is likely to have fallen to a comparatively low
level, and this, of course, will result in the loop gain (AB) reducing in a similar
ratio. The point is that for oscillation to commence, due to the change from
negative to positive feedback, the factor AB must be unity or greater. In
other words, to meet the conditions of stability, the product AB cannot be
greater than I.
Feedback design margins are employed to ensure this condition, the
"phase margin" being defined as the angle by which the phase differs from
180 deg. at the frequency where the loop gain (AB) falls to unity. A margin of
30 deg. is usually aimed at, giving a maximum phase shift of 150 deg. The
"gain margin", on the other hand, is defined as the amount by which the
loop gain (AB) is below unity at the frequency where the phase shift is 180 deg.
A typical value for the gain margin is IO db.
The application of positive feedback within a negative feedback loop,
as already discussed, can play havoc with the stability margins, and extreme
caution is needed when this form of feedback is applied. It sometimes pays
to make the positive feedback somewhat frequency-selective so that it is
applied in full force over one side of the spectrum only. Special attention can
then be given in terms of stability to the circuit parameters concerned with
phase shift at the frequency over which the feedback is applied.
When negative feedback is applied over more than two stages the
possibility of instability is considerably increased. Very low-frequency
oscillation may commence, causing the speaker cone to pump to and fro at
one or two cycles per second. This effect may not be heard as such, though it
35
 THE PRACTICAL HI-FI HANDBOOK
can seriously degrade the reproduction. Similarly, very high-frequency oscilla-
tion is inaudible, but it can upset the operating condition of the output stage,
causing the grid circuits to pass excessive current.

ADVANTAGES OF NEGATIVE FEEDBACK

Apart from reducing harmonic and intermodulation distortion to a very
large degree, negative feedback also reduces spurious noises and signals,
such as hum and microphony, in the same ratio as it reduces the gain. It also
has a marked effect on the stability of the parameters of the amplifier due to
random changes within the feedback loop. An example in this latter respect
is an amplifier which without feedback has its overall gain reduced by some
IO per cent as a result of a drop in mains voltage; the same amplifier with
20 db negative feedback has its gain reduced by only I per cent, with the same
drop in mains voltage. The reason is that the effect of gain changes is reduced
by the feedback factor, which in the case cited is 10:1 (20 db).
Negative feedback also considerably improves the overall frequency
response of an amplifier. In this case, however, it is not true to say that the
feedback improves the frequency response in the same ratio as it reduces the
gain, as in the other cases mentioned. The reason is that when the response
of the amplifier falls the frequency is well removed from the frequency design
centre of the amplifier, and at these frequencies the phase of the signal fed
back differs somewhat from the ideal of 180 deg. The feedback is thus not
100 per cent negative, though its phase may be well within the range required
for stability.
An important function of feedback so far as hi-fi amplifiers are concerned
is the effect it has on the anode impedance of the output valve. The application
of feedback does not alter the optimum load of an output valve, as is some-
times thought, though it does alter the source impedance or the impedance
as "seen" by the loudspeaker. The source impedance is decreased by the
application of negative voltage feedback or positive current feedback, and
increased by negative current feedback or positive voltage feedback.
We shall see later that a low source impedance is desirable from the
point of damping the loudspeaker so as to avoid the cone oscillating to and
fro after the application of a steep transient signal. For negative \loltage
feedback the source impedance (sometimes known as effective output
impedance or resistance) is equal to ra . For current feedback ra is
I+ AB
multiplied by the feedback factor.
DAMPING FACTOR
The ratio of the nominal output impedance of an amplifier to the
source impedance is known as the damping factor. A damping factor of 30
36
VOLTAGE AMPLIFIERS, FEEDBACK AND CONTROL CIRCUITS
is typical for modem equipment. This would be due to the nominal 15 ohms
output impedance divided by 0·5 ohm source impedance. Clearly, a low
source impedance results in a high damping factor. A number of commercial
amplifiers utilize a variable damping control so that the damping factor can
be "optimized" to suit the speaker used. However, in practice it is found that
little is to be gained by increasing the damping factor above about 20,
particularly if a good-quality speaker is correctly loaded in an enclosure.
Nevertheless, some claim that all types of loudspeaker call for different
damping factors, and that a variable means of achieving the critical damping
is essential.
One should take into consideration the resistance of the speech coil,
which is effectively in series with the damping circuit and represents the
dominant impedance at high damping factors. For example, an amplifier
having a damping factor in the region of IO will have a net damping circuit
impedance of some 13 ohms with a speaker whose speech coil measures
12 ohms. An increase in damping factor to 40 will reduce the damping circuit
impedance by less than 1 ohm, bringing it down a little above 12 ohms.
The use of positive and negative feedback loops in the output stage, and
a means of varying the positive feedback, allows the variation of the damping
factor from about 30 to infinity. An infinite damping factor means that the
source impedance falls to zero ohms; beyond this point the source impedance
becomes negative, and if the negative output impedance is greater than the
load impedance the amplifier commences to oscillate.
It must be stressed that negative feedback is not a cure for all amplifier
ills; a poorly designed amplifier cannot be made into a high-quality one simply
by applying or increasing the feedback. Indeed, as has already been intimated,
increasing the feedback on an amplifier of dubious characteristics may well
tum it into an oscillator, even if the feedback appears to be of negative mode.
The phase shift in the output transformer and coupling impedances, par-
ticularly in a not-too-costly output transformer, extends to high degrees at
the limits of the audio band. This factor severely limits the amount of feed-
back which can be applied. There are one or two methods of reducing this
phase shift which will be considered later.
Apart from instability, an amplifier of inherently high distortion will
benefit but little by negative feedback. It may well happen that the order of
the harmonic distortion may change by the application of feedback in a case
such as this; for example, second and third harmonic distortion may be
changed to fourth and ninth.

PROGRAMME SELECTION AND EQUALIZING

The section of the hi-fi amplifier which deals with voltage amplification,
programme selection, the control of volume, loudness and tone, signal
37
 THE PRACTICAL HI-FI HANDBOOK
filtering and slope of filtering is known as the pre-amplifier, or control unit.
This section may be independent of the power amplifier and connected to it
by means of a cable, as with the Pamphonic Type 2,001 amplifier, the
Pye HF25/ A and many others, or it may be an integral part of the amplifier,
as with the Pamphonic Model 1,004 and many smaller amplifiers of IO-watt
rating.
Whether independent of or integral with the power amplifier, the function
and general characteristics of the pre-amplifier are essentially unchanged.
With independent units, features in addition to the basic requirements are
sometimes embodied, additional filtering, a slope control and extra equalizing
positions being typical in this respect. Hi-fi outfits comprising separate units
are invariably more expensive than their composite counterparts. More scope
can thus be given to the designer to facilitate the development of his pet
feature; more money is available for the extra components needed and there
is more room available on the chassis since size is not restricted as is the case
with some composite units.
Most pre-amplifiers are designed to cater for four programme sources,
namely, pick-up (gram), radio, tape, and microphone. The programme
required is selected by a rotary switch (selector switch) and the signal
eventually finds its way to the control grid of the voltage-amplifier valve.
There are four input sockets, of course, corresponding to the channels
available, and the signals can be present on each of the four sockets ready for
immediate selection when required. To avoid a strong signal on a channel
which is not selected from breaking through along with the signal on the
selected channel, the sockets corresponding to the channels not in use are
sometimes short-circuited by means of an additional wafer switch ganged to
the selector switch. The Pye HF25 has such a feature.
In addition to the four programme positions, the selector switch may
also have three or four positions relating to the pick-up channel, giving six
or seven positions in all. The extra positions on the pick-up channel permit
the selection of the most suitable equalization characteristic for the record
being played.
Unfortunately, over the years, records have been cut with a diversity of
recording characteristics, each demanding a slightly or greatly modified
equalization characteristic. However, since the universal acceptance of the
R.I.A.A. (Radio Industries Association of America) recording characteristic
-known in Britain as British Standard No. 1928-there will soon be
little need for complex switched equalizing circuits. The bulk of the long-
playing repertoire in future years will have been recorded to this characteris-
tic. For the present, three or four degrees of equalization are desirable to
cater for disks which have already been cut to suit one or other of the re-play
curves given in Fig. 2.6.
38
VOLTAGE AMPLIFIERS, FEEDBACK AND CONTROL CIRCUITS
20
- . 1:-. ........
~
,..._
IS
... --...,~,- ,....
~~
lo, ~
10
~~~
'-:'~ ~
s
... ~!I:!.... h
..... :: ~
lb

-s
0
~~
.......
.. .
~ ...
-...
-10
- -R.I.A.ALP.
••••••• R.I.A A 78
',r--..'1, "~.~

..
-IS

-20
- ---E.M.I.
- -•-•-AM.COL. LP.

I I I
" ""
~-"'-
-25
so 100 500 1000 5000 10000
FREQUENCY • c/s
FIG. 2.6. Equalization curves in current use.

For reasons closely related to the design of pick-ups and signal-to-noise

ratio on re-play, disks are cut with a rising high-frequency response and a
falling bass, the opposite to the curves in Fig. 2.6. The reasons for this are
given in a later chapter. Clearly, in order to secure a straight-line frequency
response, equalization curves must be the opposite of the record curves.

PRE-AMPLIFIER FIRST STAGE

Let us now investigate a circuit providing the functions so far outlined.

In Fig. 2.7 is given the first-stage circuit of the Pamphonic Model 2,001. In
this, the circuit arrangement associated with the selector switch clearly
follows the lines laid down in this chapter. The signal, after selection by
SWIA, is applied to the grid of VIA by way of R2 (valve VI being of the
double-triode variety, Mullard ECC83). Pre-set controls are available on the
pick-up, radio and tape channels so that the signal level can be pre-adjusted
to avoid overloading the first stage where the programme-source signal is
of high level.
The amplified signal appearing across R4 is coupled capacitively to the
grid VIB through C2 and CIO. A portion of the signal appearing across R4
is also fed back to the grid of VIA by way of the switched resistor and
capacitor networks. This results in selective attenuation due to negative
feedback and thus provides various degrees of equalization as selected by
switch SW I B, which is ganged to the selector switch SW I A.
Bass correction is controlled by capacitors C6 and CS (bass boost) with
39
 THE PRACTICAL HI-FI HANDBOOK
.----------2'-s_o_v_ _ _ _ _ H.T.+
NOTE . ALL VOLTAGES MEASURED WITH
AVO MO!)El 8 :t; C9 Rl2
HT VOLTAGES ON 1000 V RANGE R4
CATHODE VOLTAGES ON 25V RANGE
C2 CIO Cl2
MIC L...-_SIGNAL
.------- OUT

30V

Rl3

MIC 04-e oTAPE

L pO ORADIO
NA8007s
FREQUENCY-SELECTIVE
SELECTOR SWITCH POSITIONS FEEDBACK LOO<>

FIG. 2.7. The first-stage circuit of the Pamphonic Model 2,001 amplifier.

a very low-frequency roll-off, due to the progressive reduction in feedback

at the lower frequencies. Capacitors C2 and C3 are concerned essentially with
the coupling of the loop and have little material effect on the response, being
of high value-32 and 50 mF respectively. Capacitors CS and C7, along with
their associated resistors, control the top-cut. A certain degree of fixed
correction is also given by RS and C4.
The arrangement does not apply any appreciable amount of negative
feedback to the programme circuits, owing to the isolating resistor R2, and
it represents one of the most popular equalizing circuits in hi-fi use today.
It is extremely quiet, stable and efficient in that the unwanted gain is employed
as a distortion-reducing agent.
On the "mic", "tape" and "radio" positions, the feedback is not
frequency-selective, being controlled by resistors R7 and RIO respectively.
Since high gain is required on the microphone channel relative to the other
channels, the feedback is considerably reduced by R7 being of much greater
value than RIO, and by R7 and RIO being in series when the selector switch
is set to "mic". Thus, apart from providing equalization, the negative
feedback can be switched to facilitate a balanced output in spite of the
various voltages on the four channels.
The equalized signal is further amplified by VIB, and appears in this
form across the load resistor Rl2. From here it is conveyed through Cl2 to
the following section of the pre--amplifier circuit, which is usually the tone-
control section.
The circuit in Fig. 2. 7 is designed to match or load most magnetic
40
VOLTAGE AMPLIFIERS, FEEDBACK AND CONTROL CIRCUITS
pick-ups of the hi-fl type. It must be stressed that every pick-up, irrespective
of type, has an optimum load which provides the correct curve to work into
the equalizing network as stipulated by the makers. With magnetic pick-ups,
a load larger than the optimum results in an increase in high-frequency
output, while an increase in low-frequency output occurs with crystal
pick-ups.
The practice of incorporating facilities for correct impedance-matching
of various pick-ups is increasing in popularity among amplifier manufac-
turers. The Armstrong and R.C.A. units adopt two input sockets, one for
magnetic pick-ups and the other for crystal types, which can be switched
independently. The Pye Mozart, on the other hand, features two controls
(pre-set), one for providing attenuation and the other for matching. This is
known as "Dialomatic Pick-up Compensation", and permits immediate
matching for any pick-up. A list of settings for a very wide range of pick-ups
is given in the instruction manual. A similar idea is used on the Decca FFR25.
Other manufacturers use plug-in matching units.
Some simple equalizing networks rely on the fact that the response of the
pick-up is altered with alteration in load resistor.

TONE CONTROL
The most popular tone-control system, giving independent control of
both bass and treble, is that due to P. J. Baxandall ( Wireless World, October,
1952). The circuit of the network is given in Fig. 2.8, from which will be seen
that it is focused around the triode valve V l, this usually being one half of a
double-triode. Basically, the operation of the circuit relies upon frequency-
selective negative feedback, the feedback loop being by way of the anode of
- - - - - HT+

VI
Rl 47OK
R4

1·2K
cs IOOp

MAX MIN
R6
SOOK 3·9 K
TREBLE

FIG. 2.8. Baxanda/1 tone-control circuit, with typical component values.

41
 THE PRACTICAL HI-FI HANDBOOK
the valve, through C4 and the resistor/capacitor network, and back to the
grid circuit through R4.
The overall control is formed by the amalgamation of two somewhat
complex independent treble and bass control circuits. In order to secure both
treble lift and treble cut, the treble control has a tapped resistive element
connected to chassis. The treble-lift elements are C5 and the section of the
treble control connected across chassis and the junction of Cl, RI, while the
treble-cut elements are C5 and the section of the treble control connected
across chassis and the junction of R2, C4. Bass lift and cut is given by the
bass control in association with C2, C3, RI, R2, R3 and back to the grid
by way of R4.
The lift and cut is confined to each end of the audio spectrum, thus
permitting extreme frequencies to be lifted and cut to a large degree without
disturbing the response at the centre of the spectrum.
There are a diversity of tone-control circuits which rely simply on
filtering the signal to varying degrees to provide the necessary boost or cut
either side of the spectrum. Such an arrangement is sometimes called a
"passive tone control". Two simple resistor-capacitor networks of this kind
are shown in Fig. 2.9; circuit (a) providing treble control and circuit (b) bass
control. A combined treble- and bass-control circuit is also a fairly common
set-up, and one which in some quarters is held in favour over the feedback
arrangement, possibly owing to the greater flexibility of response over a
wider frequency range.
Passive networks can be inserted between two voltage amplifiers since,
in common with all tone controls and equalizers, inevitable attenuation
results from the circuit, and this must be made good by additional amplifica-
tion. Generally speaking, it is not a good idea to include the network in a
low-level stage, such as in the input circuit of the first voltage amplifier, for
the reason given above. This may not apply, however, to pick-up equalizers
if the pick-up used has a large output voltage.

SIGNAL
IN

TREBLE ---1 SIGNAL OUT

( al (bl

FtG. 2.9. Two simple resistor-capacitor passive tone-control circuits: (a) providing
control of treble and (b) control of bass.
42
VOLTAGE AMPLIFIERS, FEEDBACK AND CONTROL CIRCUITS

db ...... ,
, MIN
-5 , SLOPE
',/
-10
''
• 15
MAX
SLOPE
' \
\
/ \
-20 '\ \
\
-25

-30

2 3 4 5 10 20 30 40 50 100
kc/s

Fm. 2.10. Two-positio:i filter characteristic with a slope control.

After suitable amplification by the valve following the tone-control

network, the signal is usually fed into a filter system.

FILTERS
A filter is included to avoid the amplification of noise due to worn
records and whistles caused by inter-channel interference when the hi-fi
equipment is used with an A.M. radio tuner. Since the noise and interference
frequencies are focused towards the high-frequency end of the audio spectrum,
the filter is arranged to have a steep treble cut.
Most pre-amplifiers have a four-position filter switch, giving three
positions of treble cut at 4 kc/s, 7 kc/sand 12 kc/s, and a "filter out" position.
In addition, a control designated "slope" is often incorporated whose purpose
is to vary the rate of treble attenuation from a minimum of some 8 db per
octave to a maximum approaching 35 db per octave. The idea is shown
graphically in Fig. 2.10. Here two filter positions are available, one at
6 kc/s and the other at 8 kc/s. The broken-line curve shows how the slope
control serves to affect the rate of attenuation. Maximum slope indicates
maximum attenuation rate.
When using new disks of recent pressing, it is desirable to commence
operation with the filter switched out of circuit and with the bass and treble
controls at "level", thus giving an extended flat response. Then, as governed
by the acoustical environment, the bass and treble controls should be
adjusted, bearing in mind that the ear is the final arbiter, as distinct from
numbers on a dial!
43
 THE PRACTICAL HI-FI HANDBOOK
0·45H 0·1 0-1 0·1
lOK
---I
SIGNAL 0-001 0-001 SIGNAL SIGNAL SIGNAL
IOK OUT IN 470K HOK 470K OUT
IN "T"

. 1
(Left) FIG. 2. 1 I. Capacitive-inductive /ow-pass filter. (Right) FIG. 2.12. FAch coupling
section provides a /ow-frequency roll-off at the rate of 6 db per octave; 18 db are
given by the three cascaded couplings shown.

With worn records, the overall performance can be enhanced with the
filter adjusted to give a cut-off at 4 kc/s or 7 kc/s coupled with the application
of a little top-boost by the treble control. The same reasoning usually applies
to noisy radio programmes. With music of a high transient content, it often
pays, if a filter position is called for, to reduce the rate of attenuation by the
slope control. This avoids "overhang" and "ringing" at high frequencies.
Filters come in two types. First, there is the tuned inductor arrangement
in which an inductor connected in series with the signal source is resonated
by capacitors. This is illustrated in Fig. 2.11. The circuit is tuned so that a
sharp dip occurs at the high-frequency end of the response, and the falling
side of the curve represents the treble-cut effect. At resonance, the circuit
offers a very high impedance to signals at that frequency. The capacitors
are usually switched, thus providing various filter frequencies, while the
control of slope is by damping the circuit with the resistor. A variable
resistive element permits a variable control of slope, as already explained.
Secondly, there is what is known as the "parallel T" circuit. This requires
a large number of low-tolerance resistors and capacitors in order to give the
desired high rate of attenuation at the various filter frequencies. Such a
network is used in the R.C.A. pre-amplifier.
Combination circuits are also used, as also are less elaborate resistance/
capacitance networks in pi and M-derived configurations. It should be noted
that resistance/capacitance inter-stage couplings affect the frequency response,
but at the low-frequency end. A sharp cut at a low frequency is often
desirable for eliminating gram motor rumble, and for avoiding unnecessary
low-frequency distortion, particularly where the programme material
possesses excessive low-frequency signal and the amplifier is not too good at
the low-frequency end of the spectrum. It is far better to cut off sharply at
about 40 c/s and achieve a "clean" bass than endeavouring to extend the
response down to about 20 c/s and create unnecessary distortion.
Rumble filters, as these devices are often called, are switched on some
units, giving only the positions "filter on" and "filter off". They are usually
simple in design, often being built into an inter-stage coupling. A single
44
VOLTAGE AMPLIFIERS, FEEDBACK AND CONTROL CIRCUITS
resistor/capacitor coupling gives a roll-off of 6 db per octave. Since this is
hardly steep enough for the purpose in hand, however, cascaded couplings
are favoured. As each coupling gives the standard 6 db per octave roll-off,
a slope of 18 db per octave is achieved by the use of three networks in cascade
(Fig. 2.12).
The signal leaving the filter may be taken direct to the output socket of
the pre-amplifier unit, or to another valve for further amplification and to
facilitate matching to the power amplifier.

PRE-AMPLIFIER FINAL STAGE

A triode valve connected as a cathode follower is favoured as the final

stage in the pre-amplifier unit. The advantage of the cathode-follower in this
application is the low impedance across which the output signal is developed.
This allows relatively long pre-amplifier /power amplifier connecting cables
without causing undue attenuation of the higher audio frequencies, whilst
also minimizing the pick-up of spurious signals, such as mains hum.
A cathode-follower stage is shown in Fig. 2.13. It will be seen that the
load impedance is connected between cathode and chassis instead of between
anode and h.t. positive, as in the more conventional arrangement. The
cathode loading feature results in 100 per cent negative feedback, and as a
consequence the distortion developed by the stage is at extremely low level,
as also is the gain, being less than unity. This is of little moment, however,
since adequate signal is usually available at the output of the filter, and the
stage serves admirably as a matching device, for apart from its low output
impedance, it has a very high input impedance and thus has little shunting
effect on the circuit to which it is connected.
In the circuit in Fig. 2.13, Rg is the normal grid resistor, C the coupling
capacitor, RI the load and Rk the ordinary cathode-bias resistor.
The pre-amplifier volume control is invariably connected between the
output signal at the filter and the cathode-follower valve grid.
There is still considerable controversy regarding the merits and demerits
of the loudness control. It seems to be an
accepted feature in the United States, ------ H T +

though in Britain its use is by no means

universal. It is not a new device, having C
~~GNAL+-1
been used many years ago in the form of a
tone-compensated volume control in broad- SIGNAL
OUT
cast receivers.

FIG. 2.13. The cathode-follower. The stage gain

is negative, being less than unity.
45
 THE PRACTICAL HI-Fl HANDBOOK
120 Fm. 2.14. Fletcher-
. PHONS
Munson equal-
~
u 100 - 100 loudness curves

- -,, show how the ear's

'!
..,
z
~ 80
-
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response varies
with intensity.

...
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30 50 70 100 300 500 700 1000 3K SK 7K I0K
FREQUENCY • c/s

It will be recalled from Chapter l that the ear does not respond in the
same way to all frequencies. How the ear responds over the audio spectrum
is revealed by the Fletcher-Munson equal loudness curves (Fig. 2.14). These
curves show the required relative levels of sound at various frequencies
for a sensation of equal loudness, being based on the reference frequency
of 1,000 c/s. It will be recalled that the loudness of sound in phons is numeri-
cally equal to the sound intensity in decibels of an equally loud 1,000 c/s note,
and that zero phon (corresponding to zero db at 1,000 c/s) is equal to a sound
pressure of 0·0002 dyne per square centimetre (this, incidentally, is equal to
10·1s watts per square centimetre).
The curves demonstrate clearly that at low-level listening a considerable
bass lift, and to a lesser degree top lift, is demanded in order to secure the
sensation of the same apparent loudness over the full frequency range. There
is little doubt that reproduction at a level of some 5 watts, with the treble and
bass control adjusted to suit the room conditions, is of a far superior quality
to that at 500 milliwatts with the tone controls left at their original settings
-this condition can be created simply by backing-off the ordinary volume
control. To secure anything like the original balance, considerable bass lift
and a small amount of top lift is essential.
To avoid having to make these tone-compensating adjustments every
time the volume control is adjusted, the loudness control has been evolved,
and is designed around the Fletcher-Munson curves to quite a high degree
46
VOLTAGE AMPLIFIERS, FEEDBACK AND CONTROL CIRCUITS
of accuracy, at least from the low-frequency point of view. Retarding the
control provides automatically the required degree of bass lift.
Two loudness-control circuits are given in Figs. 2.15 and 2.16. The first
is arranged around a tapped volume control to which are connected capaci-
tive elements. With the control in the maximum position the capacitive
reactance shunting is at minimum, but as the control is rotated towards the
minimum position the shunting increases progressively and the higher
frequency components of the signal are attenuated with respect to the low
frequencies, which effectively provides a bass lift. The cross-over point is
somewhat governed by the resistance R.
The negative-feedback scheme in Fig. 2.16 makes use of an ordinary
volume control and a frequency-selective negative-feedback loop by way of
capacitor C. The loudness control and resistor R form a potential-divider
in the feedback circuit, feedback being at maximum at the minimum position
on the loudness control. Thus, as before, when the loudness control is
retarded towards minimum the feedback of the higher audio frequencies
increases, and a progressive boost of bass results.
It is usual to employ an ordinary volume control as well as a loudness
control, the two controls often being connected in cascade, though in some
cases they may be independently positioned in different stage couplings.
Their actions are somewhat related, and for this reason their settings should
be established with some care so as to avoid over-emphasis of the bass,
possibly falling outside the range of the bass control proper.
The following procedure should be adopted where possible: turn the
loudness control to maximum, or switch it out of circuit completely if a
switch is provided for this purpose; set the volume control to a fairly high
level; balance the sound to suit the room acoustics by means of the bass and
treble controls; reduce volume to normal room level by backing-off the
loudness control. A reasonable balance should be maintained throughout
SIGNAL --------1.,___ __
IN ----.-,

(Left) FIG. 2.15. Loudness control formed by tapped frequency-selective volume

control. (Right) FIG. 2.16. Negative-feedback loudness control.
47
 T-'PE RECORD

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FIG. 2.17. Complete circuit of the Pye HF25A pre-amplifier. To ensure accurate voltage
reading a high-resistance voltmeter (more than 10,000 ohms per volt) must be used.
VOLTAGE AMPLIFIERS, FEEDBACK AND CONTROL CIRCUITS
the range of the loudness control over ordinary volume levels if this process
is followed. Severe bass distortion will result, however, if the loudness control
is turned to minimum, and the room-level volume adjusted solely by the
volume control.

PRE-AMPLIFIER COMPLETE CIRCUIT

Fig. 2.17 is a complete circuit of the Pye HF25A pre-amplifier, and it

will be instructive to look at the various sections in the light of our previous
discussion. The required signal source is selected by switch SIA and applied
to the grid of the first voltage-amplifier valve VIA. The unused programme
sources are short-circuited by switch section S 1C to avoid breakthrough, and
a variable attenuator is included in the radio channel input so that any high-
level signal on this channel can be suitably reduced to prevent overloading
of the first valve.
The input selector switch is ganged to the equalizer switch SIB and four
positions of equalizing are available on the pick-up channel on settings 4, 5,
6 and 7, negative feedback being used for this purpose and for controlling
the gain over the four input channels. The equalized signal is further amplified
by V1B, and control of bass and treble is secured by reason of the Baxandall
negative-feedback system in conjunction with valve V2A. The signal is then
passed into a three-position filter circuit, giving a sharp treble cut at 4 kc/s,
7 kc/s and 12 kc/s. The filter is adjusted by switches S2A, S2B and S2C,
which also give a "filter out" position. The filter is of the inductive-capacitive
type, with LI as the inductive element.
The signal at the output of the filter, at the rotor of S2C, is passed
through the volume control and on to the final valve V2B, which is arranged
as a cathode follower. The signal is finally conveyed across R31 to pin 5 on
the octal output plug.
A handy feature, and one which is found on many control units, is the

•20
MAX MAX
+ 15
+10
+5
db O

-5
·10
-15
-20
-25
10 100 1000 10 000 20000
BASS els TREBLE

FIG. 2.18. Tone-control characteristics of the Pye HF25A pre-amplifier.

49
 THE PRACTICAL HI-FI HANDBOOK

0
db "
5
-10
-15
-20
-25
-30
-35
-40
-45
-so
20 100 1000 10000 20000

F1a. 2.19. Filter characteristics of the Pye HF25A pre-amplifier.

"tape record" output socket. At this point appears the amplified signal
voltage exclusive of filter influence. This can be used for feeding another
amplifier, if required, or used as an input signal for a tape recorder. Both
h.t. and l.t. for the unit is derived from the main power amplifier, which is
the subject of the next chapter.
Tone control and filter characteristics of the unit are given in Figs. 2.18
and 2.19 respectively.
To summarize, the pre-amplifier serves to match into the various signal
sources so as to secure maximum signal transfer and signal-to-noise ratio, to
equalize for the shortcomings of the programme material, to provide control
of volume (sometimes loudness) and tone, and to raise the low-level pro-
gramme signals to a level of about I ·5 volts for application to the power
amplifier. To do this the pre-amplifier invariably requires an overall gain of
some 60db.

50
 CHAPTER 3

The Power Amplifier

ELECTRICAL power measured in watts (W) is the product of

the voltage (£) and current (/), that is, W = E x I. Thus, for a constant
power we can either increase the voltage and decrease the current or decrease
the voltage and increase the current. At the output of a pre-amplifier we have
in relation to the very small signal current a rather high voltage-because at
the pre-amplifier stage we are mainly concerned in obtaining voltage ampli-
fication. However, at the output of the power amplifier we need to produce
a larger current, since this is what is wanted by the loudspeaker. In fact, a
signal current of I amp. is required to flow through a 15-ohm speaker to
represent a power of 15 watts. This means that 15 volts would be measured
across the speaker at this power. This is considerably different from the few
millionths of an amp. of signal current which is present in the final load of
the pre-amplifier.
The valve is inherently a high-resistance device, which means that the
signal current in the anode circuit is limited to a few thousandths of an amp.,
though, in order to secure this condition, the voltage for a given power is
relatively high. Clearly, a means is necessary for transforming this high signal
voltage and low current so that the current is considerably increased and the
voltage proportionately decreased for presentation to the loudspeaker. This
transforming process is achieved quite easily by the use of a transformer-in
this application it is termed the "output transformer".
The winding connected to the anode circuit of the valve, across which is
developed the high signal voltage, is known as the primary winding, and is
composed of a large number of turns of fine wire since it is required to pass
only a small current. The winding connected to the speaker, through which
is passed the high signal current, is known as the secondary winding, and is
formed of only a few turns of heavy-gauge wire. The arrangement is repre-
sented in Fig. 3.1.
A transformer devoid of losses has a voltage ratio equal to the turns
ratio, that is, Voltage across primary: Voltage across secondary = Primary
turns: Secondary turns. Thus, a transformer with a 10:1 turns ratio will have
51
 THE PRACTICAL HI-Fl HANDBOOK

t
LOW CURRENT
----Hf•

HIGH VOLTAGE

i
o,. . . ./4
LOW IMPEDANCE

:_ 'ilGH CURRENT
(Z2)

J
FIG. 3.1.Matching the speaker to
the output stage is an important
consideration in hi-ft practice.

LOW VOLTAGE

TIJRNS RATIO: ,/Zlf':.2

IO volts induced across the secondary when a signal of I00 volts is applied
across the primary. The current will be transformed in the same ratio,
assuming ideal conditions.
Across the secondary winding of the transformer (Fig. 3.1) is connected
the resistance or impedance (impedance is the term used when we are dealing
with alternating quantities) of the loudspeaker. This impedance-let us call
it 22-is reflected across the primary winding, but is altered in magnitude
according to the expression (Z~) 2
x 22, where NI is the number of turns
forming the primary winding, and N2 the number of turns forming the
secondary winding.

to primary), the factor z~

Thus, if the transformer has a turns ratio of 10:1 (step-up from secondary
resolves to IO, the square of which is 100. Now,
if the impedance of the loudspeaker (22) is equal to IO ohms, then the im-
pedance reflected across the primary is 100 x IO=l,000. This is the impedance
as "seen" by the valve. It is rather important to get these facts clear, as they
have a direct bearing on the performance of the power amplifier.
The transformer, therefore, apart from altering the voltage and current
transfer, also serves as an impedance-matching device. In the case cited, the
transformer of ratio l O: l has transformed the l 0-ohm loudspeaker impedance
to 1,000 ohms. Impedance-matching in the output stage is an extremely
important consideration, for unless the impedance of the loudspeaker is
transformed to match exactly the optimum working impedance of the valve
or valves, the loudspeaker will not receive the full available power.
Re-arranging the expression given above, we can discover the turns
ratio required to match the speaker impedance 22 to the high impedance
21.
.
The turns ratio = . I -
/21
'V 22
Most power amplifiers have switching or plug-and-socket facilities
available for adjusting the output impedance of the amplifier to suit a wide
52
 THE POWER AMPLIFIER
range of speaker impedances. As the fraction of voltage fed back as feedback
is affected by impedance adjustments, it is desirable to alter the value of the
feedback resistor (and sometimes the capacitor) when the output impedance
is changed. Commercial amplifiers have indicated in their instruction books
the value of components required for the various output impedances. Where
a positive-feedback loop is incorporated, this also requires adjustment when
changing impedance.
A power amplifier should never be operated with the speaker removed
unless a resistor of equal value to the speaker impedance is used instead. An
unloaded amplifier wi11 develop very high audio peak voltages across the
primary of the output transformer, and these can quickly destroy the
transformer insulation.

OUTPUT TRANSFORMER
In practice, there is no such thing as a perfect transformer, since losses
occur in both the primary and secondary windings and in the core itself.
The inductance of the primary governs the relative amplification at low audio
frequencies, while leakage inductance in both windings results in a loss of
high audio-frequency response. These losses also promote phase shift at the
low- and high-frequency sides of the audio spectrum, and sometimes make it
difficult to maintain a reasonably high degree of negative feedback without
instability.
The windings also possess distributed capacitance, and due to this there
is invariably a peak in the response curve at the frequency at which the
equivalent leakage reactance resonates with the Jumped equivalent
capacitance.
In transformers designed for hi-fl work, all these losses are kept at the
absolute minimum, resonances are damped and arranged to fall outside the
audio spectrum, considerable iron is used for the core so as to maintain
adequate low-frequency response, and distributed capacitances are minimized
by dividing the windings into sub-sections, as shown in Fig. 3.2. This
specialized design is inevitably reflected in the
cost of the item, and is the reason why the out-

~~~l:r;::;::;:t ~: ;~;:~~~ a!;1i;;;_s~;;:t~~

much truth in the saying that an amplifier is only
_!~.
;
. _~vi=s~"' _!-,,.
;
_Is~,,. _a:a:;:-..,
:
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as good as its output transformer.

FIG. 3.2. Distributed capacitances in the output trans-

former are minimized by dividing the windings into
sub-sections, as illustrated. SPEAKER

53
 THE PRACTICAL HI-Fl HANDBOOK
R6

T2 226-250•

RI

l7 AC MAINS

FIG. 3.3. Circuit diagram of the Cossor 562K amplifier kit. The component values
are given below.

RI 47K RVI 500K log. Cl 5,000pF ±20%

R2 IOOK RV2 500K log. (with SI) C2 0·05µ.F 250V
R3 2·2K Note: All resistors ± 10% C3 50µ.F 12V
R4 100 ohms C4 50µ.F 275V
R5 330K cs 50µ.F 12V
R6 3·9K C6 5,000pF ±20%
R7 5·6K C7 2µ.F 150V
RS 150 ohms cs 50µ.F 275V

Hi-fi transformers need to be bulky devices in terms of iron in order to

permit the full rated power of the amplifier to be handled down to about
30 c/s, and to avoid harmonic distortion due to core saturation. The core
material affects the primary inductance, which needs to be relatively high for
an adequate low-frequency response for the smallest possible amount of wire.
Both the cross-sectional area of the core material and its permeability are
governing factors on the inductance; increasing either of them results in an
increase in inductance.
In recent years a core material having a higher permeability value than
the conventional "Stalloy" laminations has been under development. This
is a cold-rolled silicon steel, having grain-orientation properties, which is
known as "Unisil". For a stated low-frequency performance, a transformer
with this new core material is approximately half the weight of a transformer
using the older grades of core material. The well-known "C-core" is the result
of further developments along these lines and, although somewhat more
54
 THE POWER AMPLIFIER
expensive than "Stalloy" laminations, permits even further reduction in both
weight and size.

SINGLE-ENDED OUTPUT STAGE

To illustrate the single-ended output stage, the circuit diagram of the
Cossor Amplifier Kit 562K is given in Fig. 3.3. This is an ideal kit, easily
made up by the non-technical experimenter, with which to study the rudi-
ments of sound reproduction. It is not claimed to possess hi-fi quality, but in
spite of its basic simplicity, the performance may pleasantly surprise even one
having long association with hi-fi equipment, especially if used with a first-
class loudspeaker.
The first valve, VI, serves as the voltage amplifier, the signal being applied
to the control grid by way of the volume control RV2, after correction by the
tone control RVI. The signal in amplified form is developed across R2, and
from here fed to the control grid of the output valve, V2, through the
coupling capacitor C2. The output valve is biased in the ordinary way by the
cathode resistor R8. The volts drop across this resistor due to the current in
the valve causes the cathode to rise positively with respect to chassis. Because
the control grid of V2 is in d.c. connexion with the chassis through the grid
resistor R5, the grid is effectively more negative than the cathode and is thus
biased, the magnitude of the bias being equal to the volts drop across the
cathode resistor R8.
This is the normal function of so-called cathode bias, which is used
extensively in radio and television receivers as well as audio equipment.
The valve is biased to class "A" conditions. This means that in the
normal condition of operation the anode current is not cut-off for any portion
of the cycle of signal applied at the control grid.
The signal in the anode circuit is conveyed to the loudspeakers, LS I and
LS2, by way of the output transformer Tl, impedance matching being
achieved in the way already described. The tapping on the primary winding
of Tl may require explanation. The portion of the winding across which the
signal voltage is developed is that across the capacitor C6. The section
between the tap and R6 serves as a hum-neutralizing device, h.t. from
the cathode of the rectifier valve V3 being applied to VI through the
upper part of the primary and R6. Thus, this part of the primary can,
in effect, be considered as a smoothing choke with the added filtering
contributed by R6, in association with the reservoir capacitor C8 and
smoothing capacitor C4.
Negative-voltage feedback is applied over the two stages from the
secondary of the output transformer through R 7 to the cathode circuit of
VI. The feedback loop is completed from the other side of the secondary
winding to chassis. This is one of the most popular ways of applying negative
55
 THE PRACTICAL HI-Fl HANDBOOK
feedback. (Of course, if the feedback and chassis connexions on the secondary
winding are reversed, the feedback becomes positive and the amplifier turns
into an oscillator. Such a mis-connexion could occur in changing an output
transformer of this kind.)
It will be seen that two loudspeakers are used. The one connected
directly across the secondary deals with the lower and middle audio fre-
quencies; the one connected by way of C7 handles the higher frequencies, and
is often referred to as a .. tweeter" or high-frequency reproducer. It is some-
what isolated from the lower audio frequencies owing to the high reactance
presented by C7 to these signals. In effect, the arrangement can be considered
as a simple cross-over filter, and will be dealt with in more detail in a later
chapter.
PENTODE OR TRIODE
In the early days of hi-fi, the triode was invariably held in favour over
the pentode (or tetrode) owing to its smaller distortion figure and lower anode
impedance as compared with the pentode. For example, at maximum output
a power triode in a single-ended circuit may produce something like 5 per
cent distortion, mainly second harmonic.
The distortion due to a power pentode operating similarly, however, may
rise to some 13 per cent, made up basically of third harmonic and higher
order harmonics. The reason for such a high distortion figure is the result of
the S-shaped grid voltage/anode current characteristic curve of the pentode.
This characteristic is shown in Fig. 3.4, and the general picture is completed
by a sine wave corresponding to the signal applied at the control grid, the
valve being biased to class A, and the resulting signal across the anode load.
It will be seen that the bends at the top and bottom of the curve cause a
certain flattening of the peaks of the amplified signal, which promotes a
spurious third-harmonic signal. An analysis of the distorted signal would
reveal that it is composed of the fundamental component plus a smaller
third-harmonic component and
higher order harmonics. ANODE CURRENT
Fig. 3.5 shows that the grid
voltage/anode current characteristic SIGNAL ACROSS
/ ANODE LOAD
curve of the triode is considerably BIASED To
more linear than that of the pentode. CLAss A ..._

FIG. 3.4. The S-shaped characteristic

curve of the power pentode results in
the production of third-harmonic dis-
tortion, shown by the flattening of the
SIGNAL APPLIED
peaks of the output signal. AT GRID

56
 THE POWER AMPLIFIER
FIG. 3.5. The triode has a more linear ANODE CURRENT
curve, and the resulting distortion is
essentially second-harmonic.

There is a slight inward bend,

however, and this results in slightly
GRID VOLTAGE +
greater amplification of one half-
cycle of the signal than the other.
This is representative of second-
harmonic distortion. The triode, SIGNAL APPLIED
on the other hand, is less sensitive AT GR1D

than the pentode, requiring a

greater input signal for a given output power. However, if the drive is
exceeded the distortion rises rapidly, and can be very disconcerting if this
happens on programme peaks in a poorly designed amplifier, or if the ampli-
fier is insufficiently large for the purpose in hand.
A greater overload margin is available with the pentode, and to a smaller
degree with the tetrode, owing to the "cushion" effect created by the gradual
bends at the ends of the curve. Unfortunately, the application of negative
feedback tends to neutralize this desirable effect, and the valves behave more
like triodes with regard to accidental overload.
In order to prevent power being wasted in the output valve, the load
impedance should theoretically match the anode impedance of the valve
In practice, this does not always follow because the load for maximum power
does not coincide with that for minimum distortion. There is, therefore, an
"optimum load" value which gives a compromise between the two conflicting
factors.
Fig. 3.6 shows curves for power output and percentage distortion
against anode-load impedance, and these clearly reveal the need for an impe-
dance compromise. If maximum power output is aimed at, then both
second- and third-harmonic distortion will be at a high level. If the load is
decreased to 4,000 ohms, third-harmonic distortion is Jess troublesome but
second-harmonic rises, accompanied by a power decrease. Generally speaking,
second-harmonic distortion is Jess disturbing than third-harmonic and, as we
shall see later, can be eliminated in a push-pull output stage. Thus, it is
desirable to work to the best compromise between third-harmonic distortion
and power output. Neither the technician nor the enthusiast need worry
about working out optimum loads, since this has already been done by the
valve maker, and the figures can readily be obtained by referring to the data
supplied with the valve.
Similar curves for a triode power valve are given in Fig. 3.7. Here there
57
 THE PRACTICAL HI-Fl HANDBOOK
3,-----------------------, FIG. 3.6. Curves,
related to a pen-
tode output valve,
for power output
...
Ill
•:.:;'s~r_o_Rr_,_o_N_ __
~ and distortion
u
~otilC _0
i 1~11>-0
~,.v- 6
~ plotted against
anode-load
4£z impedance.
2~ "'
Ill
0
6000 7000
ANOOE LOAO OHMS

is only second-harmonic distortion to deal with, and it is desirable to use a

large-value load if minimum distortion is required.
Account must be taken of the fact that in practice the impedance
presented to the output valve changes considerably over the operating
frequency range owing to the varying impedance of the loudspeaker. More-
over, as the load is not a pure resistance, but possesses a reactive component,
the task offinding the best load compromise is made somewhat more difficult.

PUSH-PULL OUTPUT

The push-pull output stage is usually a feature of hi-fi amplifiers, and the
basic circuit is shown in Fig. 3.8. The signal is applied to the two valves in a
way that when the control grid of one is swung positive the grid of the other is
swung negative, and vice versa. The anode current is thus rising in one valve
while it is falling in the other; hence the term "push-pull".
The signal is applied in relation to a fixed "zero signal" point, such as
chassis, either from a transformer with a tapped secondary winding or from a
phase-splitting valve. In this way, not only is the power output increased but,
more important from the hi-fi aspect, all second and even harmonic distortion

...z
"'u
Ill
...
IOffi

FIG. 3.7. Curves, ~ z

0
related to a triode 3::
;::
output valve, for 20
·
50
...""
Ill
power output and 0
distortion plotted
against anode-load 1·6 ~ - - ~- - -~ - - -~ - - - , ~- - -~ - - -- '
0 1000 20 00 3 000 000 5 000 60 0
i
impedance. ANOOE LOAD. OHMS

58
 THE POWER AMPLIFIER
is automatically cancelled. Thus, a properly designed and adjusted push-pull
stage should produce virtually zero second-harmonic distortion. Third and
odd harmonic distortion will still be present, of course, but this can be
reduced to a very low figure by the application of negative feedback. Even
without feedback, a triode push-pull stage will exhibit a very low distortion
figure, bearing in mind that second harmonic is the prominent factor in each
valve, and will be eliminated in a push-pull stage. This is one of the reasons
why triodes used to be very popular a few years ago, before negative feedback
was fully developed.
In hi-fi application, the output valves are arranged to operate in class A,
though this is by no means a necessity, for if greater power is called for, and
an increase in distortion is permissible, then the valves may be biased towards

H.T. •

Cl VI
SIGNAL I N ~ - - - - . -

§.o
~SPEAKER

i OUTPUT

SIGNAL IN -------+
C2

V2
TRANSFORMER

Fm. 3.8. Basic circuit of push-pull output stage using tetrode valves.

anode-current cut-off and the signal may be increased to push the valves
into grid current on the positive half-cycles. In this way a very much larger
signal swing occurs in the anode load, and the output power is correspondingly
increased. This scheme is used extensively in public-address and sound-
reinforcement amplifiers, where quantity rather than quality is demanded.
In most hi-fi power amplifiers the output valves are biased by reason of
the volts drop across their cathode resistors (R3 and R4 in Fig. 3.8). The
variable potentiometer serves to adjust the two valves for d.c. balance.
In other amplifiers, particularly those whose operating conditions deviate
from pure class A, a separate bias line is used, sometimes derived from a
separate bias power unit.
59
 THE PRACTICAL HI-FI HANDBOOK
CENTRE-TAPPED SIMPLE COUPLING
/TRANSFORMER /' TRANSFORMER

GML
~T ORIVER
RI

R2
PUSH-PULL
OUTPUT

ORIVER
VALVE VALVE

H.T • HT+
( a) (b)

FIG. 3.9. Phase-splitting (a) by means of a centre-tapped transformer, and (b) by means
of a simple transformer and resistive divider.

PHASE-SPLITTING
The simplest method of providing two input signals 180 deg. out of
phase for a push-pull stage is by means of the centre-tapped transformer
(Fig. 3.9a). An alternative arrangement, which uses a simple coupling
transformer, is shown in Fig. 3.9b. In both cases the secondary of the trans-
former provides two equal anti-phase signals. In the latter case, however, a
zero-signal reference point is obtained at the junction of the resistive divider
(R l and R2). If required, the transformer can give a signal step-up to the
push-pull valves and, as with the output transformer, it will also serve to
match the anode circuit of the driver valve to the grid circuits of the push-
pull valves.
In arrangements which deviate from true class A working, the secondary
of the transformer is usually placed in the grid circuits of the output valves.
The reason for this is to provide a relatively low d.c. resistance in the grid
circuits as a means of avoiding severe overload distortion when the output
valves are driven into grid current. Transformer phase-splitting is thus usually
confined to large public-address and sound-reinforcement amplifiers.
Since a transformer inevitably introduces some degree of distortion,
transformer phase-splitting is rarely if ever used in hi-fl amplifiers. Instead,
a method of phase-splitting by means of a valve arrangement is always
favoured.
A very popular valve phase-
splitter circuit is shown in Fig. 3.10. RI
In this circuit, which is designed s1GNAL
PUSH-PULL
around a triode or a pentode IN --- OUTPUT
HT •
strapped as a triode, the load R4
resistance is split; half of it is con-
Cl
nected in the anode circuit (R l)
FIG. 3.10. Circuit diagram of the simple
"split-load" phase-splitter stage. The gain
of this circuit is less than unity.
60
 THE POWER AMPLIFIER
Fro. 3.11. In order to --.------------HT+
obtain sufficient drive
signal for the push-pull
power valves, additional
amplifying valves may AMPLIFIED
be necessary, as shown. PUSH-PULL
OUTPUT

SIGNAL

and the other half in IN ADDITIONAL

AMPLIFYING
the cathode circuit VALVES
(R2). The input signal
from the voltage
amplifier is applied
between the grid and
chassis, and due to
the high-value load
(R2) in the cathode the resulting large amount of negative feedback gives
rise to considerable loss of gain over an ordinary voltage amplifier, in which
the net load is in the anode circuit.
In fact, the actual gain between the grid and either side of the push-pull
output is around 0·9, giving a total gain between the two output terminals of
I ·8. The voltage output at either terminal is half that of a comparable voltage
amplifier, which means that if a fairly high drive is required by the push-pull
output valves additional amplifying valves have to be introduced between
either output terminal and the grids of the output valves (Fig. 3.ll). This
arrangement will be found in some modern equipment.
The phase-splitting valve is biased by the usual cathode resistor (R3
in Fig. 3.10) and the grid-return resistor, R4, being connected to the junction
of the cathode load and the biasing resistor. The heater of the valve is some-
times connected to a positive potential so that it approaches that of the
cathode. The reason for this is to
reduce hum which may otherwise be
reflected from the cathode circuit into
--+---•PUSH-PULL
..,__ _ OUTPUT
the grid circuit.
Fig. 3.12 shows the circuit of a
cathode-coupled inverter, used in the
Pamphonic Model 2,001. As with
a number of modern phase-splitting

C2

Fro. 3.12. Circuit diagram of a cathode-

coupled phase-splitter stage.
61
 THE PRACTICAL HI-Fl HANDBOOK
_ _ _,.__HT+
FIG. 3.13. Circuit diagram of a floating
paraphase inverter stage.
PUSH
PULL circuits, it calls for the use of two
OUTPUT
triode valves, a double-triode unit
being admirably suited to the func-
SIGNAL tion. Section (a) of the valve serves
IN
as an ordinary earthed-cathode
amplifier, in which the input signal
is applied through CI and delivered
across RI. Section (b) receives
its signal by way of C2 from that
present across the common cathode resistor R2, and delivers its output
across R3. Section (b), in effect, is an earthed-grid amplifier, whose output
signal is exactly 180 deg. out of phase with the signal at the output of
section (a).
The so-called paraphase circuit is also used in various forms for phase-
splitting. The version given in Fig. 3.13 is sometimes called the "floating
paraphase", in which two triodes (a double-triode valve) are used. The
signal applied to triode (a) grid is developed in amplified form across RI
and applied through Cl to one of the push-pull valves. Resistors R3 and R4
in series also form a load on triode (a), and the signal voltage at their junction
is applied to the grid of triode (b). Triode (b) thus gives rise to an amplified
anti-phase signal across its load R2, which is conveyed by way of C2 to the
grid of the other push-pull output valve. Resistors RS and R4 in series also
form a load on triode (b). The circuit is maintained in a good state of balance
since opposing voltages are developed across R4 due to the two triode
sections, and the signal voltage applied to the grid of triode (b) is maintained
at just the right level for optimum balance. In effect, the circuit has a self-
balancing action.
There are many variants of the phase-splitting arrangements mentioned,
but the information given should be sufficient to assist the technician and
enthusiast with any servicing and adjustments which may be required in this
section of the amplifier.
To summarize, the requirements of the phase-splitting stage, sometimes
known as inverter stage, are to provide a balanced drive signal for the push-
pull output valves from a common signal source, such as a voltage amplifier;
to provide a signal of sufficient voltage, without distortion, to ensure maxi-
mum output of the power amplifier; and to provide signals at each output
valve grid differing in phase by exactly l 80 deg. throughout the audio
spectrum.
The phase-splitting valve itself is usually fed from a voltage-amplifier
62
 THE POWER AMPLIFIER
Fm. 3.14. Fre- dD
quency response +5
characteristic of
the Pye Provost -5
orc-----'---------------.c-------t
amplifier. -10

I els I0c/s I00c/s I000c/s IOkc/s 100 kc/s I Meis

stage, which invariably forms the first stage of the power amplifier unit, into
which are fed the signals from the pre-amplifier. The response characteristics
of the power amplifier, therefore, should be sensibly flat over the entire audio
range and, with regard to feedback stability and the correct handling of
transient signals, should remain reasonably flat without peaks for two or
three octaves beyond the highest usable frequency. The excellent response
characteristic of the Pye HF25 (Provost) power amplifier shown in Fig. 3.14
illustrates this point.

ULTRA-LINEAR STAGE
The majority of hi-fi amplifiers incorporate an ultra-linear (sometimes
called "distributed load") output stage. The arrangement uses tetrode output
valves, but instead of their screens being connected direct to the h.t. positive
line, they are each connected to a tap on the primary of the output trans-
former. The basic circuit is given in Fig. 3.15.
This form of connexion gives the output stage a characteristic which in
most respects is between that of a tetrode and a triode. The desirable low
distortion and good linearity of the triode is maintained, as also is the high
output and sensitivity of the tetrode. Comparatively less negative feedback
is required for a given result, resulting in a greater margin of stability. The
distributed load effect also results in a reduction in total d.c. variations in the
output stage at high output levels.
Also, since the low capacitances of the HTt
tetrode are maintained, there is less
reactive shunting at high frequencies
and a reduction of phase shift at the
high-frequency end of the passband.
The tapping point for the screens INPUT
is rather important from the distortion
PUSH-PULL
TO·:~
SPEAKER

aspect. There is an optimum point for

different valves, and it usually ranges
between 20 and 43 per cent.

FIG. 3.15. Basic ultra-linear circuit.

63
 THE PRACTICAL HI-Fl HANDBOOK
--------HT.• F1G. 3.16. Partial cathode loading, as
used in the Quad II amplifier.

CATHODE LOADING
Total cathode loading of an out-
PUSH-PULL put stage (i.e., the load connected in
INPUT
the cathode instead of the anode
circuit) results in I00 per cent negative
feedback and calls for driver stages
capable of supplying some 150 to 200
volts of signal. Whilst this method has
received considerable attention, it is
rarely used in hi-fi equipment. Partial
cathode loading is used successfully,
however (for example, in the Acousti-
cal Quad II amplifier); see the circuit diagram in Fig. 3.16.
The cathode winding results in a portion of the output signal being
returned to the grids in the form of negative feedback. A conventional
negative-feedback loop is also incorporated and the combined effect gives rise
to a low output impedance coupled with very low distortion at high power.
Cathode loading combined with ultra-linear operation has been
experimented with in the United States (see Fig. 3.17), but the arrangement
does not lend itself to class A operation, though quite low distortion figures,
it is claimed, are possible by operating at class AB or class B.

SINGLE-ENDED PUSH-PULL
By the connexion of two output valves in series and the application of a
push-pull drive signal, the output impedance is somewhat reduced and
rendered less critical than that of a
HT+
more conventional stage. This idea
lends itself to transformerless opera-
tion, meaning that a loudspeaker (of
higher impedance than normal) can be
connected direct to the output stage
without introducing an output trans- PUSH-PULL
former. INPUT

FIG. 3.17. Combined partial cathode

loading and ultra-linear circuit. This is not
very suitable for Class A operation, but is
ofgreater use for Class AB and B operation.
64
 THE POWER AMPLIFIER
FIG. 3.18 The Philips version of the single-
ended push-pull output stage.

R2
When it is realized that most of the
non-linearity and phase shift (the
latter limiting the amount of negative
feedback in the power amplifier) are soon
aggravated by the transformer, any
device leading to its elimination is
well worth consideration. We have
already seen that a current of the order
of one ampere is required in a 15-ohm
loudspeaker to give 15 watts. While
such a large current can be obtained easily by the use of an output trans-
former, very large valves would be required in order to obtain this current
without a transformer, though matching could be secured by loading into
the cathode circuits.
Experiments have been directed along these lines, but instead of using
large valves a number of smaller valves have been tried. In one experiment
16 6AS7G valves were required to obtain 12 watts of power in a 16-ohm
load. This is hardly economical, and one might well spend a lot of money
on a special transformer.
Philips have solved the problem, however, by the use of two series-
connected output valves driven in push-pull and a speaker having an
impedance of 800 ohms, against the conventional 15 ohms. The circuit is
given in Fig. 3.18. When valves are connected in this way in an output stage,
the circuit is often referred to as a single-ended push-pull stage.
Broadly speaking, the valves are biased so that they would pass equal
current in the event of zero drive signal. When a signal is applied to the control
grid of VI, however, this balance is disturbed at the frequency of the signal.
For example, a positive swing at the VI control grid results in an increase in
anode current and a reflected negative-going signal at the control grid of V2
with respect to cathode. In this way a push-pull signal is created and the
valves are push-pull driven. The out-of-balance current at audio frequency
flows through the loudspeaker coupling (d.c. isolating) capacitor CI and in
and out of the speaker's speech coil, as would be the case if it were connected
across the secondary of a conventional output transformer.
In the Philips "Hi-Z" power amplifier, four valves are used in a parallel
single-ended push-pull arrangement, and the speaker is fairly heavily damped
by the low cathode impedance of V2.
65
 ALL TEST VOLTAGES MEASURED WITH AYO MODEL 8 HT VOLTAGES ON
IOOOV RANGE, CATHODE VOLTAGE VI ON 2~V AANGE V2 ON 2!1011 LI
RANGE, V3 &, V4 ON IOOV RANGE '5()V o&OOV T2

100-ISOV

O~v -I
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R6I Cl
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=
0
0
SI ~

FIG. 3.19. Circuit diagram of the Pamphonic 2,001 power amplifier. The negative-feedback loop is by way of R23
and Cl 2, and the positive-feedback loop by way of PI/R22 andR21. Pl serves as the damping-factor control.
 THE POWER AMPLIFIER
There are various modifications of this circuit, some of which are driven
from a phase-splitter stage which is designed along with the output stage to
secure optimum balance of drive signal. In general, the systems cancel
second-harmonic distortion by the distortions of each valve appearing in
antiphase in the common load, though it is sometimes desirable to unbalance
the distortions of the two valves, so that it is less in VI than in V2. This may
be secured by shifting the working point of V2 away from that of minimum
distortion or by applying negative feedback to VI by excluding a cathode
bypass capacitor (i.e., removing C2).

APPLICATION OF FEEDBACK
The best way of getting to know the feedback loops is to study them in
an actual circuit. In Fig. 3.19 is shown the complete circuit of the Pam-
phonic 2,001 power amplifier. Here stage VI serves essentially as the first
voltage amplifier. It raises the level of the signals from the pre-amplifier
sufficiently to operate the cathode-coupled phase-splitter V2. This in turn
drives the ultra-linear push-pull output valves V3 and V4.
Negative feedback is applied over the whole of the power amplifier
by feeding back to the cathode circuit of VI a suitable fraction of the signal
voltage developed across the secondary of the output transformer Tl. The
phasing of the feedback is such that it is negative, while the degree of feed-
back is governed by resistor R23. Cl2 is usually known as a phase correction
capacitor, whose purpose is to render the feedback loop very slightly
frequency-selective (the capacitor is usually of fairly low value) and thus
enhance the feedback stability margin. R23 and Cl2 are required to be
altered in value so as to maintain optimum feedback on changing the loud-
speaker impedance tapping.
In transferring a signal in opposite phase from the reverse side of the
secondary winding of the output transformer also to the cathode circuit of
VI, a positive-feedback loop is provided. The positive-feedback signal is
developed across R22 and Pl and fed back through R2l. Pl is, in fact, a
pre-set potentiometer which allows adjustment of the damping factor of the
output stage. It will be recalled from the description of negative feedback in
Chapter 2 that the output impedance as "seen" by the loudspeaker reduces
as the negative voltage feedback is increased, and further reduces as the
positive current feedback is increased. It is on this principle that the damping
factor control operates.
To recapitulate, the damping factor is equal to the nominal output
impedance of the amplifier divided by the impedance as "seen" by the
loudspeaker (source impedance); a low source impedance results in a high
damping factor and an infinite damping factor results when the source
impedance falls to zero ohms.
67
 THE PRACTICAL HI-FI HANDBOOK
EFFECT OF DAMPING FACTOR ON LOUDSPEAKER
A low source impedance or high damping factor affects the cone of a
loudspeaker in much the same way as the shock absorbers affect the stability
of a car. A car with worn or faulty shock absorbers oscillates vigorously up
and down on riding rough ground or when a sudden stop is demanded. In
the same way a high source impedance causes the cone of the loudspeaker to
oscillate beyond the normal pattern of the audio signal when this is of the
nature of sharp transients.
The effect is illustrated in Fig. 3.20. At (a) is a square wave which may
be applied to the input of an amplifier, causing the loudspeaker cone to move
rapidly in one direction as represented by A-B. Ideally, at point B the cone
should stop dead and remain still until point C, when it should move rapidly
in the opposite direction represented by C-D. This ideal will be approached
when the amplifier source impedance as "seen" by the loudspeaker is very
low (when the damping factor is high).
If the source impedance is high, however, there will be no electronic
shock absorption, and the loudspeaker cone will follow the pattern as shown
in Fig. 3.20b. Here, from A to B the cone will move rapidly in one direction,
but instead of coming to a halt at B it will continue oscillating between B
and C. It will change direction at C, but again oscillate about point D. This
is known as a "damped oscillation", which may not only develop in the
loudspeaker owing to a low damping factor, but may also appear as current
oscillations in resonant elements of the amplifier. In this latter connexion,
the effect is often referred to as "ringing".
Apart from being damped electronically, the loudspeaker is also
acoustically damped by the loading in its enclosure. It is said, therefore, that
every speaker has a certain critical damping factor from the electronic
aspect, and for this reason a large number of amplifiers are provided with a
damping control. The author feels that a loudspeaker can be over-damped
as well as under-damped, the effect in this case being like that of a light door
coupled to one of those large automatic door-closing devices-the door can
be neither opened nor closed sharply. The effect on the loudspeaker is that
the cone is unable to follow very sharp transient waveforms.
This electronic damping effect can be demonstrated with a moving-coil
milliammeter. With the terminals of the meter open-circuited, vigorous
twisting of the meter will cause the pointer to oscillate in a very disturbed

n n (a)
c

0
A (bl 0
Fla. 3.20. A square-wave signal, as shown at
(a), would cause cone oscillation (b) as the

68
result of insufficient damping.
 THE POWER AMPLIFIER
Fm. 3.21. Typical har- 5
monic distortion curve j
4
for an amplifier rated at a
nominal IO watts. I
I
:__...,"'
J
4 8 10 12 14 16
WATTS

manner on its pivot; by short-circuiting the terminals, however, the pointer

oscillation will be found to be considerably curtailed. The moving coil is
damped by the low resistance shunt (short-circuit). This is one way of
protecting such instruments during transit.
An infinite damping factor is when the source impedance of the
amplifier "looks" to the loudspeaker as a dead short. In amplifiers having a
damping factor control the setting corresponding to this condition can be
established by feeding into the amplifier a 1,000 c/s signal and adjusting the
volume control until about 1-2 volts is measured across the loudspeaker
terminals (an a.c. voltmeter is necessary). The damping factor control should
then be adjusted whilst alternately disconnecting and reconnecting the
loudspeaker. The position corresponding to an infinite damping factor is
revealed by the output voltage remaining the same whether the speaker
is connected or disconnected.

POWER OUTPUT

A hi-fl amplifier should be capable of supplying a speaker with at least

10 watts of power with a total harmonic distortion not exceeding O· l per
cent. Obviously, the amount of power required will depend upon not only
the size of the room, but also the efficiency of the loudspeaker system. A
speaker of 10 per cent efficiency (a good figure) will produce only I watt of
radiated acoustical power when connected to a fully loaded JO-watt amplifier,
but this level of sound cannot be endured with any comfort in an average-
sized living room or lounge.
Most amplifiers are very conservatively rated, and quite a number of
British instruments nominally rated at IO watts can be pushed up to 14 watts
without the distortion rising above I per cent. This is illustrated by the typical
distortion curve in Fig. 3.21; at 10 watts the distortion is in the region of
O· I per cent, and is still reasonably low up to 14 watts, but rises very seriously
at greater outputs.
It is highly desirable for a high-power amplifier to have a margin of
reserve power in hand before distortion becomes serious. For example, an
69
 THE PRACTICAL HI-Fl HANDBOOK
orchestral peak of sound may increase the input signal voltage beyond that
specified for maximum output. In the case of a JO-watt amplifier working at
nearly full volume, such peaks will push the amplifier into heavy distortion.
In the case of a 30-watt amplifier adjusted for the same nominal sound inten-
sity, however, the peaks will still be within the low-distortion power-handling
range of the unit, and will be reproduced with equal fidelity as the low-
intensity sounds.

FREQUENCY AND POWER RESPONSE

The frequency response is taken as a measure of the deviation of output
signal voltage as the input signal voltage is altered in frequency over and
beyond the audio spectrum (see Fig. 3.14). The output voltage deviation is
expressed in decibels over the frequency range employed. Unfortunately,
the frequency response is usually given for a level considerably below the
maximum power output of the amplifier. Thus, the response characteristic
may be virtually level from, say, IO c/s to 20,000 c/s at an output of l watt,
but deviate considerably from this at full output.
An amplifier rated at IO watts may give this figure at O· I per cent
distortion at 1,000 c/s (the frequency at which measurement is usually made),
but for the same distortion at 100 c/s the maximum power may be only
5 watts. The distortion may well rise to 2 or 3 per cent at 100 c/s unless the
input signal at and below 100 c/s is reduced accordingly. This, of course, is a
hypothetical consideration and to such a degree may not be common to all
amplifiers; nevertheless, it indicates a need for a low-cut filter somewhere in
the pre-output stages. To a smaller degree the same reasoning applies to the
higher frequency end of the audio spectrum, but here the trouble is not as
serious, as the higher-frequency signal components usually comprise low-
level harmonics.

POWER SUPPLIES
The mains transformer, rectifier, smoothing choke and filter components
are mounted on the power-amplifier chassis, or on the power-amplifier side
of a combination chassis. In the case of two-unit models, power for the pre-
amplifier is fed from the power amplifier by way of a multi-cored cable and
plugs and sockets to suit the individual design.
Referring to Fig. 3.19, the mains transformer T2 has an adjustable
primary winding to suit almost any mains supply, and three secondary
windings. The h.t. winding is centre-tapped and supplies 400 volts (with
respect to chassis) to the two anodes of the h.t. rectifier valve V5. This is a
straightforward full-wave rectifier circuit, which needs little comment. H.t.
smoothing is provided by the smoothing choke LI and the electrolytic
capacitors Cl3 and Cl4, and a full 450 volts d.c. is present on the main h.t.
70
 THE POWER AMPLIFIER
Fm. 3.22. The Pam-
phonic amplifier,
Model 1,004.

line, which is also fed to the pre-amplifier power socket on point 4. The
rectifier heater has its own winding supplying 5 volts a.c., and a 6·3-volt
winding serves to supply the heaters of all the valves, the l.t. feed for the pre-
amplifier valves being taken to points 7 and 8 on the connecting socket.
The potentiometer P2 serves as a humdinger control. Apart from
producing an exact balance in the heater chain, it also introduces a small
positive bias to the heaters from the cathode of V4. Correct adjustment of
this control can reduce the residual mains hum by as much as 20-30 db.
Adjustment is best made by connecting a sensitive a.c. voltmeter across the
loudspeaker terminals, short-circuiting all the input terminals of the amplifier,
and rotating the potentiometer for minimum reading on the voltmeter. The
adjustment can also be made by ear for minimum hum level.
Illustrations of three typical equipments are given in Figs. 3.22, 3.23
and 3.24. The Pamphonic Model 1,004 (Fig. 3.22) is a JO-watt combination
unit, having a maximum distortion of0·5 per cent at 1,000 c/s at full output,
and a frequency response which is substantially flat from 20 c/s to 50 c/s.
It has 20 db of negative feedback, and inputs for microphone, tape, radio and
gram. Volume, contour (designed in accordance with the Fletcher-Munson

FIG. 3.23. The Pam-

phonic amplifier,
Model 2,001.
71
 THE PRACTICAL HI-FI HANDBOOK
Fm. 3.24. The Pye
Provost power ampli-
fier, Model HF25, and
the Proctor remote-
control unit, Model
HF25A.

curves), bass and treble controls are featured, the latter employing the
Baxandall system. Provision is also made for a plug-in pick-up attenuator so
that widely differing types of pick-up can be used. Three types of equalizing
are catered for, selected on the main programme-selector switch.
The Pamphonic Model 2,001 is a larger two-unit amplifier (Fig. 3.23),
capable of giving a full 25 watts at very low distortion. This model incorpor-
ates all the features described in this and the previous chapter.
Fig. 3.24 shows the Pye "Provost" power amplifier and the associated
"Proctor" remote control unit. This also has a power output of 25 watts with
less than 3 per cent harmonic distortion at 1,000 c/s. At 15 watts the distor-
tion is less than 0· l per cent, but approaches 3 per cent at 30 watts output.
The amplifier has an excellent frequency characteristic (see Fig. 3.14) and is
substantially flat from 2 c/s to 160 kc/s. All the requirements of a hi-fi
amplifier are catered for, including the Baxandall-type tone control and four
recording characteristics, by the use of feedback networks (see Fig. 2.17).
The power units of most hi-fi amplifiers are over-rated so that h.t. and
l.t. can be fed to an auxiliary unit, such as a radio tuner. Provision is also
usually available for the connexion of a gram motor or tape recorder. A
signal output socket is often to be found on the pre-amplifier or control
unit. This is usually picked-up from the output of the voltage amplifier, after
the tone-control circuits and before the volume control. This feature enables
the programme signal to be applied to the input of a tape recorder or
additional amplifier.

72
 CHAPTER 4

Tracing and Clearing Faults

in Amplifiers

WITH a knowledge of the somewhat critical and specialized

circuits used in hi-fi equipment (and also of the peculiarities of temperament
of the enthusiastic hi-fi owner!) the service technician will soon find himself
as much at home with hi-fi equipment as with radio or television receivers.
Basically, the hi-fi chassis is far less complex than a modern television chassis,
but because of the delicately balanced circuits in the former and the subjec-
tive nature of hi-fi, servicing compromises are never worth adopting.
For example, if an anode-load resistor is found to be open-circuit, and
it is a close-tolerance component valued at 50,000 ohms, then replacement
should be made with a component of identical characteristics. Whilst a
preferred-value 47,000-ohm resistor would restore operation of the amplifier,
and the service technician may feel that the performance is then well up to
standard, the owner who is highly sensitive to every characteristic of his
amplifier will soon sense that something is not quite right.
He will probably possess the circuit diagram of the unit and may eventu-
ally find the incorrectly valued resistor. Immediately he will attribute to this
the shortcoming in performance. He will obtain and himself fit the correct
component, and whilst technically the results may not be improved, the
enthusiast will believe that they are and will be quite satisfied that the ampli-
fier is now up to its former standard. Word will soon get around the local
hi-fi world about the wicked ways of the unfortunate service technician, and
he may have difficulty in regaining the confidence of the local enthusiasts.

COMPLETE FAILURE

This is one of the easiest of faults to locate. The first check would be to
establish the connexion of power to the equipment. If the valves and pilot
bulb are not alight, the fault is almost certainly in the mains input circuit.
The plug and socket connexions at both ends of the mains lead should be
examined carefully, and it should be established that power is actually present
73
 THE PRACTICAL HI-FI HANDBOOK
on the mains socket. There may be a break in one of the conductors of the
mains supply lead. This often happens if the equipment is moved around
extensively, but the trouble is usually of an intermittent nature.
The next move would be to check the amplifier fuse or fuses for conti-
nuity. If these are in order the on/off switch, associated connecting cables,
connexions to the primary of the mains transformer and the voltage-selector
plug and socket should be carefully examined. Sometimes a poor or inter-
mittent connexion exists on the voltage-selector connector or a dry joint
develops on the mains circuit connexions. Testing along these lines will soon
reveal the cause of the trouble.
If it is found that the fuse is open-circuit, a check for short-circuits on
the h.t., l.t. and mains circuits should be made before a new fuse is fitted and
the amplifier switched on. Fuses are fitted to protect these circuits, and a fuse
rarely blows without provocation. Test for shorts can be made with a simple
ohmmeter. A check on the 1.t. circuits should first lead to removal of all
the valves and connexion of the ohmmeter across the heater line, bearing in
mind that the line is shunted by the heater winding on the mains transformer
and possibly a humdinger control. Removal o( these components may also
be called for if the meter used cannot indicate low ohms.
To check for a h.t. line short, the meter should be connected between the
chassis and the h.t. line, bearing in mind the charging and discharging kicks
promoted by the electrolytic capacitors. If a reading of some hundreds of
ohms, or less, is given, the probe of the instrument should be transferred to
the various h.t. feeds until the source of the low resistance or short-circuit
is brought to light. Typical faults in this respect are shorting smoothing
electrolytics, a short in the h.t. rectifier, a winding-to-core short in the
smoothing choke, and valve-holder shorts to chassis.
If the circuits appear completely free from excessive leakage resistance,
the fuse should be replaced (one of stipulated value is essential for optimum
protection) and the amplifier re-connected to the mains and switched on.
While it is warming up the valves, particularly the h.t. rectifier and output
valves, should be carefully observed for signs of an internal flashover. A
heavy flashover of this nature will immediately cause failure of the replace-
ment fuse. In this event, both the valve and the fuse should be replaced.
If the valves are alight the programme signal should be disconnected
or the volume control fully backed-off. At this point it should be made clear
that the audio-frequency voltages developed across the primary winding of
the output transformer rise to a high level if a signal is conveyed through the
amplifier at normal level with the loudspeaker load removed. It is thus
essential to establish continuity of the loudspeaker circuit with the signal
removed. Many a good and expensive output transformer has been damaged
by operating the amplifier without a correct load. The amplifier can be run
74
 TRACING AND CLEARING FAULTS IN AMPLIFIERS
at full power, of course, by using a suitable load resistor in place of the
loudspeaker; a wire-wound component rated at the full output power of the
unit should be used in this case.
It is a simple matter to check loudspeaker and connecting-lead continuity
and resistance by disconnecting the loudspeaker wires from the speaker
terminals on the amplifier and then connecting the leads to the terminals of a
battery-operated ohmmeter. A crackle will be heard in the speaker on
connexion and disconnexion of the wires.
Once it has been established that the loudspeaker is, in fact, acting as a
load and is in good condition, the volume control can be advanced to its
normal setting without fear of damaging the output transformer. At this
stage, the programme-selector switch can be turned over the various
positions, which, provided the appropriate programme signals are available,
will indicate whether or not the failure is common to all channels.
Assuming that all channels are dead, tests should be made to find out
whether the trouble lies in the pre-amplifier or power amplifier. This is a
simple matter with two-unit amplifiers, it being necessary to unplug the pre-
amplifier from the power amplifier and apply one of the programme signals
direct to the input socket on the power amplifier; the pick-up signal is usually
suitable for this test. Whether or not the power amplifier will be fully loaded
by this signal will depend upon the overall sensitivity of the amplifier and
the level of the pick-up signal-depending upon the type of pick-up used. At
this stage, however, we are not interested in the quality or quantity of the
sound, and provided we get a reasonable form of reproduction, it is fairly
safe to assume that the trouble lies in the pre-amplifier section.
With a single-unit combination amplifier, a similar test can be made
by applying the signal between the chassis and control grid of the valve
immediately prior to the phase-splitter stage.

POWER AMPLIFIER FAILURE

When making signal tests on certain power amplifiers with the pre-
amplifier or control unit disconnected, it may be necessary to short-circuit
the points on the control-unit socket corresponding to the mains on/off
switch, since this switch is usually situated on the control unit. Fig. 4.1 shows
the circuit diagram of the well-known RD Junior power amplifier, in which
points 7 and 8 on the control-unit socket are those associated with the mains
on/off switch; shorting these will complete the mains input circuit. Referring
to the same circuit, the signal would be applied across points 5 and 6 of the
same socket, with the earthy side of the signal source to point 6.
A probable cause of the lack of response is failure of the h.t. circuits,
and this can be established quite rapidly by testing the temperature of the
output valves, V2 and V3, with a finger. The output valves usually operate
75
 THE PRACTICAL HI-FI HANDBOOK
,,.....,,,,, CHI Rl4

J;cs
R4

CONTROL
UNIT
SOCKET

FIG. 4.1. Cirrnit dial(ram of the RD

Junior amplifier.

at a fairly high temperature, and if they are only just warm there is a strong
likelihood of a burnt-out h.t. rectifier valve, V4; failing that, the h.t. feed
resistor Rl4 should be subjected to a continuity test. It is surprising how much
can be done in the way of simple servicing and diagnosis without instruments
-merely by applying a little well-concentrated thought.
It is extremely unlikely that the fault would be caused by simultaneous
failure of both output valves; whilst failure of one of the output valves would
result in the offending valve losing temperature, the good valve would remain
too hot for comfortable touch, and the amplifier would reproduce after a
fashion. The same applies with regard to the output-transformer primary
windings-it is most unlikely that both sections would go open-circuit
at the same time.
There is one more possibility, however, and that is open-circuit of the
common cathode resistor Rl2. This trouble would cause both valves to lose
temperature, though it is possible that the bypass capacitor C6-being a low-
voltage electrolytic-would leak heavily and give some sort of cathode-
circuit continuity. In this case, the amplifier would reproduce, but the dis-
tortion would be high. Some amplifiers have separate cathode resistors and,
again, both would hardly fail simultaneously, though there is a remote
chance of this happening!
If both output valves are working at fairly high temperature, and there
is a very slight trace of normal residual mains hum from the loudspeaker,
76
 TRACING AND CLEARING FAULTS IN AMPLIFIERS
one can be fairly certain that the voltage-amplifier/phase-splitter stage, VI,
is defective. To check the phase-splitter section, a signal could be applied
to its grid-pin 7 in the circuit of Fig. 4.1 ; a pick-up signal may not be strong
enough at this high-level point (depending upon the output voltage of the
pick-up) and it may be necessary to bring into service an audio oscillator.
Failing this, however, one side of the heater line could be connected to the
grid through an O· l mF capacitor. This action will inject into the grid circuit
a 50-c/s mains signal (at about 3 volts) and, if the phase-splitter section is
operational, will give rise to a very loud mains hum from the loudspeaker.
The remaining stage is the first triode section of VI-the voltage amplifier.
Open-circuit of the anode-load resistor R3 or the coupling capacitor
C2 represent the most likely causes of the trouble. However, first a valve
change and then a check of anode voltage will soon bring to light the trouble.
The same simple tests are all that are necessary if, for instance, the
previous tests indicate trouble in the phase-splitter stage.

PRE-AMPLIFIER FAILURE
Fig. 4.2 shows the circuit diagram of the RD Junior control unit (pre-
amplifier). If it is found that the power amplifier passes a signal, but some
fault is preventing its passage through the pre-amplifier, it is best to make
tests with the two units connected together in the normal manner. However,
before delving too deeply into the pre-amplifier circuit from the servicing
aspect, it often saves considerable time to ensure that the signal-carrying
conductors of the multi-core pre-amplifier connecting-cable not only possess
continuity, but that they are also in good electrical connexion with the tags on
the plugs.
It is best to work back from the tone-control valve, V2b, to the first
voltage-amplifier, VI. The signal fed to the power amplifier, by way of point
5 on the octal cable plug, is developed across the volume control P7, being
picked-up from the anode of V2b. To check the goodness of stage V2b,
the volume control should be advanced about three-quarters of maximum,
and pin 7 of V2 touched with the blade of a screwdriver, with the blade
making contact with a finger. The other hand should be kept well away from
the amplifier, preferably in a pocket to avoid the risk of electric shock. If
all is well, a loud hum will emit from the loudspeaker, as the result of the small
mains signal being picked up by the body and injected into the grid. This test
can be repeated at the grid of V2a (pin 2) and the grid of V1 (pin 9).
If there is a loud hum at pin 7 of V2 and no hum (or a very weak hum)
at pin 2 of V2, the trouble lies either in V2a, in R22, R23 or in the coupling
capacitor C23. The valve is the most likely cause, and should at least be
checked by substitution. If there is a loud hum at the grid of V2a, but no hum
at the grid of VI, VI itself, R8, R7 and C4 should be checked in that order.
77
 R9 R32

Cl
7 -I
::c
PU 2 rr,
-,:,
PUI ,ii
>
(")
TAPE
RE LAY ~

MIC
P2 ->
-I
(")

r-'
oil RADIO
I I -.,.,
::c
'
-::c
P3 -;J
R30
I
>
z
FII.TER SHOWN AT SI AND S2 GANGED 56
' I I fc 0
4•5 kc/5 POSITION SI FITTED WITH MUTING CONTACTS a,
C5 SHORT· CIRCUIT S3, s• AND S5 GANGED 0
REMOVED IF CCIR
87G SOCKET
TAPE CHARACTEIUSTIC
S6 AND P7 GANGED 0
IS REQUIRED ~

Fm. 4.2. Circuit diagram of the RD Junior control unit, Mk. II.
 TRACING AND CLEARING FAULTS IN AMPLIFIERS
If R5 (the screen-feed resistor) appears to be overheating, suspect a short in
C2. A few simple voltage and resistance checks will soon bring to light the
component responsible.
H.t. power for the pre-amplifier is applied from point 3 on the octal
cable plug. Make sure that h.t. is present here, and that it is getting past
R32 and R9 (filter resistors). Overheating of R32 would indicate a short in
C30, while the same trouble in Cl would cause R9 to overheat.
There is usually no need to set up elaborate instruments to diagnose for
total failure if the tests outlined above are followed logically. Once the
defective section has been revealed, the problem is virtually solved, for it is
then only a matter of testing a few small components and the voltage at a
couple of key points.
Instead of relying on the hum method of testing, the signal from an
audio oscillator or generator can be applied to the various stages in tum
until the point is reached where the signal is blocked; but generally speaking,
the hum method is the quickest, and just as reliable. Alternatively, a pair of
headphones, or an ear-piece, can be used to trace the signal through an
amplifier up to the stage or component which is preventing it getting any
farther. This method of testing calls for a normal input signal from one of
the programme sources and average settings of the various controls. The
phones can be used to trace the signal from the programme source right up
to the point of the trouble. For more complex faults, test instruments are
usually required.

DISTORTION
Distortion in one form or other probably accounts for the majority
of troubles in hi-fi amplifiers. The symptom ranges from a very low-level
distortion, which invariably demands some curious instinct to detect, to a
very high-level distortion, whose presence is obvious to any listener.
The reader should understand that there is no such thing as a com-
pletely distortionless reproducing channel. Somehow, somewhere, in the
electro-acoustic link between the live programme in the studio or concert
hall and the ear of the listener at the loudspeaker end, the original sound will
be altered slightly in character. It may be "coloured" by the position of the
microphones in the studio and by the position of the loudspeaker and room
acoustics at the listening end of the link. It will most definitely be modified
during its passage in electrical form through the various electronic circuits
If a number of microphones are used close to the instruments of an
orchestra, the pick-up of direct sound will be far in excess of the pick-up of
reflected sound and the reproduced sound will lack "atmosphere"; it will not
sound the same from the loudspeaker as it would in the middle of the concert
hall. Little can be done by the enthusiast to correct this trouble, however.
79
 THE PRACTICAL HI-Fl HANDBOOK
OSCILLOSCOl'E
LOAD kESISTOR

AUDIO GENERATOR AMPLIFIER

FIG. 4.3. Arrangement of instruments for

checking frequency-response and over-
loading.

At the reproducing end, the room acoustics will obviously differ from those
at the transmitting end, and even if a desirable degree of "atmosphere" is
introduced, the final result will be further coloured by the listening-room
acoustics. If "atmosphere" is purposely excluded by the sound engineer, it
is most unlikely that the acoustics of the listening room will resemble those
expected of a concert hall. A compromise is necessary along these lines, and
this is the main reason why hi-fi amplifiers use elaborate tone-control
circuits.

FREQUENCY DISTORTION
Frequency distortion is present when the output signal deviates widely
in amplitude as a constant-amplitude input signal is altered in frequency
over the entire audio spectrum. Almost all hi-fi amplifiers are substantially
flat in response over, and beyond, the audio spectrum, as we have already
discovered, and they are rarely troubled with this form of distortion. How-
ever, at high power outputs, the response may not be quite as flat as suggested
by the appropriate response curves.
In Fig. 4.3 is shown an arrangement of instruments which can be used
for frequency-response checking and plotting. An audio oscillator or generator
is coupled to the input of the amplifier under test, ensuring that it is correctly
matched to the input channel selected, a load resistor of suitable value and
rating is employed in place of the loudspeaker and the voltage (a.c.) across it
is measured by the output meter. The output signal is also monitored on an
oscilloscope.
For high-level testing, the amplifier volume control is turned to maxi-
mum, the tone controls to the "flat" position, the filters switched out, the
generator tuned to 1,000 c/s and the generator gain control adjusted for
maximum power of the amplifier as given on the output meter. The waveform
is synchronized on the oscilloscope to ensure that it is not highly distorted
owing to overloading of the amplifier by too great an input signal.
With the various controls set, the generator should be tuned to about
80
 TRACING AND CLEARING FAULTS IN AMPLIFIERS
20--30 c/s, the oscilloscope re-synchronized to that frequency and the wave-
form checked to ensure that it is still free from distortion. Normally, a pure
sine wave will be displayed, depending upon the signal given by the generator,
but if the peaks of the wave appear to be flattened, the input signal should
be decreased until the distortion disappears, a note being made first of the
original setting of the gain control. The output level should be noted at each
point as the test is made over the audio spectrum, up to the limit of the
generator, and plotted against frequency to give the response curve.
If it was necessary to decrease the input signal at the lower-frequency
end, the gain should be advanced progressively up to 1,000 c/s, ensuring
each time a test is performed that the signal is not overloading the amplifier.
For low-level testing, the same procedure is adopted, but this time the
input signal is adjusted to give about I watt power output. In this case, there
will be little danger of overloading the amplifier, and an oscilloscope is not
essential.
If a proper output meter calibrated in watts of power is used, it will
probably incorporate its own load resistor, but it must be ascertained that
this represents the correct match to the amplifier; the mete1 should also have
a level response itself over the audio spectrum. It is similarly pointless making
such tests with an audio generator whose output voltage varies greatly over
the band; if the instrument does not have a voltage-output indicator of its
own, then its response should be plotted on a curve, which can later be
used to correct the amplifier response curve.
If an output meter is not available. a high-resistance level-response a.c.
voltmeter can be used equally well. The power output can be computed by
using the expression: W=£2/R,
where E is the voltage and R is the

~
112'., """'~-,-
.
/.,;·· ,.-.;,:
,'
~
.

-:,.,
.. :·
··.. - ~.:_;::....-
... .·
.-.~
' '
resistance of the load in ohms.
The oscilloscope (which is in-
valuable for many tests on hi-fl
equipment) should possess a good

•
low-frequency response in relation
. Ll~ :•~;_ .~ to its Y amplifier (preferably from
-~
-. ... .
~
d.c. to I Mc/s or above), have a
linear timebase and ease of syn-
•· •·-··-· . . chronizing the test signal. An instru-
I·. -.· ~-· -~-·_· '·..:·.
~ ~ ;1' ment highly suitable for this work is
the Serviscope, by Telequipment.
, ... }. ... ·•.::'.- t::.. / Among many other refinements, this
.. . :

. -,..;. ··•· ..--~'\

~

~......... ''- FIG. 4.4. The Serviscope, by Telequip-

ment.
81
 THE PRACTICAL HI-Fl HANDBOOK
FIG. 4.5. Input-voltage/output-voltage characteristic:
8
t A-Bfor a distortionless amplifier: A-Cfor a practical
...
I!)
non-linear amplifier .
~
_,
0 has a Y amplifier which is substantially flat from
...> d.c. to 6 Mc/s and a trigger-type timebase which
...i:, obviates complex synchronizing re-adjustments
0
on altering frequency (see Fig. 4.4).
A INPUT VOLTAGE . -
Incidentally, an audio oscillator having
provision for a square-wave output is most
desirable, since many tests can be made by injecting a square-wave signal
into the input and observing its form after passing through an amplifier.
An amplifier suffering from frequency distortion is characterized by its
somehwat "mellow" tone, which is caused by severe attentuation of the higher
frequencies in relation to the low frequencies.

NON-LINEAR DISTORTION AND HARMONICS

Owing to the curvature of valve characteristics, deficiencies of the output
and coupling transformers, etc., the input voltage/output voltage characteris-
tic of any amplifier deviates from a straight line over the major portion of
its range. The effect is shown graphically in Fig. 4.5; here line A-B would
represent the characteristic of a distortionless amplifier, but in practice the
characteristic takes the form of the broken line A-C. From this it will be
seen that the non-linearity is aggravated as the input voltage is increased.
There is no such thing as a distortionless amplifier!
The generation of harmonics of the fundamental frequency of the input
signal is one of the by-products of this non-linearity. For example, if the
input signal is a pure sine wave of frequency 250 c/s, the output signal will
consist of the fundamental 250c/s signal, a second harmonic at 500 c/s, a third
harmonic at 750 c/s, a fourth harmonic at 1,000 c/s, and so on. The magnitude
of the spurious harmonic signals will depend upon the extent of the non-line-
arity,and they are usually expressed in the form of a percentage of the magni-
tude of the fundamental signal. Hence, if the power of the fundamental signal
is 20 watts, and there is a second harmonic power of I watt, it could be said
that the amplifier has a second harmonic distortion of 5 per cent. Usually,
however, it is the total harmonic distortion of an amplifier which is given in
the form of a percentage. With push-pull amplifiers, as we have already seen,
the second and even harmonics are largely precluded by cancellation in the
balanced load, and the third ar.d possibly higher-order odd harmonics are
the troublesome ones.
Harmonic distortion when present in a large degree is chdracterized
by the harsh, "rough" nature of the reproduction. At lower levels, it is
82
 TRACING AND CLEARING FAULTS IN AMPLIFIERS

la) ID) (C)

FIG. 4.6. Various modes of second-harmonic distortion.

difficult to define objectively, but its presence has a fatiguing effect on the
listener; hi-fi enthusiasts are able to sense that something is not quite as it
should be, and are glad to get out of audible range! Some harmonics are
distinctly unpleasing, to say the least, particularly those which are dissonant
with the fundamental frequency, such as the seventh, ninth, eleventh, etc.
of a fundamental of 250 c/s.
Conversely, the emphasis or suppression of certain harmonics of certain
sounds tends to enhance the original sound, and in some cases makes a
displeasing sound more pleasing. This effect may be created by the use of the
various tone controls and filter controls on the amplifier.
The deformation of the waveform produced by the harmonic compo-
nents depends upon the phase of the harmonic relative to the fundamental.
In Fig. 4.6a is shown, in broken line, a fundamental and second harmonic,
which combine to form the distorted wave in full line. The combined wave
is obtained by adding or subtracting the instantaneous values of the two
waves. In (b) the harmonic component is displaced from the fundamental by
45 deg., resulting in a combined wave of somewhat different character, while
in (c) the harmonic is displaced by 135 deg., which has the effect of inverting
the distorted combined wave.
A third-harmonic component has the effect of distorting the waveform
like that shown in Fig. 4.7. A characteristic of waves distorted by odd
harmonics is that the positive and negative halves of the combined wave are
similar, while with even harmonics the positive and negative half-waves are
mirror images. In Fig. 4.8 is shown severe harmonic distortion created by iron
saturation as the result of overloading of an output transformer.

(Left) FIG. 4.7. Third-har-

monic distortion. (Right)
FIG. 4.8. Severe harmonic
distortion caused by iron
saturation in an output trans-
former due to an overload.
83
 THE PRACTICAL HI-FI HANDBOOK
INTERMODULATION DISTORTION
Another effect of non-linearity is intermodulation distortion. The effect
occurs when more than one input frequency is applied to the amplifier,
giving rise either to the production of sum or difference frequencies, or to the
amplitude modulation of one frequency by the other.
Although in practice there are a host of input frequencies applied
simultaneously, an illustration of the first effect is afforded by considering
the application of only two frequencies, one at, say, 40 c/s and the other at
1,000 c/s. A whole string of sum and difference frequencies will be formed;
for instance, the 40 c/s note will add to and subtract from the 1,000 c/s note,
thus producing spurious signals at 1,040 els and 960 c/s. If the 40 c/s signal
is of greater amplitude than the 1,000 c/s signal (tests are usually made in a
ratio of magnitude of 4 to I), harmonics of the 40 c/s signal will also add to
and subtract from the 1,000 c/s signal, thereby giving spurious signals at
1,080 c/s, 920 c/s, I, 120 c/s, 880 c/s, and so on. Harmonics of the 1,000 c/s
signal may also come into play if the non-linearity is severe, and make matters
even worse! Intermodulation distortion of this type is very unpleasing to the
listener, being far more disconcerting than simple harmonic distortion because
the spurious sum-and-difference tones are not harmoniously related to the
fundamental frequencies. Such distortion is characterized by a "buzz"
or "rough harshness" in the reproduction, and is apparent to almost any
listener.
Amplitude modulation of one frequency by another has been illustrated
admirably by G. A. Briggs in his book, "Sound Reproduction". Instead of
being connected to a source of d.c., the field coil of an early-type loudspeaker
was inadvertently connected to a source of "raw" a.c. and, to quote the
writer: "when a record was played through this equipment, music came out
of the loudspeakers but it was almost unrecognizable, sounding as though it
had been chopped up in a high-speed slicing machine. One effect of inter-
modulation is similar to this but not quite so bad."
Another example of intermodulation of this kind is the reproduction of
a choir with an organ accompaniment. The author has had occasion to
investigate this effect; a tape-recording made during a choir practice was said
to have a curious "warbling" characteristic. This was because the choir was
amplitude-modulated by the organ!
Harmonic distortion is checked on a wave analyser which is connected
to the output of an amplifier. A sine-wave input is applied to the amplifier,
and the wave analyser removes the fundamental frequency, and passes only
the harmonic components, which are measured as a percentage of the
fundamental.
An intermodulation analyser is required for measuring intermodulation
distortion. The analyser usually supplies the two input signals over a selected
84
 TRACING AND CLEARING FAULTS IN AMPLIFIERS
range of test frequencies, and has an attenuator which alters the ratio of the
two signal voltages over the range of I : I to IO: I. The composite signal is
applied to the input of the amplifier under test, and the output of the ampli-
fier is fed through a high-pass filter which eliminates the low-frequency test
signal. The remaining signal, consisting of the high-frequency test signal plus
the intermodulation, is demodulated. The high-frequency test signal is also
eliminated in another filter, and the spurious intermodulation signals are fed
to a measuring instrument which usually reads in terms of percentage
intermodulation.

PHASE AND TRANSIENT DISTORTION

Phase distortion causes the output waveform to differ from the input
waveform, due to alteration of the phase angle between the fundamental
frequency and an associated harmonic, and of the phase angle between any
two component frequencies of a complex wave. Although phase distortion
has an effect on the reproduction of transients, it would, generally speaking,
appear to be the least troublesome distortion encountered in audio work,
though its presence is clearly visible in television receivers. With a hi-fl
amplifier of wide frequency response, phase distortion is usually negligible,
but it rises somewhat by the inclusion of filters which serve to limit the
frequency range.
Distortion of the transients which, with certain kinds of music, occur at
very high level, tends to impair the "attack" performance of the equipment.
Transients are representative of sounds of short duration, such as those
produced by certain string instruments-the piano, for example-and by
percussive instruments. The general effect is that such reproduced sounds
tend to "hang-on" after the energizing pulse or waveform has decayed, and
where the distortion is severe, the frequency emitted during the period of
decay may differ from that of the actual energizing waveform. Reproduction
becomes very "slurred" on peaks.
Apart from a good transient response, depending to a large degree on
both the electrical and acoustical damping of the loudspeaker system, the
amplifier should also possess (1) a wide frequency response, extending
beyond the limit of audibility, (2) no phase distortion, (3) a high output
damping factor and (4) all resonant circuits, such as tone-control networks,
filters and transformers should be sufficiently damped to avoid "ringing".
A circuit which is subject to damped or supersonic oscillations will be
triggered into transient distortion by transient pulses, and the spurious
oscillation-at frequencies depending upon tuned frequencies of the offending
circuits-will become superimposed on the signal waveform.
Testing for transient distortion is performed by injecting a square-wave
signal into the input of an amplifier and observing its character on the
85
 THE PRACTICAL HI-Fl HANDBOOK
FIG. 4.9. The input square wave shown at

n~ n
(al (b) (cl
(a) promotes severe "ringing", as shown at
(b), Slight "ringing", as at (c), usually
has little effect on the transient response.

screen of an oscilloscope connected across the output terminals; the amplifier

should be properly matched at both the input and output terminals for this
test. Fig. 4.9a shows the input square wave. Diagram (b) shows a very high
degree of "ringing", which may be of such high amplitude on the peaks as to
cause overdriving of one or more stages of the amplifier. The waveform at (c)
shows a trace of "ringing" which, in practice, may have little significant
effect on the reproduction.

OTHER SQUARE-WA VE TESTS

Square waves can tell us other things about an amplifier; they are useful
because their formation depends upon the fact that they comprise harmonic
components of their fundamental frequency extending well above the limit
of audibility.
The low-frequency performance of an amplifier can be checked by
applying a square wave having a fundamental frequency of, say, 50 c/s.
Fig. 4.10a shows the usual resultant waveform on the screen of the oscillo-
scope. There will be some slope on the top of the waveform, but provided it is
no greater than 40-50 per cent of the height of the waveform, the low-
frequency performance can be considered satisfactory. It is important to
check the oscilloscope on the square wave direct, however, to ensure that the
wave is, in fact, square and that the Yamplifier itself(ifused) is not responsible
for distortion.
Having first ensured that the square wave given by the generator is
maintained in accuracy over the whole of the audio spectrum, in terms of
generator and oscilloscope performance, the square-wave signal applied to
the amplifier can be varied in frequency up to 20,000 c/s, and the display
observed at various intermediate frequencies.
With a good amplifier, the square-wave display should remain essentially
uniform up to about 5,000 c/s, at which point very slight "ringing" may be
FIG. 4.10. The frequency response of
an amplifier can be checked by square
waves; (a) reasonable /ow-frequency
response: (b) poor high-frequency res-
ponse: (c) very poor high-frequency
response: (d) good high-frequency
response.
IL r\ IV flu
(a) (b) (cl (dl

86
 TRACING AND CLEARING FAULTS IN AMPLIFIERS
evidenced. If the amplifier has a poor high-frequency response, the waveform
will deteriorate to that shown in Fig. 4.10b. As the input frequency is
increased up to I0,000 c/s, a good amplifier will maintain a reasonable square-
wave display, similar to that of Fig. 4.10d, but with a poor amplifier the
display may deteriorate from waveform (b) to waveform (c). Increasing the
input frequency up to 20,000 c/s really tests the upper-frequency response of
the amplifier, but with a good hi-fl unit the waveform should differ little
from that shown at (d).

PHASE-SHIFT TESTS
If two voltages of the same frequency are applied to the X and Y terminals
of an oscilloscope, the result on the screen is either a straight diagonal line
or an ellipse. The straight line is produced when the two frequencies are
exactly in phase; an ellipse is produced when the signals differ in phase,
but the dimensions of the ellipse will depend on the phase angle and the
relative amplitudes of the voltages. However, should either of the signals
not obey the sine law, the displays will be irregular in appearance.
Here, then, we not only have a method of checking the phase shift
between the input and output terminals of an amplifier, but also, if we apply
a pure sine wave to the input terminals, we can obtain an idea of the distortion
given by the amplifier. The sequence of patterns shown in Fig. 4. I I illustrates
such a display of two pure sine-wave signals of equal amplitude and fre-
quency, but differing in phase angle from in-phase to 180 deg. out-of-phase.
When the amplitudes of the signals are equal and there is a 90-deg. phase
shift a perfect circle will result, and intermediate ellipses will occur either side
of this point.
The sine-wave signal can be applied to the input terminals of the
amplifier under test in the usual manner and a sample of the signal at this
point applied to the Y terminals of the oscilloscope. The oscilloscope's time-
base should be switched off and disconnected, and the output signal of the
amplifier-that appearing across a correct-value load-applied to the
X terminals. An attenuator may be required at this point so that the Y and X
signals can be balanced. The degree of phase shift occurring over the pass-
band of the amplifier will be revealed on the screen as the sine-wave generator
is tuned over the audio spectrum.
Fm. 4.11. Sequence of pat-
IN PHASE 45° 90° 135° 180°
terns illustrating two pure

ICQ\J\
sine signals of equal ampli-
tude and frequency but
differing in phase angle
from in-phase to 180 deg.
out-of-phase.
87
 THE PRACTICAL HI-Fl HANDBOOK
FIG. 4.12. Phase-shift patterns. The

(1//U/{/
(a) tb) (c) (d) (e)
wa~·<forms at (a) to (e) reveal clipping
of the output signal due to overloading
or incorrect operating conditions oj
the valves: (f) and (g) show the

0/&/
presence of harmonic distortion as the
result of excessive non-linearity:
waveforms (h) and(i) show the presence
,,, (g) ( h) ( i) of a spurious signal.

Deviation in the symmetry of the display will result if the amplifier

output signal is distorted on one cycle only, while if both cycles are equally
distorted, the trace will remain symmetrical, but distorted in shape. Spurious
oscillations in the system will also be shown on the trace. The various effects
are illustrated by the waveforms in Fig. 4. I 2.

CORRECTION OF DISTORTION
Having first established that an amplifier is, in fact, producing distortion,
and that the distortion is not present on the actual programme signal or
caused by maladjusted controls, steps can be taken to locate and remedy the
cause of the trouble. It is a good idea to work from a pure sine wave given by
an audio oscillator or generator. and have this signal fed through the
amplifier under test and monitored on the screen of an oscilloscope. To ensure
correct balance of the circuits, both the input and output terminals should be
terminated by the impedance (resistance) specified in the maker's handbook,
and an output indicator should be connected across the output load. In this
way the output power can be observed in relation to the distortion, and it can
be immediately observed whether or not the distortion varies in magnitude
as the strength of the input signal is varied.
If it is found that distortion is present only towards the maximum output
limit of the amplifier, the most likely cause is overloading of a valve resulting
in its being driven into the non-linear portion of its charact<.'ristic curve.
Low h.t. voltage, due to a low-emission rectifier, or impaired emission of one
of the output valves, is a possible cause of the trouble.
If the h.t. voltage is normal, the oscilloscope can be removed from
across the output load resistor and the signal at the input and output of
the phase-splitter checked for distortion. If there is no distortion at the
input of the phase-splitter, but distortion is present at the output, the phase-
splitter itself may be responsible. However, there is a possibility that grid
current in the output valves is affecting the signal here, and if this is suspected,
a test should be made with the phase-splitter coupling components disconnect-
ed from the output valves. If the signal from the phase-splitter is free from
distortion after this action has been taken, and it remains virtually distortion-
88
 TRACING AND CLEARING FAULTS IN AMPLIFIERS
less when the input signal is increased, there is little doubt that the trouble
lies in the output stage.
The valves should be checked for emission and balance, and the cathode
and grid resistors should also be checked for balance. If all seems well here,
and the valve test is normal, the output transformer should be suspected for
shorting turns. Shorting turns or trouble in the output transformer, apart
from an open-circuited winding, is not always an easy fault to diagnose, and a
suspect usually calls for a substitution test.
The chief symptom in this respect is lack of power, and if the amplifier
is opened-up towards full volume, the reproduction becomes progressively
more distorted without an apparent increase in output power; also, the faulty
transformer tends to overheat. A short-circuit in one half of the primary
winding promotes unbalance in the output stage with a resulting increase in
second-harmonic distortion.

CHECK FOR OUTPUT STAGE BALANCE

Amplifiers with an adjustment for output-stage balance have an arrange-
ment whereby the bias of one valve can be altered slightly in relation to the
bias of the other valve. One method of achieving this is shown in the circuit
in Fig. 4.13. When the stage is in balance, the current in either valve is the
same, and the current is equal but opposite in each half-section of the primary
of the output transformer. Thus, a voltmeter connected across the two
anodes, as shown, will indicate zero voltage when the "balance" control is
adjusted correctly.
It should be stressed that this adjustment serves only from the d.c. aspect
of the circuit, in which case the circuit can be considered as a balanced bridge.
In practice, there is little difference between the setting corresponding to
optimum d.c. balance and that corresponding to optimum signal balance, but
to secure optimum results in the latter case the "balance" control should
be adjusted for minimum distortion, as indicated on a suitable distortion meter.
As the "balance" control usually has a very limited range, its inability
to balance the circuit should first
lead one to suspect low emission HT+

of one of the output valves. If VI

both valves show reasonable -i••____..
balance on a valve tester, the

Fm. 4.13. With o voltmeter con-

nected across the anodes of the
valves, as shown, the "balance"
co'ltrol should be adjusted for zero V2
reading.
89
 THE PRACTICAL HI-FI HANDBOOK
remaining circuit elements should be checked for balance. If a valve tester
is not available, and it is found that the voltmeter pointer remains one
side of zero over the full range of the "balance" control, the position of the
output valves should be reversed. A low-emission valve is definitely respon-
sible if this change causes the voltmeter pointer to remain on the other side
of zero over the full range of the "balance" control.
If changing the valves in this way does not reverse the movement of the
pointer in relation to zero, a change in value of a component should be
suspected. Some amplifiers use separate cathode resistors as well as a
"balance" control, in which case the trouble may well be caused by a change
in value of one of these. There is also a possibility that one half of the
primary of the output transformer has a bad short, causing a decrease in
resistance of one half with respect to the other half. This trouble would
promote severe distortion and lack of power, as already described.
A leak or poor insulation-resistance of one of the coupling capacitors
(Cl and C2 in Fig. 4.13) would also seriously affect the balance of the stage.
These capacitors are in connexion with a source of d.c. at the phase-splitter
side, so that poor insulation would cause the control grid of the associated
output valve to go positive. The negative bias given by the cathode circuit
would thus be neutralized, and the affected valve would pass considerably
more current than the other. Heavy distortion would occur, and it is most
likely that the anode of the affected valve would glow a dull red; in any event
the temperature of the valve would be considerably higher than normal. The
valve would not last very long under this condition, and the resulting
abnormally heavy current would most likely cause failure of the h.t. fuse.
Controls available for balancing the output stage should never be used
as a means of neutralizing severe unbalance caused by alteration in the
characteristic of a valve or value of a component. The control serves essen-
tially to permit a little extra reduction in distortion content which would
otherwise not be possible. Such controls are rarely found in equipment for
sound reinforcement and public-address use, where the distortion content
is in any case greater than that associated with hi-fi equipment.
Optimum balance of the output stage also serves to minimize the residual
mains hum. In fact, balance is sometimes made in this respect; a sensitive a.c.
voltmeter is connected across the output load, and with the signal input
terminals short-circuited the "balance" control is adjusted for minimum hum
voltage. It will be remembered that a similar form of adjustment was recom-
mended for the "humdinger" control.

CHECK FOR SIGNAL BALANCE

If the output stage balances correctly from the d.c. aspect, but slight
distortion is present across the load in spite of a distortion-free drive signal,
90
 TRACING AND CLEARING FAULTS IN AMPLIFIERS
the drive signal should be checked for balance at the control grid of each
output valve. A square-wave or sine-wave signal should be applied to the
input of the amplifier, and the signal amplitude measured at each grid in
turn, relative to chassis, on the screen of an oscilloscope. A high-resistance
or valve voltmeter can be used if an oscilloscope is not available and, if
necessary, a suitable signal can be obtained from the heater supply.
The signals should be almost identical, but opposite in phase. If con-
siderable deviation in amplitude is observed, the phase-splitter valve and
associated components should be carefully checked for value and balance.
With a signal applied, the coupling capacitors (Cl and C2, Fig. 4.13) can be
checked for balance by measuring the voltage across them with an oscillo-
scope or valve voltmeter. Unbalance of these components may not affect the
high- and medium-frequency performance, but may incite harmonic distor-
tion at the low frequencies as the result of the reactance of one capacitor
differing considerably from that of the other.
Harmonic distortion may rise above that stipulated for the amplifier by
open-circuit or low-value of the bypass capacitor across a common cathode
resistor. If such a capacitor is not used even harmonics will be injected into
the grid circuit. This does not apply where separate cathode resistors are used.

CHECK FOR NEGATIVE FEEDBACK

If the distortion is sudden and severe, an investigation should be made
of the main negative-feedback loop if the tests outlined above have failed to
reveal the cause of the trouble. If the negative-feedback loop is in order,
there should be a distinct increase in output, with the input signal kept at a
constant level, on disconnecting the loop either at the cathode, where it is
applied, or at the secondary of the output transformer. No apparent increase,
or only a very slight increase, in output would indicate that the feedback loop
is either open-circuit completely or that the loop resistor has increased in
value. A few simple tests will establish the defective component in this case.

PARASITIC OSCILLATION
Although most amplifiers of hi-fi type have a reasonable margin of
feedback stability, an increase in value of the cathode resistor where the
feedback loop is connected or a reduction in value of the loop series resistor
may increase the feedback above the safety margin and incite parasitic
oscillation.
There is a possibility that the frequency of oscillation will be above the
audio spectrum, in the supersonic region, where its presence will not be
audible as such, but will play havoc with the quality of reproduction. High-
frequency parasitic oscillation will immediately be revealed on an oscilloscope
test of the output signal, but where such an instrument is not to hand, and
91
 PRE· AMPLIFIER MAIN AMPLIFIER AND POWER PACK
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FIG. 4.14. Circuit diagram of the Pye Mozart combined pre-amplifier and power amplifier.
 TRACING AND CLEARING FAULTS IN AMPLIFIERS

FIG. 4.15. The Pye Mozart amplifier, Model HFIO.

the trouble is suspected, a milliammeter connected in series with the h.t. feed
to output valves can be used as to indicate oscillation. A definite drop in
current reading when the feedback loop is disconnected is indicative of trouble
of this nature.
If the feedback components appear to be of reasonable tolerance, the
output valves themselves should come under suspicion, since a severe
unbalance of emission has been known to promote oscillation. In certain
amplifiers low-value anti-parasitic resistors are sometimes connected in
series with the anode and grid circuits of the output valves, and it should be
ascertained that these are in good order.
Other expedients for maintaining stability over the very wide frequency
range characteristic of modern equipment are (I) a capacitor and resistor in
series in the anode circuit of the first valve of the power amplifier, which
serve to reduce the gain at the unstable frequency within the amplifier's
passband, and (2) a capacitor in parallel with the feedback loop resistor. The
latter component promotes a phase shift opposite to that of the output
transformer at the high-frequency resonance of this component, and thus
prevents the feedback from turning positive at this frequency. Such devices
are sometimes adopted in the Williamson amplifier. These components should
be checked for value.
If a replacement output transformer introduces parasitic oscillation, then
it may be necessary to modify slightly the value of the phase-shift feedback
capacitor. The optimum value is best found by trial and error, and if an
oscilloscope and a square-wave generator are available, the value giving the
least distortion and "ringing" at 20,000 c/s should be chosen. The correct
93
 THE PRACTICAL HI-Fl HANDBOOK
value feedback resistor and phase-shift capacitor must be used for the
output impedance selected.
Apart from supersonic oscillations, very low-frequency oscillation may
result from a fall in value of an electrolytic decoupling or filter capacitor;
this may not be directly associated with the h.t. supply, but serve as a low-pass
filter in a voltage amplifier. The effect is usually described as "motor-boating",
but in certain instances the oscillation may be less than 10 c/s and inaudible.
If all the filter capacitors are up to standard, the output valves should be
checked for balance, as also should any push-pull driver valves.
If the feedback connexions on the secondary of the output transformer
are reversed, the feedback will be positive instead of negative, and very bad
oscillation will occur immediately the amplifier warms up. This trouble will
not normally be encountered unless the transformer has been replaced and
incorrectly connected.

TRACING DISTORTION THROUGH THE AMPLIFIER

As a basis for our tests we shall now refer to the circuit diagram of a
commercial amplifier. Fig. 4.14 shows the circuit diagram of the well-known
Pye "Mozart" combined control unit and power amplifier (Model HFIO};
see also Fig. 4.15. This is a remarkable amplifier with a single-ended output
stage, having an output of 10 watts with a total harmonic distortion content
of about 0·3 per cent at 9 watts. It has three inputs-"tape", "radio" and
"pick-up", and an output for connecting to a tape recorder. In addition, it
has a comprehensive tone-control system, a four-position filter and the Pye
"Dialomatic" pick-up compensation, which permits easy matching to any
pick-up. The single-ended output stage is worthy of note, since the design
follows an ultra-linear arrangement centred around a grain-orientated output
transformer.
If distortion is well in evidence, and the tests already described eliminate
the output stage, there are two general methods which can be adopted to
locate the source of the distortion. The procedure, of course, applies to all
amplifiers.
The oscilloscope, having been adjusted for distortion tests and aided by
a distortion-free input signal from an oscillator or generator, can be moved
from the output load to the control grid of each preceding valve in tum,
working towards the input signal. For example, if distortion is present across
the output load, the oscilloscope should be connected to the control grid of
the output valve, the Y-gain adjusted accordingly, and the quality of the
waveform noted. If distortion is still present, the signal should be monitored
at the grid of V2b, then at the grid of V2a, and so on until a point is reached
where the waveform is free from distortion.
Of course, the gain of the Y amplifier will need to be increased as the
94
 TRACING AND CLEARING FAULTS IN AMPLIFIERS
signal is traced towards the low-level sections of the amplifier. It is also
important to avoid overloading the amplifier, and it is best to set the signal
level to the point where distortion just occurs-it is assumed that this is well
below the maximum power output of the amplifier.
Let us suppose that distortion is present at the grid of V2b, but not at the
grid of V2a. It is obvious that the distortion is being produced by mis-
operation of V2a; a likely cause would be low emission of the valve section
itself, though an increase in value of R9 or a leak in CI4 would also cause the
trouble. Attention should also be paid to the components associated with
the cathode circuit, these being related to the feedback network. In this way the
signal can be traced back to its source and any deviation in wave-shape
observed at each point of test.
If an oscilloscope is not available, an actual programme signal can be
applied to the amplifier by way of its appropriate channel, and the signal
monitored at each grid in turn from a pair of headphones or earpiece. In
order to avoid interference from the loudspeaker, the loudspeaker can be
disconnected and its place taken by a suitable resistive load. The point at
which the distortion occurs will quickly be traced by this method, and then
the circuit section can be analysed in detail.
Unfortunately, low-level distortion cannot usually be traced easily by
this method, since headphones are rarely able to detect distortion at a level
of, say, 5 per cent. Indeed, one has to be a very critical listener to detect
distortion at such low level by way of the loudspeaker-programme material
often contains distortion above this figure!

TONE CONTROL AND EQUALIZATION FAULTS

Maladjustment of the various tone controls, equalizers and filters is
possibly responsible for the majority of reports of impaired performance at the
high or low frequencies. Although the purist may be correct in the assumption
that his amplifier has an absolutely flat response only when both the bass
and treble controls are adjusted to the centre of their range, the controls
should, nevertheless, be varied from this ideal setting as a means of securing a
better balance of sound in relation to the listening room and associated
equipment. It is surprising how some enthusiasts are extremely reluctant to
use tone controls for the purpose for which they were designed.
It is also possible, however, that the overall frequency response may be
far from linear at the centre setting of the controls, even though the designer
may have intended a centre balance. Slight alteration in value of components
associated with the tone-control circuits may shift the "linear" point well
towards the end of the range of one (or both) of the tone controls. This
possibility should be suspected if there appears to be a boost of bass or treble
when the controls are set to the centre of their range. In extreme cases, it may
95
 THE PRACTICAL HI-Fl HANDBOOK
be desirable to plot the frequency response of the amplifier to prove this
point.
Maladjustment of the loudness control (if fitted) will incite excessive
bass boost and possibly low-frequency distortion. If this trouble is suspected
the loudness control should be switched out of circuit, or turned right off.
and the volume and tone controls then adjusted in the ordinary manner.
Finally, the loudness control can be brought back into circuit and adjusted
for the correct level of sound, which will automatically give the correct degree
of bass lift.
Some loudspeakers like more bass and/or treble than others, and the
same applies to the listening room, as governed by the acoustics. It is quite
in order to swing the tone controls over the whole of their range to get the
"feeling" of the acoustics of the room and the response of the loudspeaker,
after which the controls can be re-set more critically to give the results most
pleasing to the listener, and most suitable for the programme material.
Newcomers to hi-fi may be tempted to turn on too much bass or treble; this
should be avoided. As G. A. Briggs points out, "if you notice the bass in the
reproduction, or if the extreme 'top' is prominent, then there is something
wrong because you do not notice bass and treble emphasis at a concert".
Too little bass is sometimes caused by misphasing of the loudspeakers
when two separate units are used on the same amplifier. Usually, the speakers
are marked at their terminals with a blob of red paint or a positive and
negative sign so that they can easily be connected together in correct phase.
When in parallel, the red terminals should be connected together (positive to
positive and negative to negative); when in series, a positive terminal should
be connected to a negative terminal, as when connecting batteries. If in doubt,
a small cycle-lamp battery should be connected across the loudspeaker
terminals and the resulting movement of the cone observed. The terminal
of the loudspeaker which is connected to the positive tag on the battery to
cause the cone to move, say, forward, should be clearly indicated with a blob
of red paint.
Distortion of bass is invariably caused by core saturation of the output
transformer. Such trouble is promoted by unbalanced output valves, causing
a greater current in one half of the primary winding than in the other half.
The output stage should be checked for balance by the method already
described.
Another cause of this trouble is low value of one of the output-valve
coupling capacitors. Here the capacitive reactance of the defective component
will be considerably below that of the good component at low frequencies,
thus resulting in overdriving of one output valve with respect to the other.
In addition, the phase of the signal on the grid of one valve at the lower
frequencies will deviate from that provided by the phase-splitter, and the
96
 TRACING AND CLEARING FAULTS IN AMPLIFIERS
phase difference will not be maintained at the ideal 180 deg. This will result
in insufficient cancellation of harmonic distortion in the output load.
If the programme material possesses an abundance of bass, the amplifier
itself may be overloaded at the lower frequencies. Some amplifiers have a
fixed high-pass filter to preclude this trouble, while others have a switched
"rumble" filter to cut the bass at the extreme end of the spectrum, essentially
to obviate transmission of gram motor rumble through the amplifier.
Troubles in the treble may be caused by faulty components in the tone-
control circuits, and this should first be suspected if the tone controls them-
selves appear not to be operating as they should. It should also be ensured
that the equalization control is adjusted to suit the record being played.
Matching of the pick-up and the various programme signals to the amplifier
is most important if the correct response is to be maintained throughout the
system. (Simple pick-up equalizers are considered in a later chapter.)
It should be remembered that the response of certain loudspeakers is
affected somewhat by the damping applied to them from the amplifier.
Insufficient damping-for example, by maladjustment of the damping control
(if fitted)-will sometimes lead to a rise in the low-frequency resonance of
the loudspeaker and an accompanying increase in the bass response. The
bass in this case is of a purely synthetic nature.

HUM TROUBLES
Audio equipment is subject to two kinds of hum. There is the residual
hum which is injected into the h.t. feed circuits as the result of a defective
component associated with the smoothing and filter networks-this being
synonymous with the hum experienced in radio receivers due to a breakdown
of one of the electrolytic smoothing or filter capacitors. Then there is the hum
caused by an alternating mains field being in proximity to the low-level
stages of the amplifier. Here the radiated hum signals are picked up by the
highly sensitive signal circuits, amplified by the equipment along with the
required signal, and emitted by the loudspeaker in the characteristic manner.
Hum which is carried by the h.t. circuits usually presents little difficulty
in remedying. The trouble is invariably caused by a reduction in value or
open-circuit of one or more of the electrolytic filter capacitors. If the effect is
present on a two-unit amplifier, the pre-amplifier should be disconnected
from the power amplifier, and the residual hum level of the power amplifier
noted. If the hum level is little different from that given when both units are
connected, the power amplifier should receive attention.
If the hum is fairly loud, all the large-value capacitors associated with
the h.t. supply should be checked either on a capacitor bridge or by substi-
tuting with good components. The connexions on the capacitors should be
examined and re-made if necessary, and if an electrolytic unit relies for
97
 THE PRACTICAL HI-FI HANDBOOK
negative connexion upon clamp-contact with its case a check should be made
to ensure that there is, in fact, a good low-resistance contact between the two
points concerned.
As almost all hi-fi amplifiers use a full-wave h.t. rectifying circuit,
residual h.t. hum will have a frequency twice that of the mains supply
(l00 c/s in Great Britain and 120 c/s in America), and it will also probably
contain several harmonics of this frequency, thereby distinguishing it from
normal 50 c/s to 60 c/s ripple.
Hum on the h.t. line can be traced with an oscilloscope or a.c. voltmeter
isolated from the d.c. component by a paper capacitor. The hum level at the
output of the h.t. rectifier should be noted, and then compared with the hum
level at the other side of the smoothing choke, and so on through the filter
chain. The hum reading should diminish considerably from section to section.
There is the possibility of a shorting turn in the smoothing choke in cases
where the hum persists. If the main filter capacitors are in order, the a.c.
reading across the choke should be approximately equal to that across the
output of the rectifier; a lesser voltage should lead one to suspect choke
trouble. Smaller amplifiers of the 10-watt rating often use a wire-wound
resistor in place of a choke, and a test should be made to ascertain that
this part is of the stipulated value.
Unbalance of the rectifier valve can also lead to excessive hum, as can a
shorting turn in one half of the h.t. winding on the mains transformer; in the
latter event, the transformer will overheat and emit wax or pitch.
If the hum is just about audible with the signal input socket shorted,
connect a sensitive a.c. voltmeter or output meter across the loudspeaker to
register the hum level and adjust the humdinger control for minimum reading.
If this does not reduce the level sufficiently, try adjusting the "balance"
control, as unbalance of the output valves is another cause of high residual-
hum level.
If the hum becomes prominent only with the pre-amplifier connected
to the power amplifier, impaired h.t. filtering in the pre-amplifier is a most
likely cause, particularly if the hum is present with the volume control at
zero. Electrolytic capacitors should be checked as before and if the pre-
amplifier has a separate humdinger control, this should be adjusted for
minimum hum, as already described.
If the hum is not reduced, poor heater-to-cathode insulation in the final
pre-amplifier valve may well be responsible. The best check is by valve
replacement. The possibility of a hum voltage being induced into the pre-
amplifier/power amplifier connecting lead should also be considered,
especially if the lead has been increased in length for any reason and if the
output of the pre-amplifier is at high impedance. A low-impedance cathode-
follower output circuit is far less susceptible to such spurious pick-up.
98
 TRACING AND CLEARING FAULTS IN AMPLIFIERS
------------------HT+
2mA

RS J-5mA

(12

I-

R 20 R 21 Cl3

..____ COMMON CONNECTING POINT

FIG. 4.16. Circuit of input stage, showing common connecting points.

If the hum level increases as the volume control is advanced, one can be
certain that the hum is getting into the stages preceding the volume control.
Make sure that it is not being induced into an open-circuit signal input
socket by shorting the socket appropriate to the setting of the selector switch.
If the hum is still present, check all electrolytic capacitors, and all valves for
heater-to-cathode insulation. Suspect hum pick-up from stray fields.
Induced hum has been dealt with in Chapter 2, but there are one or two
additional points which are worthy of note. Having first ascertained that the
programme material is free from hum, and that hum is not being picked up
on the programme-source connecting leads, attention should be paid to such
things as high-resistance connexions between "earthing tags" and chassis,
magnetic and electrostatic screens (including valve screens), misplaced grid
or heater leads (particularly if the wiring has been disturbed during a previous
servicing operation), the proximity of mains cables to grid circuits, etc.
It is surprising how much hum can be induced into an amplifier if it
happens to be standing on the floor with the base cover removed, and if there
is a mains cable running beneath the floor at this point! Even if the amplifier
is lifted on to a table in similar proximity to the mains cable, the hum level
may still be well above normal. Never run high-gain amplifiers with the
screens removed, for it is remarkable how much a.c. mains field exists under
domestic conditions. One can prod for hours trying to clear a slight hum
which suddenly disappears on re-orientating the amplifier!
To avoid hum voltages being introduced into a low-level stage from the
"earthy" points of the circuit, a chassis connexion common to the associated
99
 THE PRACTICAL HI-FI HANDBOOK
FIG. 4.17. A bus-bar taking all the
"return" circuits minimizes hum, provid-
ing it is connected at one point only to the
chassis.
HT NEG BUS·BAR

-~+---------
CHASSIS CONNEXION
CHASSIS

circuits is often adopted in commercial and home-built equipment. The idea

is shown in the circuit in Fig. 4.16; apart from a common chassis point, it will
be seen that there is also a common cathode point.
Owing to circulating alternating currents in the chassis itself, particularly
if it carries the power transformer and smoothing choke, the common-
"earthing" device is often taken a stage farther. A heavy-gauge h.t. negative
bus-bar is used for all the earth-return connexions (Fig. 4.17), and this is
"tied" to the chassis at one point only. In this way, there is no danger of the
difference in a.c. mains potential, which may exist between two points on
the chassis when circulating currents are present, being reflected back into
the grid circuits of the low-level stages.
Hum troubles may also be experienced if the pre-amplifier and power
amplifier are earthed separately as well as being connected together electric-
ally. As before, this results in a circulating alternating current, but this time in
the loop between the two earth points and the conductor connexion between
the two units. The disturbance can be reduced considerably by disconnect-
ing the earth from the pre-amplifier.
Whilst this condition may not be incited purposely, it may exist in
slightly different form between, say, a pick-up and pre-amplifier, due to
earthing at the motor as well as the amplifier, or between a f.m. tuner and
pre-amplifier, in which case the tuner may be efficiently earthed in the
normal manner while the earth point of the amplifier is connected to the earth
tag of a three-pin power plug. In both cases a common impedance, in which
is circulating small alternating currents at mains frequency, is developed
between the signal source and the
amplifier input. This presents to the MAINS TRANSFORMER

amplifier a spurious 50 c/s (60 c/s in J

America) signal along with the pro- ~
gramme signal. ~Ns

BIAS - - - -

FIG. 4.18. The .first-stage heaters can be

energized by the rectified h.t. current to
reduce hum injection.
100
 TRACING AND CLEARING FAULTS IN AMPLIFIERS
In very high-gain low-level stages the valve heaters are sometimes
energized by direct current as a means of reducing the hum level. If the
required d.c. is not obtained from a small Lt. rectifier in association with an
1.t. winding on the mains transformer, the h.t. current is suitably adjusted
and allowed to pass through the heater chain.
The basic idea is illustrated in Fig. 4.18. Instead of the centre-tap of the
h.t. winding on the mains transformer being connected direct to the chassis,
as is usually the case, it is first passed through a variable resistance R and
the valve heaters. It is assumed that the total h.t. current is more or less equal
to the current required by the heaters, and any small discrepancy is corrected
by the variable resistor. If the total amplifier h.t. current is, say, 90 mA, and
the pre-amplifier valve heaters are rated at 100 mA, a h.t. bleeder resistor is
connected across the h.t. circuit to pass the additional IO mA, so that the
total current flowing from chassis through the heaters into the centre-tap
matches the 100-mA valve rating.
Capacitor C serves to hold the circuit down to chassis at low frequencies,
and also acts as a part of a filter when the negative voltage, relative to chassis,
at the centre-tap is used as a bias for the output valves.
This arrangement avoids having to run a.c. heater leads into the pre-
amplifier section where either capacitively or inductively they may inject a
hum signal into the grid circuits. Normally, however, if all the basic pre-
cautions are observed, and the heater leads are twisted together to cancel
hum fields, there is little need for d.c. operation these days, particularly with
modern valves such as the EF86.
The practice of providing a slightly positive potential on the heater line,
either from a decoupled potential-divider across the h.t. circuit or from the
cathode of one of the output valves, is frequently adopted in modern equip-
ment. This prevents the a.c.-modulated electrons emitted from the heater.
at the point where it enters and leaves the cathode, from reaching the anode
and causing hum. Since the heater is made more positive than the cathode,
random emission of electrons from the heater section which is outside the
cover of the cathode are attracted back to the heater, and thus do not con-
tribute to the normal electron stream.
The hum and noise levels are usually given as a composite figure in
terms of decibels relative to the full output of the amplifier. Figures range
from - 60 db to - 90 db; for example, the GEC BCS23 I 7/8 is approximately
- 66 db relative to full output ( 12 watts), while the Pye H F25 is given as
- 90 db on 25 watts. In neither case can the hum be heard.
The general "hiss" that a high-gain amplifier gives at full volume
represents the noise output. As already mentioned, this is often referred to
as "white noise", since it is not confined to any particular frequency, and is
contributed mainly by the valves and resistors in the low-level stages.
IOI
 CHAPTER 5

Loudspeakers and Enclosures

LE electrodynamic or moving-coil loudspeaker in hi-fi use

today has exactly the same fundamental principles of operation as the first
moving coil unit ever manufactured, and it remains as popular as ever.
However, whereas in early units the reproduction was severely "coloured"
by the limited frequency response and by disturbing resonances in the centre
of the audio spectrum, development over the years has resulted in the
elimination of most of these shortcomings, at least in the better class of
speaker.
There has been a reduction in all forms of distortion, the weight of the
unit has been reduced and the overall efficiency increased by the use of new
magnetic materials, while the main cone resonance has been pushed well
down to the low-frequency end of the spectrum by improved methods of
cone suspension. Although the response curve is by no means as sleek as that
attributable to hi-fi amplifiers, there are twin-cone units which have a fre-
quency range from 20 c/s to 20,000 c/s.
The moving-coil speaker has a strong magnetic field, produced by a
permanent magnet, in which is placed a free-moving speech coil loaded by a
cone. The speech coil moves axially either into or out of the magnet as the
result of the signal current in the coil setting up a field which interacts with
the magnetic field. The cone thus acts as a piston on the surrounding air and
gives rise to pressure waves.
Apart from being proportional to the length of the conductor (the speech
coil) and the signal current, the driving force-and hence the acoustic
output-is also proportional to the strength of the field produced by the
permanent magnet. The gauss is the measure of the field or flux density;
17,000 gauss being a typical value for a JO-inch unit. The total flux, however,
is measured in maxwells, the gauss unit being equal to one maxwell per
square centimetre. The strong magnets and large pole pieces of modern
units create a total flux value approaching 200,000 maxwells.
The developments which have taken place in the production of permanent
magnets have not only improved the electro-acoustic efficiency of modern
102
 LOUDSPEAKERS AND ENCLOSURES
loudspeakers, but have resulted in improved transient response and extended
high-frequency performance. Smoothness of response over the audio spectrum
is a very desirable characteristic of hi-fi units. Even though the overall
response curve still has its ups and downs, the undulations of a good modem
speaker are mild compared with those attributable to a general-service unit
of a few years back.
Fig. 5.1 shows the response curve of the Model HF816 8-inch unit by
Whiteley Electrical (WB Stentorian). A unique feature of speakers made by
this firm is the universal-impedance speech coil, providing instant matching
to transformers of 3 ohms, 7 ·5 ohms and 15 ohms. The response curve is
made up of major and minor resonances over the frequency range. The major
resonances can be dealt with from the loading aspect, but the minor reson-
ances tend to give small coloration to the reproduction, and since these vary
from speaker to speaker it follows that each speaker contributes its own

+10

:,,_-
V
/ "- ' :--,.

~..,
_,/
' I"
'i--

'
-IO

50 2 3 4 567891 2 l 4 5 678910 1420

HUNDREDS THOUSANDS
C. P. S.

Fm. 5.1. Response curve of the WB 8-inch loudspeaker, Model HF816.

characteristic sound to the reproduction. Normally, this coloration is not

very noticeable when speakers are compared on a complex signal such as
orchestral music, but if a white-noise signal is used, the difference in rendering
of the wide-band signal is remarkable.

CONE AND SUSPENSION

Both the character of the cone and its method of suspension on the
loudspeaker chassis have a bearing on the various resonances evoked. The
new plastic-foam method of cone suspension has brought about a much
flatter overall response. The old method made use of a moulded corrugated
surround of cone material glued to the speaker chassis; the new method does
away with the moulded surround, and the periphery of the cone is suspended
to the chassis by a soft plastic-foam material (see Fig. 5.2). This lighter
suspension tends to damp the cone movement somewhat, and hence improve
103
 THE PRACTICAL HI-FI HANDBOOK
PLASTIC FOAM SUSPENSION FIG. 5.2. The modern method of plastic-
/J \ foam suspension for loudspeakers.

the transient response. It also reduces

the main resonance to a very low
frequency, and has the effect of alleviating cone break-up problems.
Cone break-up is essentially responsible for the irregular response
towards the middle frequencies. At low frequencies, the whole cone moves
up and down after the style of a piston, but as the frequency is raised its
movement becomes so rapid that the direct energy of the actuating force is
not simultaneously conveyed to the whole area of the cone. The movement
towards the speech coil is reasonably faithful, but towards the periphery the
amplitude of the movement decreases, and whole sections of the cone break
up and vibrate in their own mode and phase, as governed by the driving
frequency and inherent resonances. The effect is rather like the waves formed
on a length of rope which is vigorously shaken up and down at one end.
Another approach to this problem is that of the General Electric Co.,
who have introduced a metal cone. The cone is made of Duralumin, and is
thus light and rigid. There are shaped deformations over its area which tend
to neutralize break-up and smooth the frequency response. Further irregu-
larities in the middle-frequency range are reduced by a special "bung" which
is secured to the pole piece. The effective working range of the speaker is from
30 c/s to 20,000 c/s. It has a maximum instantaneous power rating of 12 watts
and a continuous power rating of 6 watts.

TWIN-CONE AND MULTIPLE UNITS

As a means of obtaining a wide overall response from a single unit, a
twin-cone assembly is sometimes used. The main low- and middle-frequency
range cone is so designed that above a certain frequency (known as the
mechanical cross-over frequency) the speech coil effectively becomes
decoupled from the cone and very little sound energy is produced by the
main cone. However, tightly coupled to the speech coil is a very light and
small cone which takes over at this point and maintains the response to the
high-frequency end of the spectrum. The Axiom range of Goodman's loud-
speakers uses this principle. The Axiom 80, a 9½-inch unit, has a frequency
range from 20 c/s to 20,000 c/s, while the 12-inch units have a range from
30 c/s to 15,000 c/s.
Instead of using a single loudspeaker to cover the whole range of audible
frequencies, two or even three units may be used, each being designed to
provide optimum results over a limited band of frequencies. This method
eases the design problems associated with full-range units, eliminates
104
 LOUDSPEAKERS AND ENCLOSURES
undesirable compromises in design and brings about a marked reduction in
intermodulation distortion.
Sounds occupying the low- and middle-frequency portions of the
spectrum are served by an ordinary moving-coil unit of fairly large dimensions,
while the top frequencies are catered for by a much smaller unit designed for
a smooth response up to 20,000 c/s. Some loudspeaker systems include an
additional unit for the reproduction of the middle range of frequencies, in
which case the low-frequency unit operates over the range of about 1,000 c/s,
at which point the middle-frequency unit takes over and responds up to about
5,000 c/s. The high-frequency unit then gradually takes over and caters for
the remainder of the spectrum. The names of "woofer", "squawker" and
"tweeter" are sometimes given to the low-frequency, middle-frequency and
high-frequency units respectively.
A frequency-dividing network (cross-over unit) is used to split the output
between the loudspeakers so that there is no call upon any unit to handle
large amplitudes of frequencies beyond its range. Since the change-over from
one speaker to the next is gradual, and occurs at the cross-over frequency of
the dividing network, each unit should be capable of handling at least half an
octave beyond the cross-over frequency at full power, and two to three
octaves beyond the cross-over frequency at reduced power. The type of
dividing network utilized has some bearing on this.
Apart from improvement in the directivity characteristics and response
to transients, multiple-speaker systems have a distinct advantage in the
reduction of distortion arising from the Doppler effect. Briefly, the Doppler
effect is that of an increase in frequency when the source of the signal is
advancing and a decrease when it is receding. We have all heard the change
of pitch of a whistle of a railway engine when it passes through a station at
considerable speed; this is an example of the Doppler effect from the aspect
of sound waves-the greater the velocity of the source of the sound (or radio
signal), the greater the change in frequency.
So far as the loudspeaker is concerned, the movement of the cone at low
and middle frequencies is often of the order of plus and minus ¼in. and,
depending upon the actual frequency, the velocity of the cone is of consider-
able value. If, while the cone is being actuated by a relatively low-frequency
sound, there is introduced a sound of greater frequency, then this will suffer
a change in pitch since the source of the sound is moving at high velocity,
and a Doppler discord will result. During the time that the cone is moving
forward under the control of the low frequency, the pitch of the high-frequency
sound will appear to increase at the front of the speaker, and decrease at the
rear, thus doubling the net effect of the discord between the two surfaces of
the cone. If the frequencies are segregated over a number of loudspeakers, the
distortion is considerably reduced.
105
 THE PRACTICAL HI-Fl HANDBOOK
MIDDLE- AND HIGH-FREQUENCY UNITS
Ordinary 8-inch or IO-inch moving-coil units are often used to cater for
the middle frequencies, and these are quite successful provided a good cross-
over unit is adopted. There are, however, more specialized pressure-driven
horn-loaded units, such as the Goodman's "Midax". This has a frequency
range from 400 c/s to 8,000 c/s, the recommended cross-over frequencies
being 750 c/s and 8,000 c/s, and it can handle something like 25 watts
(British rating).
Pressure units are often used for the reproduction of the middle and high
frequencies in conjunction with a horn for the purpose of increasing the
acoustic loading on the cone or diaphragm. The pressure unit is, in effect, a
moving-coil loudspeaker, but employing an aluminium-alloy diaphragm
instead of a conventional cone. This reduces the mass, which is desirable for
high-frequency work, while permitting easy pressure loading to the horn.
The horn is usually of exponential nature, though other shapes, such as
conical or parabolic, are sometimes used. The overall length of the horn
determines the lowest frequency at which it will load adequately. For a
reasonable low-frequency response the horn may have to be several feet in
length. For the middle range of frequencies, the length is less of a problem
(the overall length of the Midax is 18-i\ in.), while for the high frequencies, a
horn length of a few inches is all that is required.
Since horn loading increases the efficiency considerably above that of
direct-radiator moving-coil units, variable attenuators are usually required
in the lines feeding the horn-loaded units, so that the sensitivities can be
easily matched.
Apart from horn-loaded pressure units using a diaphragm, ribbon loud-
speakers are sometimes used, also in conjunction with a horn, for the faithful
reproduction of high frequencies. This type of unit has a very thin corrugated
aluminium ribbon suspended in a powerful magnetic field. The ribbon itself
serves as the conductor, being analogous to the speech coil of conventional
moving-coil units, and through this are passed the audio-frequency currents
from the output transformer. The ribbon vibrates in sympathy with the
original sound, and is pressure-loaded to a small horn of dimensions to match
the higher audio frequencies.

PLESSEY IONOPHONE
A remarkable high-frequency reproducer which has no moving parts at
all is the Plessey ionophone, an invention of Mr. S. Klein of Paris. The func-
tional portion of the unit is a small quartz-glass tube. One end of this is open
and the other drawn down to a small hole in which is inserted an electrode
known as the "Kanthal" (see Fig. 5.3).
106
 LOUDSPEAKERS AND ENCLOSURES
FIG. 5.3. The functional portion
of the Plessey ionophone.

A glow discharge is
arranged to take place in the
air within the open end of
the tube by applying a high CONTACT PLUG BRIMISTOR INNER ELECTRODE

voltage at radio-frequency
(KANTHAL)
~ ✓ (KANTHAL)

across the Kanthal contact

plug and the outside counter
electrode. The r.f. oscillator
providing the discharge voltage is modulated by the amplifier's a.f. signal,
and since the intensity of the discharge at any instant is proportional to the
instantaneous value of the applied r.f. voltage, the glow discharge will vary in
sympathy with the a.f. signal. Because one end of the quartz tube is closed,
pressures varying in sympathy with the a.f. signal will be set up in the open
end of the tube, and these are conveyed directly to the mouth of an exponen-
tial horn for resolution into sound waves, of frequency range governed by the
dimensions of the horn.
The Plessey unit is designed to work into a cross-over frequency of some
2,000 c/s, and from this frequency the response is perfectly smooth up to
about 17,000 c/s. Since the arrangement uses no moving parts, the response
to transients approaches the theoretical ideal.
The oscillator unit is built into the rear of the horn mounting, and is
arranged to operate at 27 Mc/s, while the power-pack and modulation
transformer are carried separately. Fig. 5.4 shows the unit in more detail,
and its connexion to the output circuit of the oscillator, while Fig. 5.5 gives a
suggested arrangement for the cabinet mounting of the ionophone, power-unit
and Plessey 15-inch loudspeaker for reproduction of the frequencies below
the 2,000 c/s cross-over.
t
ELECTROSTATIC UNITS
Of recent years con-
I
71n.

l
siderable attention has been
focused on the electro-
static type of loudspeaker
and its potentialities as a OSCILLATOR
OUTPUT HORN SOCKET
CIRCUIT ~/

MAIN ~.rHORN
FIG. 5.4. Connexion of the
ELECTROD;/__~~
/onophone to the output cir-
COUNTER / QUARTZ FINE GAUZE
cuit of the r.f. oscillator. ELF.CTROD~ CELL

107
 THE PRACTICAL HI-Fl HANDBOOK
FJG. 5.5. Suggested cabinet and
mounting for the ionophone.

full-range unit. High-

frequency electrostatic
units have been available
DAMPING
MATERIAL
for a number of years, and
are adopted in a number of
15in L.F UNIT
commercial radio sets,
as well as in hi-fl loud-
speaker systems. The prin-
MODULATION ciples involved for the
TRANSFORMER
reproduction of sound by
::::;::.-.::::..-::::.-:::..~·\ VENTED
POWER UNIT
this means are not new
ENCLOSURE
by any means, but the
development of sound
reproduction has revived interest in electrostatic units, formerly known as
"condenser loudspeakers".
As will be seen from the basic arrangement of a simple unit in Fig. 5.6,
two plates are used, as in a capacitor. One plate is fixed and perforated, while
the other is very thin and movable, and mounted in such a way that it can
vibrate without touching the fixed plate. In some designs the loudspeaker is
curved to widen the angle of sound radiation.
The signal from the output stage of an amplifier (at high
impedance) is applied across the two plates, sometimes through an isolating
capacitor (as in Fig. 5.6). A fairly high voltage varying at the signal
frequency is thus present across the plates, and this gives rise to mechanical
forces which cause the thin movable plate to vibrate in sympathy with the
sound signal. This is the normal
electrostatic action. Under this con- I
dition, however, the movable plate I
will be attracted towards the fixed I
MOVABLE :/FIXED PLATE
plate on every half-cycle of the signal, PLATE
and two vibrations will occur on I
each full cycle. To avoid this useless I POLARIZING
function a polarizing voltage is applied I VOLTAGE

along with the signal, across the

'l t
I
A.F.SIGNAL
LIMITING
I"IG. 5.6. Basic arrangemelll <!fa simple
electrostatic loudspeaker.
RESISTOR
~._J
108
 LOUDSPEAKERS AND ENCLOSURES
FIG. 5. 7. One method by which an electrostatic -------HT

unit can be coupled to the output stage.

\I [}J ~o~6~~E~KER

plates, as shown in Fig. 5.6. The polariz-

ing voltage causes an initial attraction s1GNAL
IN
of the movable plate towards the fixed ELECTROSTATIC
LOUDSPEAKER
plate, and provided the peak signal
voltage does not exceed the polarizing
voltage, the plate will vibrate in correct
unison with the signal pattern. The
limiting resistor in the circuit prevents
the low-impedance polarizing source from short-circuiting the signal.
For the simple application of an electrostatic unit, the h.t. voltage at the
anode of the output valve can be used as a polarizing voltage, as shown in
Fig. 5.7. This voltage is varying at the signal frequency, and is fairly high
(and at high impedance)-which is just what is required.
More elaborate electrostatic units employ a push-pull system (sometimes
known as bilateral) in which the movable plate is situated ~tween two fixed
plates (Fig. 5.8). This idea is extended to full-range units, such as the Quad
electrostatic loudspeaker (developed by P. J. Walker, of the Acoustical
Manufacturing Co.). The two fixed plates are energized by equal and opposite
signal voltages from the secondary of a push-pull transformer, the primary
being connected to the signal source. The polarizing voltage is applied, along
with the signal as already described, from a small e.h.t. generator. As current
is not required for this operation, a simple generator similar to that used in
certain television receivers is all that is required.
Electrostatic units can be loaded either by a horn or by an ordinary
reflex cabinet, but certain designs provide automatic loading and do
not usually require artificial means.
I MOVABLE
_, PLATE
CROSS-OVER NETWORKS
FIXED PLATES
A high-frequency unit or tweeter can
be added easily to any loudspeaker
system to augment the high-frequency
response, as shown in Fig. 5.9 (a).
In effect, the tweeter is supplied by way
"' POLARIZING
_;_ VOLTAGE
of capacitor C, the reactance of which
decreases with rise in frequency, and
thus limits the current in the speech coil
'
PUSH-PULL
TRANSFORMER
FtG. 5.8. Push-pull electrostatic system.
AF SIGNAL

109
 THE PRACTICAL HI-FI HANDBOOK
(Left) Fm. 5.9. Simple
....
I-
z
method of connecting a
AMPLIFIER
OUTPUTU541
BASS
SPEAKER .......
:,
u
TWEETER tweeter (a), showing its
operating curve (b).
154
TWEETER
...
I-
(Below) Fro. 5.10.
~ Showing how a tweeter
FREQUENCY volume control can be
(a) (b) introduced.

at low frequencies. The curve in

Fig. 5.9 (b) illustrates the action
graphically. BASS
AMPLIFIER
If the tweeter is required to ouTPUTUSJU LOUD·
SPEAKER
start taking over at about 5,000
c/s, then at this frequency the
reactance of C should equal
the impedance of the tweeter.
Excluding any inductive effects of the tweeter's speech coil, and assuming
an impedance of 15 ohms, a capacitor of 2 mF will serve the purpose (a
2-mF capacitor has an impedance of 15 ohms at 5,000 c/s). Under this
condition, the current in the tweeter's speech coil at 5,000 c/s will be approxi-
mately 30 per cent below that in the speech coil of the bass unit: 5,000 c/s
can thus be considered as the cross-over frequency (or more accurately,
the take-over frequency).
If the speech coil impedance is, say, 7·5 ohms, or if a tweeter volume
control is used, as shown in Fig. 5.10, and the "effective" impedance of a
I 5-ohm tweeter is reduced approximately to 7 ·5 ohms, then the value of C
should be doubled. The same applies if it is required to lower the "take-over"
frequency one octave when using a 15-ohm unit.
In the circuit in Fig. 5.II (a) a choke L has been interposed in series
with the bass speaker, the treble section remaining as in Fig. 5.9 (a). An
inductor or choke has a reactance which is opposite to that of a capacitor;
it rises with increase in frequency, and thus limits the amount of current in the

CROSS-OVER
FREQUENCY
t BASS i TWEETER O
AMPLIFIER I
OUTPUTUSn) 3db
BASS
SPEAKER
IS.II.
FREQUENCY_
(a) (bl

Fro. 5.11. Simple quarter-section cross-over filter (a), and its operating curve (b).
110
 LOUDSPEAKERS AND ENCLOSURES
(Right) FIG. 5.12. Circuit of a half-
section cross-over filter. (Below) Fm.
5.13. A more elaborate arrangement AMPLIFIER
OUTPUT(R)
for three speakers. BASS TWEETER
C SPEAKER

21'

AMPLIFIER ,mH so.n.

OUTPUT(l5ll)
MIDDLE
SPEAKER

811 15Jl. 15.ll.

speech coil of the bass unit at high frequencies. In this way a simple quarter-
section parallel cross-over filter results. Provided all the impedances are
15 ohms and essentially resistive (which, unfortunately, is not always the case
in practice), a 5,000 c/s cross-over can be secured by using an inductor
valued at 0·5 mH and a 2-mF capacitor. The curves in Fig. 5.1 I (b) illustrate
the operation of the filter. These show that the current in the speech coils of
both units at the cross-over frequency is 3 db below full current. This has
little effect on the acoustical power, since both units will be contributing
towards the total sound more or less equally.
This simple quarter-section filter is not always favoured owing to each
section having an attenuation of only 6 db per octave at the cross-over
frequency, and the greater attenuation of a half-section arrangement is often
preferred. The circuit of such a network is given in Fig. 5.12. It will be seen
that it is very similar to the quarter-section filter already given, but it has the
addition of an inductor across the tweeter and a capacitor across the bass unit.
The result of having these is to speed up the rate of attenuation at the
cross-over region by a further 6 db per octave, thereby giving a total rate
of attenuation of I 2 db per octave.
As this kind of filter is extensively adopted, the following expression
relating the component values with the cross-over frequency will be useful:

R 103
l(mH)=---
1r fc v2
)08
C(mF) = 21rfc Rv2
111
 THE PRACTICAL HI-Fl HANDBOOK

F1G. 5.14. Quarter-section series cross-

BASS over network. With 15-ohm speakers,
SPEAKER
AMPLIFIER the component values given provide a
OUTPUT 1,000 c/s cross-over.
TWEETER

where R is the common impedance of the amplifier output and speakers in

ohms and Jc is the cross-over frequency in c/s.
Fig. 5.13 shows a more elaborate network suitable for three loudspeakers,
in which the bass and middle units cross-over with a 12-db per octave slope,
while the tweeter has a 6-db per octave take-over.
There are a host of cross-over networks, ranging from the simple high-
pass/low-pass arrangement forming the quarter-section parallel system
already discussed to a complex full-section arrangement incorporating three
inductors and three capacitors. So-called "series" and "parallel" formations
are used in practice. A quarter-section series network is given in Fig. 5.14,
which makes interesting comparison with the quarter-section parallel circuit
in Fig. 5.11. Similarly, the half-section series network in Fig. 5.15 can be
compared with the half-section parallel network in Fig. 5.12.
The values given to the components in the circuits in Figs. 5.14 and 5.15
provide a cross-over in the region of 1,000 c/s when used with 15-ohm
speakers. Since the component values are inversely proportional to the cross-
over frequency, the cross-over is lifted an octave by halving the values, and
dropped an octave by doubling the values. If the speaker impedance is halved,
the capacitor values should be doubled and the capacitor values halved, to
maintain the cross-over frequency for 15-ohm units.
All the networks given are of the constant-resistance type, since these
are preferred by most equipment manufacturers and users. It should be
noted, however, that all cross-over networks are much of a compromise-
even though they perform well in practice-since the impedance of the speech
coils of the loudspeakers is not maintained at a constant 15 ohms throughout
the audible range. The theoretical ideal is to employ a power amplifier for
each speaker, the high-frequency amplifier and speaker being fed from the
pre-amplifier by way of a high-pass coupling, and the low-frequency amplifier

BASS
SPEAKER
FIG. 5.15. Half-section series cross-over 4MPLIFIER
network. With 15-ohm speakers, the OUTPUT
TWEETER
component values given provide a 1,000-
c/s cross-over.

112
 LOUDSPEAKERS AND ENCLOSURES
and speaker through a low-pass coupling, each coupling being arranged to
have the required characteristic shape and attenuation rate.

SPEAKER PHASING

Two or more speakers are in phase when their cones move together in
perfect unison under the control of the same signal. If one cone moves in
while the other moves out, then the speakers are exactly 180 deg. out of
phase. Intermediate phase displacements occur between these two extremes.
For example, at the cross-over frequency, a quarter-section cross-over
network usually introduces a phase displacement of 90 deg. For this reason,
it is often desirable to position the tweeter one-quarter of a wavelength
behind the plane of the bass unit, so that the sounds from the two units reach
the ears of a listener at the same time (provided the speakers are phased
correctly from the d.c. point of view). A cycle-lamp battery can be used to
check this, as already described.
Incorrect phasing often gives the effect of "emptiness", "disembodied
treble", lack of bass or lack of middle register, depending upon the frequency
range fed to the units and the acoustics of the listening room.

MATCHING
An amplifier can only give its maximum undistorted output when the
loudspeaker system represents a perfect match to its output terminals. If
two 15-ohm units are connected in parallel across the output terminals, then
the speaker impedance as "seen" by the amplifier is 7·5 ohms. Similarly,
two 15-ohm units connected in series add up to a total impedance of 30 ohms,
and in both instances the impedance adjustment on the amplifier must be
altered to correspond.
Generally speaking, it is not a good idea to connect two hi-fi speakers in
series, even though the amplifier output is adjusted to match the sum of the
two impedances. The reason for this is that the actual resistances of
the speech coils are also added in series, which has an adverse effect on the
damping mechanism.
From Chapter 2 it will be recalled that the resistance of the speech coil
represents the dominant impedance at high damping factors. When the two
speakers are connected in series, each speaker "sees" something like 30 ohms-
its own resistance in series with the resistance of the other speaker across the
low source resistance of the amplifier. When they are in parallel, however,
each speaker "sees" only its own resistance of 15 ohms or thereabouts.
In other words, the introduction of another series-connected speaker is as
futile as putting a resistor in the speech-coil circuit, and then trying to damp
the actual speaker by reducing the source impedance of the amplifier.
Speakers connected in series tend to have a very poor transient response,
113
 THE PRACTICAL HI-Fl HANDBOOK
and disconcerting "hangover" effects will develop. The same applies with
series-parallel networks. These arrangements may suit public-address
systems, where ease of connexion of speaker units is demanded, but they
should not be used with hi-fi or sound-reinforcement equipment.

BAFFLES AND VENTED ENCLOSURES

Sound is emitted from both sides of the cone of a loudspeaker, and as a

compression wave is formed on one side, a rarefaction wave is formed on the
other side. At the lower frequencies, where the wavelengths of the sound waves
approach the dimensions of the cone, sound waves from the rear tend to
cancel sound waves from the front and the acoustical response of the unit
diminishes almost to zero as the frequency is lowered. This undesirable
effect can be avoided by mounting the unit on a baffle board so that the rear
and front sound waves are isolated. Since the baffle cannot have infinite area,
front-to-back cancellation will occur at some low frequency-governed by the
dimensions of the baffle used. So that the cancellation will not be common in
all dimensions, as in the case of a circular baffle on which the speaker is
mounted in the exact centre, a rectangular baffle should be used and the
speaker displaced from the centre.
A near approach to the infinite baffle is secured by mounting the speaker
on a wall dividing two rooms. This idea is used by many enthusiasts, and will
be encountered from time to time by the technician. Disconcerting irregu-
larity of response can be evoked by this method, however, unless the hole
in the wall is bevelled so as to avoid the cylinder formed by the wall thickness
from resonating. Some enthusiasts make an extra large hole and introduce a
small sub-baffle on which the speaker is mounted.
Baffles for treble and middle units need not be as large as their counter-
part for the bass unit, though they should be large enough to respond down to
an octave below the cross-over frequency.
A synthetic method of providing infinite baffle characteristics is to
enclose the rear of the loudspeaker in a substantial air-tight box. The air
trapped in the box acts as a cushion on the cone and thus provides acoustical
damping, whilst also reducing the main cone resonance, depending upon the
volume of the box.
A vented enclosure encloses the rear of the loudspeaker, is formed of
a substantial box and has a vent hole cut in one of its walls, usually near the
speaker aperture. The enclosure thus acts as a resonator (see Chapter I),
the resonant frequency being governed by the volume of air within the
enclosure and the area of the vent. The effect that this has on the loudspeaker
and acoustical response is rather interesting. At the resonant frequency of
the enclosure most of the sound is radiated by the vent and the speaker cone is
subjected to maximum acoustical damping. For this reason it is often
114
 LOUDSPEAKERS AND ENCLOSURES

Fm. 5.16. Impedance characteristics of a

speaker correctly loaded by a reflex t LOWER CONE
RESONANCE
I
UPPER CONE
RESONANCE
I
I
enclosure. I

advantageous if the cone and enclosure wu

z
resonances coincide in frequency. An a<
w
increase in low-frequency radiation is Q.

secured whilst the disturbing effect of the !

speaker resonance is alleviated. This,
FREQUENCY -
however, represents an ideal situation.
As the frequency decreases below the resonance of the enclosure, the
output from the vent falls off, but the output from the speaker rises due to a
lowering of its resonant frequency as the result of the air-pressure build-up
between cone and vent. However, since the sounds from the two sources
approach the anti-phase condition, the resultant sound output diminishes
fairly quickly, and no useful increase in bass response is obtained by lowering
the resonance of the loudspeaker.
As the frequency rises above the resonance of the enclosure, the loading
on the speaker cone is lifted and its resonant frequency rises. Output from
the vent tails off and the normal middle- and high-frequency characteristics
of the speaker take over.
From the foregoing it will be realized that two auxiliary resonances of
the cone occur on either side of the enclosure resonance. These are often
illustrated in the form of a curve as shown in Fig. 5.16.
MEASURING SPEAKER IMPEDANCE
As an impedance curve of this kind provides useful information with
regard to the loading of a speaker by its enclosure, it will be instructive at this
stage to consider a method by which speaker impedances can be taken.
Precise results require the use of an impedance bridge, but as such an instru-
ment is not generally available to the enthusiast or technician other methods
less exacting but practicable, will be described.
Fig. 5.17 shows a set-up which utilizes the Ohm's Law method. The
voltage across the speech coil and the current in it are measured throughout
the frequency range by the a.c. voltmeter V and the a.c. ammeter A. The test
signal is provided by an audio oscillator, and the necessary power produced
by the amplifier (any good hi-fi amplifier will serve). The impedance value at

AUDIO
LINEAR AMPLIFIER SPEAKER
OSCILLATOR

FIG. 5.17. Set-up for measuring speaker impedance.

115
 THE PRACTICAL HI-Fl HANDBOOK

FIG. 5.18. Experi-

mental impedance
curve of Goodman's
Axiom 12-inch unit,
Type 150, Mk. ll,
in a reflex enclosure
which has a 1½-inch
slate front.

12---~--~----.__. . . . . . . . . . ..___...__. . . . .
10 100 1000
FREQUENCY c/s

any frequency is computed by dividing the voltage reading by the current

reading (in volts and amps). Thus, if 3 volts at 0·2 amp. are indicated, the
speaker has an approximate impedance of 15 ohms at the frequency of
the test.
Fig. 5.18 shows an experimental impedance curve using the above method
of measurement. The speaker unit was a Goodman's Axiom 150 Mk. II, and
was mounted in a reflex enclosure having a slate front of 1½in. thickness.
It will be seen that the two resonance peaks either side of the enclosure's
resonance are nicely spaced and of equal amplitude, indicating that the
speaker is ideally loaded by the enclosure.
Another method of checking impedances (see Hi-ft News, February,
1957) is shown in Fig. 5.19. Here a 30-ohm I-watt resistor is interposed in the
speech-coil circuit and the voltage across it measured by a Model 40 A vometer
set to the 2·5-volts a.c. range. The speaker terminals are short-circuited and
the oscillator and amplifier controls adjusted to give full-scale deflection
-making sure that the oscillator is not overloading the first stages of the
amplifier. The short-circuit is then removed, and the impedance of the speaker
in ohms, at any particular frequency, is given on the "ohms" scale divided by
100. Thus, if the meter reads 2,000 ohms on the scale, the impedance is
20 ohms.
The signal voltage from the AYO-MODEL 40
amplifier must remain constant
throughout the frequency test range,
but this can be monitored, if
30AIW
SIGNAL
FIG. 5.19. Another method for impedance FROM AMP.
measurement, using a Model 40 Avo-
LOUDSPEAKER
meter.

116
 LOUDSPEAKERS AND ENCLOSURES

Fm. 5.20. Suggested en-

closures for the TSL Lorenz
LP 312-2 wide-range
speaker system.

necessary, by an a.c.
voltmeter or on an
oscilloscope. The output
voltage from most hi-fi
amplifiers is reasonably
consistent over the fre-
quency spectrum, pro-
vided the oscillator
voltage applied is con-
stant.
A few words of
warning: the impedance
curve is not an indication of acoustical output from the speaker. The
only way that this can be assessed is by measuring the sound field produced
by the speaker system. Since quite a lot of sound comes out of the loud-
speaker during impedance tests, first make sure that the neighbours are
out or that the test room is sound-proof! Even though hi-fi enthusiasts
experiment late into the night, be absolutely certain that the neigh-
bours are not in bed, for it is surprising how the bed springs can resonate
when the oscillator is tuned to the critical frequency and plenty of watts
are emitting from the speaker-even though the springs are heavily
damped by the bedding.

ENCLOSURE CONSTRUCTION AND DIMENSIONS

Reverting to enclosures, the cabinets should be of solid and rigid con-
struction. A i-in. wall thickness is desirable, though double walls of thinner
material can be used if the space between the walls is filled with dry sand. At
least three walls, at right-angles to each other, should be lined to a depth in
excess of I in. with sound-absorbing felt or plastic foam, as a means of reduc-
ing standing waves within the enclosure. Acoustic curtains are hung in the
enclosure for the same reason.
The size of a reflex cabinet, for a given low-frequency response, can be
reduced by the inclusion of an acoustical filter over part of the vent. Such
filters take various forms, one type being marketed by Goodmans under the
name "acoustical resistance unit". The filter reduces spurious resonances,
while also minimizing the magnitude of the upper resonance which, as we
have seen, is a characteristic of conventional reflex enclosures.
117
 THE PRACTICAL HI-FI HANDBOOK

FIG. 5.21. The Pamphonic Victor

speaker system. This has a treble unit
at the top, the volume of which can
be controlled from the side of the
cabinet. The preferred position for this
system is in the corner of a room.

In Fig. 5.20 is given the

dimensions for suggested en-
closures for the TSL Lorenz Type
LP 312-2 loudspeaker system.
This is a 12-inch unit, which also
incorporates two Lorenz Type
LPH 65 treble units mounted in
front of the cone. Its overall
frequency response is from 20 c/s
to 17,000 c/s.
Fig. 5.21 shows the Pam-
phonic "Victor" speaker system.
Apart from the large bass unit,
this has a cone-type treble unit
at the top of the enclosure. A
cross-over network is contained
in the enclosure, and the treble unit is fed by way of an attenuator which can
be adjusted to give the correct degree of "presence". The treble unit is fully
enclosed so that there is no interaction between the two sound sources, and
it is mounted at an angle so that the sound is directed towards the corner of
a room, whence it is deflected into the room at large.

OTHER ENCLOSURES
There are numerous other enclosures and speaker-loading devices used
by enthusiasts, full constructional details of which are given elsewhere (i.e.,
in manufacturers' literature and in "Sound Reproduction" by G. A. Briggs).
It is not intended to study them all here, but a few words on the more popular
arrangements will not be amiss.
The exponential horn in relation to treble and middle-range speakers
has already been mentioned. Whilst this system represents the most efficient
way of loading a speaker, its large size at the lower frequencies is a dis-
advantage. Nevertheless, horn loading in the bass is adopted, usually by
folding the horn in various ways within an enclosure.
The tuned-pipe arrangement is worth consideration. The pipe has one
closed end and is critically dimensioned so that a fundamental anti-resonance
118
 LOUDSPEAKERS AND ENCLOSURES
occurs at the frequency of the major low-frequency resonance of the speaker.
This gives the system a characteristic similar to that of the reflex enclosure,
whilst being less difficult to manufacture. The principle of operation is rather
like that of an organ pipe, though in some cases the pipe is tapered and the
speaker mounted one-third of the overall length away from the closed end.
At resonance the sound radiated from the open end is out of phase with that
radiated from the cone.
The acoustical labyrinth is a type of enclosure which avoids resonance
effects, but has a slight falling off at the low-frequency end of the
spectrum. It consists essentially of a very long pipe (about 11 ft.), heavily
lined with a thick felt or other acoustic damping material, and has the
effect of completely absorbing radiations from the rear of the cone.
Another method of securing adequate low-
frequency performance is by the use of a battery
of some nine speaker units mounted on a wood
baffle with short sides, which is left open at the
rear. This idea is sometimes used by American
enthusiasts .

. LINE-SOURCE SPEAKER
One type of loudspeaker system, used parti-
cularly for sound-reinforcement applications
rather than domestic hi-fi work, consists of
several speaker units mounted close together
one above the other in a wood or metal enclosure,
depending on whether it is to be used indoors or
outdoors. In the latter case the enclosure is
weather-proofed. These loudspeaker systems are
sometimes called "sound columns", and by
Pamphonic Reproducers, Ltd., "line-source
loudspeakers". Fig. 5.22 shows the Pamphonic
system, with the units mounted one above the
other. All the units are connected in phase, and
the small units are connected by way of a suitable
cross-over network to cater for the high
frequencies.
The reason for mounting the units in

FIG. 5.22. The Pamphonic line-source loudspeaker

system, showing how the speaker units are mounted
in a straight line one above the other.

119
 THE PRACTICAL HI-Fl HANDBOOK

FIG. 5.23. The sound is radiated from a

line-source speaker system rather like the
flat broad beam of light emitted from car
fog lamps.

a straight line is to secure a direc-

tional effect in the radiated sound
rather like the flat, broad beam of
I ight emitted from certain car fog lamps
(see Fig. 5.23). Whilst the length of
the column does not affect the
horizontal distribution, which approxi-
mates 120 deg., it does affect the vertical distribution-the longer the
column the smaller the vertical angle of distribution and the more con-
centrated the vertical distribution of the sound.
The greatest advantage of this concentrated distribution is that very little
sound energy is wasted unnecessarily outside the range of the listeners. This
leads not only to greater efficiency, but also avoids difficult reverberation
problems which always arise in lofty buildings when fairly high-level sound
is provided by single-source speaker systems. Service technicians having
experience of public-address installations in swimming baths and churches
will appreciate how the hard reflecting surfaces of such buildings evoke many
indirect reflected sounds as well as the directly radiated sound. In extreme
cases, speech is made almost unintelligible. Line-source speaker systems
alleviate this trouble to a large degree owing to the reduced radiation of
superfluous sound.
During tests with Pamphonic line-source equipment in St. Paul's
Cathedral, the improved efficiency of line-source speakers was demonstrated
by the fact that a power-input of only one watt provided sufficient sound from
one I I-ft. line-source speaker standing beside the pulpit to cover the whole
dome seating area of 9,000 sq. ft.
Another advantage of the system is that a listener situated close by a
column is not disturbed by a very high sound level as he would be close by a
high-level single source unit. Because the intensity of sound at a short distance
from the column is less than the sum of the sound intensities of all the units,
acoustical feedback between the microphone and line-source speaker
represents less of a problem than when single-source units are adopted. The
microphone can, in fact, be brought remarkably close to a line-source unit
before feedback takes place, even when the amplifier's gain control is set for
a normal level of reinforced sound.
For a large number of installations a single line-source unit is all that
is required, for both indoor and outdoor use, and from the practical aspect
120
 LOUDSPEAKERS AND ENCLOSURES

FIG. 5.24. Two Pamphonic line-source speakers mounted in the rear of the dome oj
Rhodes House, Oxford, in connexion with the Duke of Edinburgh's Study Conference
held there. The imtal/ation was carried out by Lowe & Oliver Ltd.

121
 THE PRACTICAL HI-Fl HANDBOOK
this saves considerable time with regard to the wiring of the installation, as
compared with the wiring of a large number of single-source units to provide
the same coverage of sound.
Fig. 5.24 shows two Pamphonic line-source units mounted in the rear of
the dome of Rhodes House (Oxford) during the course of the Duke of Edin-
burgh's Study Conference held there in 1956. Fig. 5.25 shows the neat instal-
lation of an outdoor unit, mounted on a specially-designed pole and stand.

GEC PERIPHONIC LOUDSPEAKER SYSTEM

The periphonic loudspeaker system evolved by the General Electric
Co. is based on the GEC metal-cone speaker units in conjunction with a
system of GEC presence units and a cabinet specially designed to eliminate
all structural resonances, whilst also exploiting the enhanced bass response
of the periphonic principle. In effect, there are two metal-cone units mounted
in a V-shaped enclosure, the units being powered in such a way that the cones
are driven in push-pull (from the mechanical aspect). This results in cancella-
tion of any residual harmonic distortions produced in the cones themselves,
and reduces the total second- and third-harmonic distortion to some 2·7 per
cent at 40 c/s. This is remarkable when it is compared with the 40-per-cent
distortion of this nature produced in a paper cone unit at the same frequency
and power.
Four presence units are installed at the front and sides of the cabinet,

~I
which can be switched if required so as to
alter the apparent nearness of the orchestra to
individual listening requirements and conditions.
The presence units are fed through cross-over
networks.
-=-
SOUND DISTRIBUTION
Although up to the frequencies of about 1,000
c/s a loudspeaker functions as a spherical radiator
and has a radiation angle of almost 180 deg., at
high frequencies the whole area of the cone or dia-
phragm is unable to serve as a "piston" on the
surrounding air. Radiation of the sound thus be-
comes confined to a narrow beam, whose diameter
reduces as the frequency is raised. This is because
the area of the cone responsible for the high-

FIG. 5.25. Outdoor version of Pamphonic line-source

loudspeaker unit.
122
 LOUDSPEAKERS AND ENCLOSURES
frequency radiation progressively diminishes towards the speech coil with
increase in frequency.
Various devices are available for diffusing the high frequencies and
thus preventing this beaming effect. A simple, though effective, method
takes the form of directing the sound from the treble unit into the corner
of the room, whence it is scattered into the listening area. It will be recalled
that this idea is adopted in the case of the Pamphonic Victor loudspeaker
system.
Another method which appears to be gaining popularity is the use of
reflectors of various shapes positioned in front of the cone of the treble unit.
The Burne-Jones tweeter unit incorporates this idea. The tweeter unit proper
is housed in an attractively finished wood cylinder having three wide-spaced
feet so that it can be unobtrusively positioned on top of the normal enclosure.
The sound is emitted from the top of the cylinder by way of a horn-loading
system of very small dimensions, and full omni-directional radiation in the
horizontal plane is secured by a cone-shaped reflector, styled in the form of a
mushroom, mounted on top of the cylinder. The unit carries a cross-over
network and balance control, and has a frequency response from 2,000 c/s
to 18,000 c/s.
Apart from a cone, other shapes can diffuse the high frequencies to any
particular pattern, though it should be remembered that simple geometric
reflection of sound occurs only when its wavelength is small compared with
the dimensions of the reflector. The Lowther Type PM6 high-frequency unit
has a diffuser actually installed at the centre of the cone.
Another method of evoking dispersion of the high-frequency beam is by
means of a so-called acoustic lens mounted in front of the treble or middle-
range unit. Theoretically, an acoustical lens serves to bend sound waves just
as an optical lens bends light rays and a radio lens bends radio waves. Since
we have a narrow sound beam to start with, however, and it is required to
radiate this over a greater angle, the acoustical lens is arranged to be of the
divergent type.
The lens is formed of an array of slant plates through which the sound
has to pass. The distance between the plates and their angle of slant determine
the frequency of operation of the lens and its comparable refractive index.
Since the slant has the effect of extending the path of the incident wave in
relation to the normal path of the wave, the waves undergo an effective
change in velocity on passing through the lens, and on leaving thus tend to
either converge or diverge from normal, depending upon how the lens is
designed.
The high-frequency limit of the device is governed by the spacing of the
plates in terms of half a wavelength, while the refractive index is given by the
reciprocal of the cosine of the angle of slant.
123
 THE PRACTICAL HI-FI HANDBOOK
Since it is necessary for the lens to receive as near an approximation of
a plane wave as possible for correct operation, acoustical lenses are usually
associated with horn-loading, which satisfies this condition. The Westrex
high-frequency horn-loaded unit "Acoustilens" very successfully combines
horn-loading and the acoustical lens principle.

ADJUSTMENTS TO SPEAKER SYSTEMS

Properly used, it is very rare these days for the loudspeaker unit to
require detailed attention. However, should a definite fault develop, it is
often best to return hi-fl units to the maker for reconditioning. This procedure
is not usually necessary with the less expensive types, and with those units
falling outside the accepted hi-fl definition.
Complete failure is an almost certain indication of an open speech-coil
circuit. Before getting the cone and speech-coil assembly replaced, however,
careful attention should be given to the flexible leads connecting the speech
coil to the terminal block or tags for, apart from a definite burn-out caused
by a severe overload, these are the most vulnerable trouble points. If the
speech coil is definitely open-circuit, as can be determined by making a
continuity check at the points of connexion of the flexible leads on the cone,
either a replacement can be obtained from the manufacturer, complete with
cone, and fitted in the workshop, or the unit can be sent to the maker or a
firm specializing in speaker repairs.
Probably the most common of all speaker troubles is an out-of-centre
speech coil, resulting in its fouling the pole pieces of the magnet. This trouble
can easily be established by grasping the cone at diametrically opposite
points and gently moving it in and out. If a scraping noise is heard when this
action is performed, the centring screw or screws should be released a half
a turn or so and the cone manipulated until the speech coil moves freely in
and out of the gap without any scraping or rubbing. If necessary, feeler
gauges can be inserted, and the centring screws carefully tightened without
imposing too much pressure on the spider or centring disk.
Extreme caution must be taken to avoid particles of metal being attracted
to the pole pieces. Speaker units are designed to prevent this happening, but
if it has been necessary to break the dust seal to recentre the speech coil and
particles of metal have been let in, the cone will almost certainly have to be
taken from the chassis in order to clear the gap. A thin slip of modelling clay
is useful for this purpose.
Excessive buzzing should lead to examination of the fixing of the cone
to the chassis and the speech coil to the cone. A good-quality cement should
be used to re-fasten these items, if necessary. A damaged cone must be
replaced, even if only a temporary repair is made with adhesive plastic tape.
Similar trouble may be caused by odd resonances of the enclosure or baffle
124
 LOUDSPEAKERS AND ENCLOSURES
A check over the audible range with a variable-frequency oscillator soon
reveals trouble of this nature, and steps can then be taken to avoid the
resonance.
The inside walls of enclosures can have their natural resonances broken
by glueing across their width stout pieces of wood at odd intervals. When this
is done it is as well to lag the inside of the enclosure with acoustical damping
material so that it actually covers the wood struts.
If the bass performance appears to be lacking when an amplifier known
to be in good order is used with a certain loudspeaker system, there is a
possibility that the enclosure is not providing optimum match to the speaker
unit. If a bass reflex-type enclosure is employed, a measurement of the
speaker's impedance curve will prove or disprove this.
If the upper resonance gives a response of greater amplitude than the
lower resonance (see Fig. 5.16), the trouble may be caused by the vent being
too small, thus evoking resonance of the enclosure at too low a frequency.
This often results if a speaker unit having a relatively high main resonance is
used in an enclosure which is designed for a speaker with a very low main
resonance. Improved results can be secured by increasing the vent area, but it
is better to use a speaker whose cone resonance matches the resonance of the
enclosure.
If the balance is disturbed in the opposite way, and the response of the
lower resonance is of greater amplitude than the upper, the trouble is likely
to be caused by either too small an enclosure or too large a vent. In this case
the area of the vent can easily be reduced until tests indicate a reasonable
match. If the enclosure is much too small, however, the required reduction in
area of the vent may promote too great a drop in bass output. This can be
overcome either by extending the vent aperture inwards by means of a duct
(Fig. 5.26a) or by inserting an acoustically lagged partition between the
speaker and vent (Fig. 5.26b). The
partition should extend about three-
LOUDSPEAKER LOUDSPEAKER
quarters of the way into the cabinet
and should very snugly join the front \~
=
~
c:>
and sides. When either of these proce-
dures is adopted, the original vent area
PARTITION+
can remain, and in consequence there
is considerably less loss in bass input.
DUCT

VENT VENT

FIG. 5.26. Methods of reducing the en-

closure resonance; (a) by means of a duct:
(b) by fitting a partition. (al (bl

125
 THE PRACTICAL HI-Fl HANDBOOK
SPEAKER PLACING

The performance of a speaker system can be greatly affected by its

position in the listening room, while the room furnishings are also a governing
factor on how the reproduction "sounds" to a listener. A room without
heavy furnishings, such as armchairs and carpets, is prone to be very "live";
little sound is absorbed and reflections occur between the walls and hard
objects. There is a rising top response and a possibility of "ringing" at
certain frequencies. Careful adjustments to the tone controls can often effect
a compromise under these conditions, but such adjustments should be made
in relation to the placing of the loudspeaker.
Standing waves, which give rise to undesirable acoustics, can be modified
in amplitude and frequency simply by moving the speaker system from one
point to another. In a room which is subject to standing waves it is often
desirable to situate the speaker in one corner, as distinct from having it in the
centre of a wall. However, the best position can be found only on a trial-and-
error basis, and it is almost useless to try to formulate any hard-and-fast
rules about this. The speaker should be situated where it sounds best and not
where it looks best, but, unfortunately, this is not always acceptable to the
lady of the house!

126
 CHAPTER 6

Disk Recording

DESPITE many recent developments in magnetic tape record-

ing, the ordinary gramophone record remains at present the most popular
medium for the storing of musical programme material. Whilst there are
available a few instruments designed essentially for the playing of tape records,
as distinct from tape recorders, their number falls very far short of the
millions of disk-record reproducers of various types that are in current use.
The advent of the stereophonic technique has not altered this general
trend, as was expected in some quarters when stereophonic tape records first
made their appearance, and such records now have to compete with stereo-
phonic disk records. Although magnetic tape is extensively used as a hi-fi
programme medium, it is principally associated with tape recorders, with
which a private library of musical works can be built up by way of a V.H.F.
radio channel.
To facilitate editing, save expense and avoid undue retakes, disk records
usually begin as tape records. After a work has been recorded, and both the
artist and technician are satisfied with regard to their own particular interests
in the material, it is transferred from the tape to a wax or lacquer disk. From
this original is made a master disk by a process of copper plating. The ridge
of the master represents the modulated groove of the original, so the master
can be used as a direct working matrix for producing the final pressings.
Usually, however-particularly where a large numberof pressings is required-
there are two more stages in the process, resulting in two more intermediate
disks, a positive "mother" and a negative "stamper". In this case the stamper
is used as the actual working matrix.
Most commercial 78 r.p.m. records are produced basically of shellac,
while their microgroove counterparts of the so-called unbreakable type are
made of a soft plastic called vinylite.
The basic theory of disk recording has already been briefly discussed at
the end of Chapter 1, and enlargement on the subject is hardly warranted in
a book of this nature. However, there are one or two rather important points
about the care of records which should be mentioned.
127
 THE PRACTICAL HI-Fl HANDBOOK
Records are rapidly ruined by the use of an incorrect stylus, a too-heavy
pick-up, by dust and binding of the tone arm on its bearing, and, of course,
by rough handling. Service technicians should bear these points in mind when
using a customer's records for test purposes. Certain hi-fl enthusiasts lavish
as much care on their records as does a mother on her new baby, and the
technician should always handle records with care.
Microgroove records require a stylus of dimensions between 0·0008 in.
and 0·001 in. (one thou' or 25 microns), while a tip radius of between
0·0025 in. and 0·003 in. is satisfactory for 78 r.p.m. records. If a microgroove
stylus is used on a 78 r.p.m. record the tip will skate along the base of the
groove and give rise to a very high noise level, besides ruining the stylus if
not the record. A 78 r.p.m. stylus used on a microgroove record will have
great difficulty in remaining in the groove; it may intermittently ride on the
"horns" between the junction of the top of the groove wall and the "land"
between the groove. There will be a considerable loss of high-frequency
response. Indeed, if persistent use is made of the incorrect stylus, this response
will disappear from the record permanently.
Dust is one of the worst enemies of disk records. Simply wiping the record
with a dry cloth does little to alleviate the problem, in fact it aggravates it by
imparting a high static charge to the disk which makes it act as a magnet for
dust particles. There are several devices on the market for the elimination of
dust, and fluids with which the records can be polished whilst also being
rendered temporarily anti-static. G. A. Briggs recommends a detergent such
as Stergene or Quix, diluted by adding 95 per cent distilled water. He states
that a medicine bottle of the liquid can be filled at a cost of less than sixpence,
and that in the dustiest of districts this should last a year.
Unless the tone arm is perfectly free to rotate in its bearing the groove
wall will be subjected to extreme pressure, the impressed modulation pattern
will become distorted and in bad cases the "land" between the groove may
collapse. There is usually an adjustment on the tone arm in relation to the
bearing which can be set to provide adequate freedom of movement whilst
avoiding undue up-and-down play. A spot of very light machine oil on the
bearing often helps.
The downward pressure of the stylus on the record is much of a com-
promise and depends upon the type of pick-up and record: 78 r.p.m. records
usually require a greater downward pressure than microgroove records in
order to hold the stylus in the groove. This applies particularly to older
pick-ups. The downward pressure of modern pick-ups may be as low as
4 grams. This has been brought about by progressive attention to such points
as the total compliance and total effective mass in relation to the stylus tip.
Whilst a reduction in the downward pressure, or playing weight as it is
sometimes called, will increase the life of the record and stylus, it is never a
128
 DISK RECORDING
FIG. 6.1. Diagram
of constant-velocity
recording for two
frequencies. ---- /+-----------,

good idea to re-

duce the playing
weight below that t
stipulated by the
---- -- ---------1
AMP2LITUOE
___ j_
I
AM~LITUOE

maker as this may

----------
result in "groove jumping", which may be more detrimental to the record
and stylus than a gram or two of extra weight.

RECORDING CHARACTERISTICS
The recording head transduces the modulation pattern of the a.f. signal
at the output of an amplifier into lateral movement of the recording stylus.
In this way the pattern corresponding to the modulation is impressed upon
the groove as it is being cut during the recording process. The actual lateral
oscillations (low-frequency) of the recording stylus can readily be felt by
lightly placing a finger on the stylus while the amplifier is receiving a pro-
gramme signal.
The two basic factors associated with the lateral oscillation of the stylus
are amplitude and velocity. If the velocity of the stylus is to be maintained
constant over the whole of the audible frequency range, as is usually required,
it is clear that the amplitude of the stylus will increase with decrease in
frequency. At high frequencies the amplitude will be very small and at low
frequencies it will be very large. This simple rule can be expressed in terms of
velocity as 21r/A, where/is the frequency inc/sand A is the peak amplitude.
Thus, in order to maintain the velocity at a constant value, A increases as
f reduces.
The idea is not clearly understood by all service technicians, and since it
is rather important in hi-fl work, Fig. 6.1 is given to illustrate it better. Waves
A and B represent the modulation pattern imparted upon the groove of a
record owing to the lateral oscillation of the stylus. The velocity of the stylus
is represented by the slopes SI and S2 of the waves. Wave B has twice the
frequency of wave A, but in order for the slopes to remain equal (representing
constant velocity) wave Bis half the amplitude of wave A. For a constant
stylus velocity, this means that the amplitude will increase by 2: I for every
2:1 decrease in frequency; or, expressed more technically, the amplitude will
decrease 6 db per octave.
Fig. 6.2 shows the effect for constant amplitude. As before, wave B is
129
 THE PRACTICAL HI-FI HANDBOOK
151 FIG. 6.2. Diagram of
I constant-amplitude
I
I recording for two

---------r---i- frequencies.

!~-------+---+-------l~
<
t
j
AMPLITUDE

j
AM~ITuDE twice the frequency
of wave A, but since
the amplitudes are
- ---- - - - - -- constant and equal,
the slope (S2) of wave B is steeper than the slope (SI) of wave A-
indicating that the velocity of the stylus required to impart the higher-
frequency wave is greater than that required to impart the lower-frequency
wave. For constant amplitude, the velocity, in fact, increases at the rate of
6 db per octave.
Since the amplitude of the recording stylus would be excessive at low
frequencies, and result in break-through from one groove to the next, if the
constant-velocity principle were applied to disk recording, the constant-
amplitude idea is adopted for the low frequencies and the constant-velocity
idea for the higher frequencies. This works well in practice, since constant-
amplitude recording would never do for the high frequencies owing to
distortion which would result from the excessive velocity of the stylus tip.
Pick-ups would never track and the modulation pattern would soon collapse,
even if it were possible to impress it at high velocity on the groove.
The point at which the constant-amplitude recording changes over to
constant-velocity recording is known as the "turnover" or "crossover". The
response of the recording head is usually equalized in such a way as to provide
constant-amplitude recording up to the crossover point, which is positioned
somewhere in the region of 500 c/s. The change-over from constant amplitude
to constant velocity is not "sharp", but occurs gradually as governed by the
equalizing network. A representative curve is given in Fig. 6.3, which is
typical of a 78 r.p.m. recording characteristic.
If a sliding-frequency record is cut to this characteristic, and is played
back with a moving-coil or moving-iron type pick-up, whose output voltage
is equal to the velocity, the output voltage from the pick-up will follow the
curve very closely. Thus, in order to maintain a constant output at the lower
frequencies, the pick-up circuit will have to be equalized to provide a bass
lift, and the equalization curve must be the inverse of that at Fig. 6.3. A
crystal-type pick-up will not require the same degree of equalization, since
its output is proportional to displacement and not to velocity, but more will
be said about this later.
130
 DISK RECORDING
FIG. 6.3. Typical re-
cording characteristic
of 78-r.p.m. record.
t +db
CONSTANT VELOCITY

....
~ Od
....::>
O ·db

The recorded
level is usually given
in terms of velocity FREQUENCY -
(r.m.s.) at 1,000 c/s
or in decibels relative to I cm/sec (zero db). Maximum recorded level may
lie between 15 db and 26 db, depending upon the type of record and recording
characteristics. In accordance with this practice, the output voltages of pick-
ups are given in terms of mV per cm/sec.
Since the microgroove record is recorded at a lower level than the
78 r.p.m. record, the noise actually generated through the playback stylus
tracking in the groove assumes proportions approaching the modulation
level of soft passages of music. This disconcerting background noise (usually
referred to as "record hiss") is reduced by the vinyl-base material of the
record itself, and additionally by the application of a progressive boost to the
higher frequencies of the recording signal.
This recorded emphasis of the higher frequencies is of no consequence
from the "quality" aspect, since on playback a de-emphasis network can be
used to linearize the response. However, it has some bearing on record hiss
because the noise frequencies which are most troublesome are also reduced
considerably in level by the de-emphasis or equalizing network. A similar
idea is adopted in f.m. receivers and adaptors (see the author's "F.M.
Radio Servicing Handbook").
This noise-reducing arrangement has led to a large number of recording
characteristics, each requiring its own particular de-emphasis curve to
provide the correct degree of equalizing. Modern hi-fi control units have
three or four record-equalizing positions on the selector switch, and whilst
these do not cater completely for the many recording characteristics, they do
permit a fairly close compromise between characteristics which have much
in common. Slight deviations can be compensated for by the use of the tone
controls.
A formidable array of recording characteristics which have been adopted
over the years by leading gramophone record manufacturers is given in Fig.
6.4. Whilst these curves are not necessarily identical with those published by
the record manufacturers, they do represent close approximations which,
when correctly equalized, have been found to give the best quality on a
perfectly linear amplifier. It will be seen that, apart from the E.M.I. 78 r.p.m
131
+10
f--

EARLY "'ffrr"'
-s
-10

-IS--

-20
f--
=
-2S
DECCA 71 r.p.111."ffrr"
+S _ _

==- f--

-10 _.......:_·

-1~

-20
----~-
---
__ c --t
~-1-
E.H.L 71 r.p.m

APPROXIHA TE AMERICAN STANOAl'.0 & MICROGROOVE

DECCA HICA.OGROOVE

E.M.1. MICROGROOVE
Frequency in Cycles p•r second

132
 DISK RECORDING
characteristic, and the early Decca "ffrr", the curves indicate the use of
varying degrees of high-frequency emphasis; this being made possible in
later years by improvements in the design of pick-ups. The American curve
is one of almost constant amplitude, apart from the slight fall-off in slope in
the region of 1,000 c/s.
Fortunately, there now appears to be a move to standardize recording
characteristics throughout the world. In 1955 a curve sponsored by the Radio
Industries Association of America (R.I.A.A.) was accepted by the major
recording companies, and in this country was embodied in British Standard
1928:1955. This curve is given in Fig. 6.5 (see also Fig. 2.6 for the equaliza-
tion curve). It is to be hoped that there will be an early move to standardize
stereophonic records before the situation there gets out of hand.

DISK RECORDING PROBLEMS

There are one or two further points with regard to disk recording which
concern the service technician. As the recording stylus cuts a modulated
groove, the speed of the record at the point of contact with the stylus decreases
linearly towards the inner diameter of the disk. With a 78 r.p.m. 12-inch disk,
the speed at which the groove is cut falls progressively from about 47 in. per
second to 17 in. per second. This means that towards the inner diameter the
modulation waves are cut at a far steeper angle than are the waves of the same
frequency towards the outer diameter.
Owing to this, there is a progressive falling-off in the high-frequency
response towards the centre of the record. With a 78 r.p.m. 12-inch record,
the actual wavelength of the impressed modulation at 10,000 c/s drops from
about 0·0047 in. to 0·0017 in. from the outer to the inner diameters of the

I
I
18
- - 78 R.P.M. _,,
.,
12
- - - • MICROGROOVE
,,,, ./
db ,, _,,, /
_,, ~....-
'
--- .-
0
_,
--- -"'-,,,
--
__,
-12

-18
--- ----
--- --
40 80 160 320 640 1280 2560 5120 10 240 20480
FREQUENCY (c/s)

(Opposite page) Fm. 6.4. Recording characteristics approximating those used by

leading record manufacturers. (Above) Fm. 6.5. Recording characteristic equal to
B.S. 1928: 1955.
133
 THE PRACTICAL HI-FI HANDBOOK
record. There is little that can be done on playback to alleviate this falling
response, apart from ensuring that the stylus is in good condition. The
smaller diameter of the tip of a microgroove stylus greatly reduces this
trouble on the overall slower speed of microgroove records, but reducing the
diameter of the tip on standard 78 r.p.m. records would not likewise enhance
the response.
During the recording process the higher frequencies are sometimes given
a boost, in addition to normal pre-emphasis, which increases progressively
as the recording stylus approaches the innermost groove. This idea, known
as "radius compensation", is used by home recordists as well as by certain
commercial recording organizations.
An optical method, devised by Buchmann and Meyer, of determining
the level of recording on a disk over the whole of the audio spectrum is shown
in Fig. 6.6. Various spot frequencies spaced over the spectrum are recorded
at intervals throughout the record, and by manipulating the record in relation
to a light source the bands of modulation pattern can be clearly observed.
Since the breadth of the bands corresponds to the velocity of the recordings,
such a record produced from a constant-input signal will readily reveal the
response deficiencies of the recording amplifier and head. The record is also of
value for checking the frequency response of pick-ups. The photograph of
the Buchmann and Meyer pattern shows that the recording response is
maintained reasonably constant at the higher frequencies, and also shows the
constant amplitude (falling velocity) characteristic at the lower frequencies,
towards the inner diameter of the record.

FIG. 6.6. Showing how a source of light reflected from a recording of spot frequencies
over the audio spectrum indicates the recorded level in terms of breadth of the bands.
(The Buchmann and Meyer test.)
l34
 DISK RECORDING

FIG. 6.7. Tracing distortion is caused because the line (full lines on diagram) between
the points of contact of the reproducing stylus progressively deviates from the line
(dotted lines on diagram) between the points of contact made by the recording stylus
on the sloping parts of the wave. The lines coincide only at the peaks of the wave and
when the groove is unmodulated. This results from the fact that a spherical-tipped
stylus is used to reproduce a waveform imparted by a chisel-edged recording stylus.
The pinch effect is the narrowing of the groove on the sloping parts of the wave,
causing a vertical movement at twice the recorded frequency of the reproducing stylus.

In order to reduce the high-frequency loss during the recording process

as the result of the increasing impedance offered to the recording stylus, the
"hot stylus" technique is sometimes adopted. A few turns of resistance wire
are wound around the recording stylus, and a controlled current is passed
through the coil, thus raising the temperature of the stylus sufficiently to
soften the cellulose lacquer. This reduces the impedance presented to the
oscillating stylus whilst recording, and not only enhances the "top" response,
but also reduces surface noise on playback.

TRACING DISTORTION AND THE PINCH EFFECT

Inherent in disk recording are other distortions which, although not of
direct concern to the hi-fi service technician and enthusiast, at least warrant
consideration. Both tracing distortion-not to be confused with "tracking
distortion", which is a direct function of replay-and the pinch effect arise
from the fact that the reproducing stylus has a spherical tip while the record-
ing stylus has a chisel edge. The facets of the recording stylus, which is usually
of sapphire, are set to produce a low-noise polished groove, of depth between
0·0015 and 0·0025 in.
Fig. 6.7 shows a waveform of modulation which may be impressed upon
a groove of a record. The cut is in the direction of arrow A, while the ampli-
tude characteristics are imparted by reason of the lateral movement of the
stylus as at B. The cutting face of the stylus is always maintained in the same
plane, at right angles to the direction of the cut, irrespective of the modulation
pattern. This is shown by the dotted lines across the groove. On playback,
however, the reproducing stylus maintains contact at the sides of the groove
at points directly opposite, thus giving the full lines shown on the diagram
which are always at right angles to the direction of the groove.
135
 THE PRACTICAL HI-FI HANDBOOK
It will thus be seen that the points of contact of the recording stylus on
the sides of the groove coincide with those of the reproducing stylus only at
the peaks of the wave and when the groove is unmodulated. On the sloping
parts of the wave, the line between the points of contact of the reproducing
stylus progressively deviates from the line between the points of contact
made by the recording stylus. Now, since the lateral movement of the record-
ing stylus was in the direction of the dotted lines and the output signal of the
pick-up is promoted by the movement of the reproducing stylus in the direc-
tion of the full lines. it will be apparent that the reproduced waveform differs
from the waveform of the signal applied to the recording head.
Briefly, harmonic and intermodulation distortions are introduced by
tracing distortion because a spherical-tipped stylus is used to reproduce a
waveform imparted by a chisel-edged recording stylus, which results in the
curve traced by the centre of the tip of the reproducing stylus not being an
exact replica of the modulated groove. It will be evident that this kind of
distortion increases with amplitude and frequency of the recorded wave,
and also towards the inner diameter of the disk, since in all these cases there
will be an increase in the slope of the waveforms recorded.
The "pinch effect" is also caused by the difference between the shapes
of the recording and reproducing styli. As is shown in Fig. 6.7, the groove is
equal to the full width of the recording stylus only when it is unmodulated
and at the peaks of the modulation. At other times during modulation, the
cutting edge is at an angle to the direction of the groove, and at these times
the groove width decreases. Modulation of a simple sine waveform will,
therefore, cause the reproducing stylus to oscillate vertically at twice the
frequency of the modulation. This is because the reproducing point is truly
spherical and thus rides up the groove when it narrows. The same effect occurs
in a much more complex way when ordinary sounds and music are recorded.
Whilst a perfect pick-up would not give an output from a vertical move-
ment, there is a small unwanted signal generated from this cause in most
pick-ups, and because of its method of generation it is usually rich in second
harmonics. Modem pick-ups which adopt the cantilever-stylus principle are
Jess prone to this trouble than their older counterparts.
Owing to the reduced width of the groove of microgroove records and
the reduced diameter of the tip of the reproducing stylus, these records show
a distinct improvement in terms of distortion over the standard 78 r.p.m.
record. Nevertheless, as the innermost grooves are approached the distortion
rises rapidly, and at 5,000 c/s it may be as high as 20 per cent at a recorded
diameter of 4 in. This is where the microgroove record scores again, since the
minimum recorded diameter is 4J in., compared with the 3¾ in. for standard
78 r.p.m. records. Home recordists should never be tempted to cut too far
towards the centre of a disk if the quality of reproduction is to be maintained.
136
 CHAPTER 7

Pick-ups and Record Playing

Equipment

THE duty of the pick-up is to change the lateral vibrations of

the reproducing stylus, as it traces the modulated groove, into an electrical
equivalent of the modulation pattern which was imparted during the recording
process. It is the aim of the designer of pick-ups to produce a unit whose
output signal is an exact replica of the signal which was applied to the
recording head when the record was made. Since there are so many mechanical
factors involved between the change-over from the electrical to the mechanical
at the recording end and the change-over from the mechanical to the electrical
at the reproducing end, the designer is presented with a great number of
complex problems which, from the hi-fl aspect, are aggravated by the extra
pre-emphasis and consequent increased velocity of the reproducing stylus.
Pick-ups can be considered as small generators of electricity. There are
two basic types, those whose output voltage is proportional to the velocity of
the movement of the stylus, and those whose output is proportional to the
force applied to the stylus. The former adopt the electromagnetic principle
for the generation of their output voltage and may be likened to the bicycle
dynamo whose output
DAMPED
voltage is governed by the
COIL PIVOT speed (velocity) of the
driving wheel. The latter

~(
C
type rely upon the property
of Rochelle salt crystals of
producing a potential
difference when subjected
------- to changes of pressure.

FIG. 7.1. Moving-coil pick-up unit.

137
 THE PRACTICAL HI-FI HANDBOOK
RUBBER TORSION FIG. 7.2. Moving-armature
~ BLOCKS ~ pick-up unit.

MAGNET
N Electromagnetic pick-
ups employ a coil or
conductor together with a

-
small permanent magnet;
STYLUS
a voltage is set up in the
coil or conductor as a
result of the variations of magnetic flux caused by the side-to-side movement
of the stylus. The stylus is coupled to one or other of the two elements either
directly or magnetically.
With the moving-coil pick-up (Fig. 7.1), the coil is directly coupled to
the stylus, and the same applies to the ribbon pick-up, which is really a
moving-coil unit having a single-tum coil. With the moving-armature pick-up
(Fig. 7.2), the stylus is coupled directly to the armature which vibrates in the
gap of a magnet supplying the steady field, and the coil is wound over the
magnetic circuit. There are several variants of this type of unit, from the older
moving-iron and needle armature units to the tiny micro-armature types and
the so-called variable-reluctance pick-ups, the latter probably being the most
popular of the electromagnetic range.
Whilst all electromagnetic pick-ups, with the exception of the moving-
coil unit, are essentially variable-reluctance types, it is the pick-up illustrated
in Fig. 7.3 which is usually known by this designation. Its operation is quite
straightforward; there are two coils wound over the magnetic circuit, which
is provided by the small cylindrical permanent magnet, the pole pieces and the
ferrous (magnetic material) stylus arm. The stylus arm thus vibrates within
the gap between the pole
pieces, which causes the flux FERROUS STYLUS
ARM
alternately to increase in one
( STYLUS
pole and decrease in the other,
thus resulting in increasing I
and decreasing e.m.f's in the
associated coils, which are
phased so that the voltages
are series-aiding.

FIG. 7.3. Variable-reluctance pick-up umr.

138
 PICK-UPS AND RECORD PLAYING EQUIPMENT
COILS Fm. 7.4. Moving-magnet pick-up
unit.

STYLUS

.
G:J1
. . . .---:.-:..-::.-::.-:.'J

It is well known that a voltage

CANTILEVER will be generated in a coil if a varying
magnetic flux cuts the turns either by
the coil moving in the flux or by the
movement of a part of the magnetic circuit. This follows the basic law of
electromagnetic induction which was first postulated by Faraday.
Exactly the same effect will be achieved, and a voltage will be developed
in an associated coil, if the magnet moves while the other element remains
stationary. This principle, which is used extensively in some bicycle dynamos,
is adopted in the Philips Novosonic pick-up. The magnet consists of a thin
ferrite rod which is magnetized across its diameter, and to which is coupled a
cantilever stylus. The magnet thus undergoes a slight twisting movement, due
to the lateral movement of the stylus, within the gap of the mumetal yoke.
As with the variable-reluctance unit, the flux will alternately increase in one
side and decrease in the other, resulting in increasing and decreasing e.m.f's
in the associated coils. The idea is shown in Fig. 7.4.
The Rochelle salt crystal used in crystal pick-ups is cut from a big
crystal at a critical angle and placed between two metallic plates so arranged
that the pressure on the crystal is varied in sympathy with the vibrations of
the stylus. The crystal element is very fragile and easily broken or damaged
by excessive mechanical pressure on the stylus and by high temperature.
Extreme care should, therefore, be taken when replacing the stylus in one
of these units. Instead of Rochelle salt crystal, poly-crystalline barium
titanate has been used in some recent units. This material is less affected by
temperature and humidity, but unfortunately does not lend to the extended
frequency range of the Rochelle salt crystal desirable in pick-ups.

PICK-UP MECHA1'ilCS

Modern recording technique, resulting in extended high-frequency

response, pre-emphasis as a noise-reducing agent, and the long-play
139
 THE PRACTICAL HI-FI HANDBOOK
microgrooverecord,has introduced records with very sharp changes in modu-
lation pattern. Thus, in order to reproduce the modulation impressed upon the
grooves of these records with true fidelity and without causing damage to
either the pick-up stylus or record grooves, a pick-up whose stylus is able to
respond immediately to the rapid accelerations and decelerations is essential.
At high frequencies and recorded levels the accelerations encountered
by the stylus may approach some thousand gravities (l,OOOg). Acceleration
involves a variation of either velocity or direction, or both, and the pick-up
stylus is subjected to both of these factors when it is tracing the modulated
groove of a record. The stylus and associated linkage also possess mass
(which is the quantity of the matter contained in the items referred to). As the
result of the accelerations of the mass, the stylus is subjected to a force, as we
have seen. The magnitude of this force is compared with the force due to the
pull of gravity-this being denoted by the letter g. Thus, the force acting on a
stylus when it is under the control of a heavily-recorded sound of high fre-
quency may rise to some 1,000 times the pull of gravity-hence the term
1,000g given above.
In order to prevent the record groove from being torn to pieces, the
pick-up stylus must be free to move very easily and rapidly from side to side.
If the pick-up is light and the stylus stiff, the stylus will be unable to respond
to sudden large changes in modulation and it will ride up on the groove wall
and skate across the record. This is what happens when an old-type pick-up
is used on a modern record.
As a means of reducing the mass of the moving parts of a pick-up and
ensuring that the effective inertia at the high acceleration rates encountered on
modern records is kept as low as possible, very lightweight stylus mounting
materials are essential; micro-armatures are often made hollow for this
purpose. The modern pick-up rates the mass of the moving parts in milli-
grams, which is a considerable achievement when it is considered that only a
few years ago these parts were rated in ounces, and remembering also the
much greater stress imposed upon these parts due to the high recorded
velocities of modern records. In this respect, it is interesting to note that the
effective mass at the stylus tip of the Goldring variable-reluctance cartridge
Type 500 is only 3 ·5 milligrams!
Another factor affecting the free movement of the stylus is the damping
arrangement needed in the pick-up to return the stylus and armature to the
point of balance in the centre of its lateral movement. In the pre-war heavy
type of pick-up in which a needle chuck was employed the iron armature was
clamped in the centre of its movement by hard rubber pads, which invariably
perished after a short time. In those days, considerable force was required to
promote a lateral displacement of the needle. There was little trouble in
tracing the grooves of the old-type records though, since the vertical force on
140
 PICK-UPS AND RECORD PLAYING EQUIPMENT
the needle due to the great weight of the pick-up (two or three ounces) left
no alternative; the recorded accelerations were also far less than they are
today. Nevertheless, neither the needles nor the records lasted very long,
and after several playings by this "brute force" method the quality on play-
back deteriorated sadly. Today, pick-ups track admirably at a pressure of
seven to ten grams.
The tracking pressure required is related to the lateral stiffness of the
stylus, oflateral compliance as it is known, compliance being the reciprocal of
stiffness and rated in centimetre/dynes. Thus, if the compliance is low the
movement is stiff and a greater vertical force is required to hold the stylus in
the groove. The compliance of hi-fi pick-ups is rarely less than 3·5 x 10· 6
cm/dyne.
In addition to the lateral compliance, a vertical compliance is necessary
to combat the pinch effect which was described in the previous chapter. If the
pick-up lacked vertical compliance, the whole of the pick-up and the arm
would have to follow the vertical movements as the result of the pinch effect.
The inertia of such a large mass would cause the stylus to plough along the
groove at high modulation levels and also evoke groove jumping. In such
cases the vertical movement of the stylus may be as much as one-tenth of the
lateral movement, and for this reason the vertical compliance is generally set
in the region of 5 x 10· 7 cm/dyne, but it is not always given in the data supplied
by the makers.
The cantilever-type stylus, which is formed of a straight or trailing shank
on which the sapphire point is mounted, provides by virtue of its design a
degree of vertical compliance, which usually satisfies the above condition.
Pick-ups suffer from resonances just as much as other items of audio
equipment. The high-frequency resonance is a function of the mass of the
moving parts; the smaller the mass the higher the resonant frequency.
With the small mass demanded of hi-fi pick-ups the upper resonance usually
falls outside the audible range, around 17-18 kc/s. The low-frequency
resonance, on the other hand, is usually governed by the tone arm and
characteristics of the pick-up head and mounting arrangements. An average
arm gives a low-frequency resonance in the region of 20 c/s, which is of little
moment since the response of the amplifier is pretty well limited at this
frequency to avoid motor rumble and low-frequency overloading troubles.
With regard to the pick-up proper, a high compliance in the lateral sense
damps low-frequency resonances and keeps them well down the low end of the
scale, out of harm's way.

STYLI
There are only two types of stylus used these days, namely, sapphire and
diamond. Pick-ups which use metal or fibre needles and needle chucks are
141
 THE PRACTICAL HI-FI HANDBOOK
useless for hi-fi work and should be discarded. There are dozens of different
kinds of styli for use in the many patterns and makes of pick-ups produced
over the last few years. It is impossible to go into detail, and in any case
there would be little purpose served in so doing. Pick-ups designed for
serious hi-fi work usually have available styli in sapphire or diamond. From
the actual playing point of view there is no difference between the two, but
from the aspect of longevity there is considerable difference.
The rate of wear of any stylus will be governed by a number of factors,
including tracking weight and vertical and lateral compliance of the pick-up,
the condition of the records, the amount of dust in their grooves and how the
equipment is handled. Generally speaking, the life of a diamond stylus is
some 20 times that of a sapphire, and under ideal conditions it is often
possible to get some 60 hours of playing time from a sapphire. The diamond
is much more expensive than the sapphire, however.
The great advantage of the diamond is that it avoids that gradual
deterioration in quality which is characteristic of a sapphire as it slides down
its life/efficiency curve. A point is reached where replacement becomes
necessary from the quality point of view, but not from the aspect of economy.
As a stylus nears the end of its useful life so the rate of wear of the records
increases. The quality of the reproduction also suffers considerably not only
by loss of the higher frequencies, but also by the introduction of harmonic and
intermodulation distortion as the result of the asymmetrical tracing motion
of the worn stylus.
Unless a microscope is available to assess the wear of the tip of the stylus,
it is good policy to replace the stylus regularly as governed by the type of
pick-up employed, and by the maker's recommendations in this respect.
Some dealers make use of the Philips "Needle Clinic" microscope, and such
an instrument is well worth acquiring if considerable hi-fi work is contem-
plated.

STYLUS REPLACEMENT
Extreme care should be observed during the operation of replacing a
worn stylus. It is often necessary first to remove the cartridge or head from the
pick-up arm. With the Acos SA and SB series, the stylus holder should be
lightly gripped with a pair of long-nose pliers or tweezers, while the worn
stylus is extracted from the holder with a second pair of pliers or tweezers
arranged to lever against the first pair. If undue pressure is applied to the
stylus holder, or if this item is not held rigidly without movement while the
stylus is being extracted, there is a strong possibility that the internal assembly
will be damaged, necessitating replacement of the cartridge as well as the
stylus.
Generally speaking, the cantilever-type of stylus can be withdrawn with
142
 PICK-UPS AND RECORD PLAYING EQUIPMENT
little difficulty in the way described above, and on replacing care should be
taken to ensure that the shank section of the new stylus is pushed well home
into the holder or "pocket".
The Collaro Studio crystal cartridges use a small machine screw to
secure the cantilever-type stylus to the body of the cartridge. Near the jewel
end of the cantilever the generating system is mechanically coupled by means
of a small pad resting on the cantilever arm. The styli for the types "O" and
"P" cartridges have phosphor-bronze shafts, while aluminium shafts are used
on the styli for the transcription cartridge, the long-play type having a series
of holes drilled along its length.
These various styli have an effect on the overall frequency response of the
pick-up, whilst also affecting the output voltage. They are, in effect, tuned to
fit in with the various responses of the pick-up itself. On no account should
the shape of the shank or cantilever be altered. If, after replacing a cantilever-
type stylus, it appears that the tracing angle of the jewel is incorrect, the
trouble will invariably be caused by incorrect fitment and not by distortion
of the shape.
Styli are usually colour-coded, red indicating the microgroove variety,
and green the 78 r.p.m. type. They should never be interchanged, since the
tip radius for microgroove is 0·001 in. and for 78 r.p.m. 0·0025 in.
The cantilever-type stylus, by virtue of its operation as two simultaneous
levers operating on the generating coupling, evokes four rather complex
resonances. Additional resonances are produced by the formation of the
cantilever section proper, as the result of twisting and torsional effects.
However, these are smoothed out to provide the required response of the
modern hi-fi unit.
There are a number of single pick-up units which can be used to play
either 78 r.p.m. or microgroove records, the cartridge having two styli, one
for 78 r.p.m. and the other for microgroove, mechanically coupled to either
a single generating system or independent generating systems. These are
usually called "turnover units" and operate quite successfully. Before this
idea was adopted it was necessary to change the head on changing from
78 r.p.m. records to microgroove records.
Another arrangement along these lines is the so-called "turnaround"
cartridge. Here the composite cantilever-type stylus is pivoted in its centre
so that it can be rotated through 180 deg. to bring the required stylus into
mechanical coupling with the common generating system. This is an admir-
able idea, since the requisite damping can easily be given to the styli to
maintain optimum results on both types of record.
One problem associated with pick-ups is "needle talk". This is the
acoustic rattle that vaguely resembles the modulation on the record which is
emitted from the mechanical function of the pick-up itself. This used to be
143
 THE PRACTICAL HI-Fl HANDBOOK
extremely troublesome in the days when heavy pick-ups were used. In fact,
the noise used to be so great that it was necessary to use acoustically-treated
lids on radiograms to prevent the noise affecting the reproduction from the
loudspeaker. (Hence the notice "Please close the lid when playing".)
The actual 5ound radiated from the pick-up and the record (the record
contributes towards it) was caused by the rise and fall of the complete pick-up
assembly and tone arm due to the pinch effect; the sound was, in fact, second
harmonic of that recorded. The trouble these days has been eased by the
smaller masses and dimensions of pick-ups and associated parts. Neverthe-
less, it is still present to some degree and is evoked by the mechanical back-
lash of the stylus on the record, as the stylus rises and falls at twice the
recorded frequency due to the pinch effect. The record in this case serves as
a sounding board.

OUTPUT FROM PICK-UPS

The output from pick-ups in terms of voltage is a function of their
sensitivity, and is measured in terms of r.m.s. volts for a certain stylus
velocity. As an example, the Goldring variable-reluctance cartridge Type 500
is said to have an average output of 3·2 mV per cm/sec. This means that for
each cm/sec of stylus velocity the output is 3·2 mV. The recorded velocity is
always changing with the programme material and pattern so the output
voltage changes accordingly. At a recorded level of 5 cm/sec the output from
the pick-up would thus be 5 times 3·2 mV, or 16 mV. This is still very low,
and requires a high degree of amplification.
Crystal pick-ups provide a greater output voltage and so require less pre-
amplification. Usually, however, the more hi-fl the pick-up the less its
output voltage. This should not be taken as a general rule, but as an approxi-
mate guide. It must also be remembered that if equalization is required, the
relating network itself absorbs some of the signal power from the pick-up,
thus reducing the mV per cm/sec across the output terminals.

SIMPLE EQUALIZING NETWORKS

Let us suppose that we have an amplifier with a perfectly linear response
within the workable limits of the audio spectrum, to which it is required to
connect a pick-up. The record, it will be remembered from the chapter on
disk recording, has a certain recording characteristic. In terms of velocity, the
bass is attenuated and the top frequencies accentuated. If we use a pick-up
whose output is proportional to the velocity of the recording (electro-
magnetic type), the reproduction will be most disappointing; there will be a
marked lack of bass and a marked increase in "top". This is because the
output voltage over the frequency range follows the recorded curve.
What is wanted is a network which lifts the bass and attenuates the
144
 PICK-UPS AND RECORD PLAYING EQUIPMENT
high frequencies to the extent that the bass was attenuated and the high
frequencies accentuated during the recording process. The equalizing
network should in fact have a response/frequency curve which is the exact
inverse of the recording curve. This will correct the output signal from the
pick-up and linearize it so that it suits the linear characteristic of the amplifier.
The reproduction will then sound much more "natural".
Although it is impossible to match inversely all the various recording
characteristics that have been adopted, a fair compromise is possible in
practice and final adjustment can be made with the amplifier's tone controls.
With this in mind, it would appear that future amplifiers will need to have
provision for only two recording characteristics-those approved by British
Standard 1928: 1955 (R.I.A.A.) which was referred to in the previous chapter.

EQUALIZING CIRCUITS
An equalizing circuit is frequency-selective in a sense which corrects the
recording characteristic, whilst also taking into consideration the pick-up and
amplifier loading and deficiencies of the response of the pick-up itself.
Assuming that the pick-up does not introduce its own response coloration
and that it has a truly constant velocity output, then the equalizing network
given in Fig. 7.5 can be used, either between the pick-up and the input of the
amplifier (or radio receiver) or after the first voltage-amplifier in the control
unit.
If we study the recording characteristic in Fig. 6.5, it becomes obvious
that the equalizing network has to perform three functions. It has to lift the
bass from about 600 c/s; to cut the lower bass at around 50 c/s; and to cut
the treble around 1,000 c/s. It has to do these things to secure an output
linear with frequency from the pick-up circuit.
So that these various functions can be understood, the circuit in Fig. 7.5
has been broken down into three basic sections which are given in Fig. 7.6.
The circuit at (a) serves to cut the lower bass, that at (b) provides an overall
bass boost, and that at (c) cuts the treble. When the circuit is in the composite
form given in Fig. 7.5, then it performs all these functions simultaneously.
The actual affect that these frequency-selective circuits have on the
overall recording characteristic depends upon the values given to the
capacitors C and resistors R. It will be remembered that a resistor and
capacitor associated in a circuit form a time-constant T whose value in

SIGNAL
FIG. 7.5. An equalizing network of this
RI SIGNAL
IN
R2 C2
OUT kind is often used for modern recordings in
conjunction with high-impedance electro-
magnetic pick-ups.
145
 THE PRACTICAL HI-Fl HANDBOOK
seconds is equal to the capacitance C in farads multiplied by the resistance R
in ohms. The time-constant is extensively used in equalizing networks, its
value being given in practice by the following expression: CR= time-
constant (T) = 108 /271'/, where C is in microfarads, R is in ohms, Tis in
microseconds and/is in c/s. The term/ is known as the turnover frequency,
and it is by reference to this that the curves are calculated so that they merge
inversely with the recording characteristic curve to produce a linear response.
The turnover frequency is reckoned to be that frequency at which the
response-as indicated by the curve-is 3 db below or 3 db above the refer-
ence response or datum line. With reference to Fig. 6.5: At about 320 c/s the
78 r.p.m. curve is 3 db below the O db datum line. Thus, to equalize, a bass
boost circuit (Fig. 7.6b) will have to be used whose time-constant suits this
frequency. Similarly, in the region of 3,000 c/s the same curve is 3 db above
the datum, indicating the necessity of a top-cut circuit (Fig. 7.6c) having a
time-constant related to this frequency. The lower bass requires cutting at
about 40 c/s, calling for the use of circuit Fig. 7.6a.
It will be obvious that the same answer for the time-constant T can be
secured from a host of C and R combinations. Generally speaking, however,
the value for the resistor RI (Fig. 7.5) is affected by considerations relating
to the matching of the pick-up to the amplifier input circuit. It is assumed that
the load presented across the output terminals of this kind of circuit will be of
high impedance, represented, for instance, by the control-grid circuit of a
voltage-amplifier valve. If the network is fed from a source of low impedance,
then a resistor should be placed in series with the signal whose value is high
in relation to Rl. Having fixed the value for Rl, in terms of matching, R2
should be arranged to be approximately 12½ times below RI.
The time-constant elements associated with Fig. 7.5 are Cl, RI and
C2, R2, whose time-constants respectively should be 2,940 microseconds and
81 ·2 microseconds for microgroove records, and 2,780 microseconds and
57·3 microseconds for 78 r.p.m. records.
All equalizing circuits of the nature described give a signal across their

r
C

I le I
LJJ·
C
IN R R
OUT IN OUT

(al (bl (cl

FIG. 7.6. Frequency-selective networks; (a) for lower bass cut: (b) for bass lift: and
(c) for top cut.

146
 FICK-UPS AND RECORD PLAYING EQUIPMENT
22K
,......-.......- - - - - - ' I N V 1 r - - - - - - - - - 0 H T+ 180 -250V

LOW-NOISE
PENTODE HT-
•-9· z 729 220K
EF86 ~

OUTPUT

INPUT 68K

RECOAOING EQUALIZER
2 I BRITISH 78 RPM RECOR.OS
...__ _... 0·008 27K 2 DECCA ffr, 78 RPM RECORDS
L ...... u -3 ,
.--•••~ 3 FINE GROOVE RECORDINGS \ BS I
~ , COARSE GROOVE RECORDINGSj 1928/Sl

Fm. 7.7. Single-stage amplifier recommended by Goldring for use with their Type
600 pick-up. The output is 80 millivolts (equalized) at 1,000 c/s at a recorded level of
3'16 cm/sec.

output terminals considerably below the level of the signal applied across their
input terminals, and thus attenuate the signal. In certain cases, particularly
where the output voltage from the pick-up is low to start with and the
amplifier is insufficiently sensitive, or where it is required to connect a low-
output equalized pick-up signal to the pick-up terminals of a radio receiver
or radiogram, a pre-amplifier will be required to make good this attenuation
and provide the amplifier with a signal of sufficiently high level to load it fully.
Two such pre-amplifiers recommended by the Goldring Manufacturing
Co. for use with their Type 600 variable-reluctance cartridge are shown in
Figs. 7. 7 and 7.8. Both circuits provide four degrees of equalization, and the
equalizing circuits in both cases follow the low-noise pentode valve. The
single-valve circuit provides an output of 80 millivolts at 1,000 c/s at a
recorded velocity of 3· 16 cm/sec (from the Decca test record LXT5346).
Without the amplifier and using the equalizing circuit recommended (see
Fig. 7.9 a and b), the output is only about 4 millivolts at a recorded level of
some 10-12 cm/sec. Thus, the single-valve circuit permits this excellent pick-up
to be used in conjunction with most makes of hi-fl amplifier.
The two-valve circuit, in which the output is by way of a cathode-
follower, is recommended for the Williamson amplifier or similar types
requiring a very low-impedance input. The Goldring 600 pick-up is intended
for operation into a resistive load of approximately 68,000 ohms, and is not
intended for use with a transformer.
As the output voltage over the frequency range is dependent not only
147
 THE PRACTICAL HI-FI HANDBOOK
IOK
. - - - - - - - - - - - - - - + _ , . , I V V l , - - - - 0 1 8 0 - 21i0 V.
+ S-7mA
22K
47K

ECC82
12AX7

INPUT

IM OUTPUT

RECORDING EQUALIZER
BRITISH 71 R PM RECORDS
DECCA ffrr 71 R PM RECORDS
FINE GROOVE 8S 1921/55 RI A.A
4 OLD LP RECORDS 8S 1921/55 COARSEGIIO<M

FIG. 7.8. A two-stage Go/tiring circuit. This is recommended for the Williamson
amplifier and similar units.

on the type of equalization network employed, but also on the frequency

response of the pick-up, the actual design of an equalizing network is not
always as simple as it may first appear. Fortunately, it is rarely necessary for
the enthusiast or technician to have to work out networks and component
values for specific pick-ups as these are nearly always given in the maker's
instructions.

CRYSTAL UNITS
The output voltage of a crystal unit is proportional to the force to which
the stylus tip is subjected when it is tracing a record. Thus, excluding resonance
effects, the open-circuit generated voltage is approximately linear with respect
to frequency with reference to the recorded amplitude. Since modem record-
ings have a characteristic approaching constant-amplitude rather than
constant-velocity (due essentially to the bass cut and treble lift), some
crystal units, such as the Collaro Studio "O", certain Acos units and others,
have an output/frequency curve which is almost the inverse of the recording
curve. The replay characteristic is thus automatically secured, and additional
equalizing networks are not required. In such cases, the pick-ups can be
connected direct to the input terminals of a linear amplifier and acceptable
reproduction is obtained.
When using crystal pick-ups of this kind care should be taken to see that
they are not connected to an equalized input of an amplifier, otherwise the
148
 PICK-UPS AND RECORD PLAYING EQUIPMENT
IIK Cl C2

a+• tc, f::+"""

WA
CURVE A O•Oll'F 0·0311F
CURVE 8 0·0051'F 0·1 l'F
A 2-POLE SWITCH MAY 8E USEO TO SELECT
(a) THE AL TEA.NATIVE VALUES OF Cl AND C2

,__
+20
~ ~I'-,
15
10
8 ....
- -
..........
~ ..........
5
-...... ~ ~
0
-....: ~

5
r--...._ r""i,.
10
15
~ ."a
'-•
20 IOO 1000 10000 20000
c/s
lb)
FIG. 7.9. (a) Equalizing network recommended for the Goldring Type 600 pick-up:
(b) associated equalization curves.

bass will be overpowering and the top considerably muted, as the result of
an effective double-equalizing function. The same trouble would result if a
record player whose pick-up signal passes by way of an internal equalizing
network is connected to an equalized pre-amplifier input socket. Some record
players. such as the Decca, employ internal equalizing so that they can be
connected direct to the pick-up sockets of a radio set. Hi-fi players, however,
rarely adopt this idea, the pick-up wires coming direct from the pick-up and,
unless the crystal types mentioned above are used, they require an equalizing
network either externally or in the amplifier. We have seen in the previous
chapters that switched equalizing circuits are usually incorporated in the pre-
amplifier or control unit of hi-fi systems.

150K
120K

100 1000 10000

FREQUENCY els

FIG. 7.10. The network recommended/or the Acos Black Shadow pick-up to provide
the substantially linear output shown by the curve for records cut to B.S.I./R.I.A.A.
standards.
149
 THE PRACTICAL HI-FI HANDBOOK
An essential element in the crystal pick-up is a capacitance formed by
the crystal as the dielectric and the plates either side. If this capacitance is
loaded by a resistance, then the signal is differentiated and the output will
approach the velocity characteristic. The extent of this modification in
characteristic depends to a large degree on the response of the pick-up unit
itself and the value of the load resistor. This makes it most important that
the value stipulated for the resistor by the makers is always used.
Fig. 7.10 shows the network recommended for the Acos Black Shadow
pick-up to give the substantially linear output from records cut to the
B.S.1./R.1.A.A. standards. The network in Fig. 7.11 should be used to modify
the output to the velocity characteristic so that the pick-up can be connected
direct to the magnetic input terminals of the control unit. The inclusion of
this simple resistor-capacitor network avoids the necessity of having to alter
the existing equalizing circuits to cater for the pick-up. Both networks work
into an amplifier impedance of 100,000 ohms, while the equalized signal at
the output of Fig. 7.10 is approximately 20 mV per cm/sec and 30 mV per
cm/sec at the output of Fig. 7.11, thus requiring amplifier sensitivities in the
region of 60 mV and 100 mV respectively.

MATCHING
Matching should not be looked upon lightly if best results are to be
secured. Fig. 7.12 shows a selection of frequency-response curves of the
Goldring variable-reluctance cartridge Type 500 taken under different
loading conditions. This shows clearly how the high-frequency response may
be considerably impaired by operation of the cartridge with resistive loads
other than the optimum value, or with capacitive or combination resistive
and capacitive loads.
The effect of capacitance on the circuit is important when it is

.: 1 rn111n11t1 100 1000

F~EOUENCY c/s
10000

FIG. 7.11. The network recommended for the Acos Black Sha.-!ow pick-up to provide a
velocity characteristic as shown by the curve. This enables the pick-up to be connected
by way of the magnetic pick-up terminals of the control unit.
150
 PICK-UPS AND RECORD PLAYING EQUIPMENT

+10

•5
db

-5
0 -- ~
--- ' .
~
D
A

-10 ""~
C
E
B

20 100 1000 10000 20000

CURVE R C

JI}··~·""
A 50K 0
B 20K 0
PU (INPUT R NOT C I0K 0
LESS THAN IM ) D >IM 250p
E 500p
F IOOOp

• 5
db

-s
0
-. r-,....._""' A

-10 r"\.."
C
B

20 100 1000 10000 20000

CURVE R C
A 75K 250p
B 30K 500p
C 20K IOOOp

F1G. 7.12. Frequency-response curves of the Goldring Type 500 variable reluctance
cartridge taken under different loading conditions of resistance and capacitance.

considered that pick-ups are often connected (of necessity) to the pre-ampli-
fier by way of screened cable whose capacitance value may be in the region
of 50 pF per foot of length. Then there are the stray circuit capacitances and
the capacitance reflected into the control-grid circuit of the first valve
because of the Miller effect. All these capacitances contribute to the general
loading and matching of the pick-up, though they cause little trouble (pro-
vided the pick-up lead is of reasonable length) if the equalizing network
stipulated by the maker is adopted.
Low impedance pick-ups, such as ribbon and moving-coil types, require
the use of an impedance-matching transformer, so that the pick-up impedance
can be stepped-up to match that of the first valve. Any odd transformer is not
suitable; the transformer designed specifically for the pick-up should always
be used. Transformers modify the overall pick-up circuit frequency-response,
and equalizing circuits designed to suit the pick-up and transformer are
always given by the makers.
151
 THE PRACTICAL HI-Fl HANDBOOK
Low-impedance pick-ups and matching transformers bring with them
hum troubles if the correct transformer is not used or if it is situated in a hum
field due to a smoothing choke or mains transformer. The transformer
should have a good-quality mumetal screen, and screened leads should be
used throughout (see also the sections on hum in Chapters 2 and 4). In some
cases mumetal screens are fitted to low-output magnetic pick-ups so as to
avoid hum voltages being induced into the pick-up windings from stray
alternating magnetic fields occurring in the neighbourhood of the pick-up
(e.g., from the turntable motor). The Goldring Type 600 unit uses a light-
weight mumetal case for this reason and as an added precaution, since the
push-pull arrangement of the pick-up coils also serves as a hum-cancelling
device.

TURNTABLE UNITS
It is not here intended to delve deeply into the mechanics and principles
of turntable units and record changers since full information of this nature
can be obtained free of charge on application to the various manufacturers.
However, there are one or two points of interest to consider.
Generally speaking, hi-fi enthusiasts are not lovers of record changers;
they usually prefer a good-quality four-speed transcription unit. Apart from
the skidding of records, one on top of the other, and the resulting wear and
speed variation, a lot of the fun of being a hi-fi enthusiast is lost when the
need for record changing is eliminated throughout a session. (This view may
not be shared by all, but it is the author's personal view and that of his
associates!)
The turntable unit, whether automatic or not, has to rotate the record
at a constant velocity whilst maintaining perfect balance to avoid such effects
as wow, flutter and rumble. This is achieved by the use of a carefully designed
and balanced turntable having a large proportion of its mass concentrated
at its periphery, well-engineered bearings and a good-quality constant-speed
driving motor.
As distinct from the centrifugal-governor type of motor which was
popular in the very early days of hi-fl, modern turntable units, including
autos, invariably use an induction-type motor to energize the turntable.
Although its speed is governed to a large degree by the frequency of the a.c.
mains supply, the induction motor is not truly synchronous, as is an electric-
clock motor for example. The motor is usually of the four-pole variety,
giving a loaded rotor speed of 1,320 r.p.m. or thereabouts.
Some transcription units incorporate a speed adjustment in the form
of an eddy-current brake which applies an even load to the motor and thus
permits control of the speed over a range of plus and minus 2½ per cent.
The Garrard transcription unit has such an arrangement. The load is applied
152
 PICK-UPS AND RECORD PLAYING EQUIPMENT
Fro. 7.13. The modern method TURNTABLE
of turntable drive. ____, SPI NOLE

by means of an aluminium
disk attached to the rotor
shaft passing between the
poles of a permanent mag-
net. The magnet is pivoted
in such a way that operation
of the speed control increases or decreases the field over the disk.
Another form of speed variation, used in the Goldring (Lenco) trans-
cription unit, takes the form of a speed-cone coupling between the motor
and the turntable; adjustment of the speed control alters the position of the
coupling wheel on the cone and hence the ratio of the coupling between the
motor and the turntable.
In most units, the drive to the turntable is by means of a rubber idler
wheel which engages with both a capstan on the motor spindle and the inner
edge of the turntable rim (see Fig. 7.13). The turntable speed is controlled
by the ratio of the diameters of the motor capstan and the turntable rim. The
idler wheel has no direct bearing on the speed of the turntable, a fact which is
not always realized. The idler wheel, or jockey pulley, is held under slight
pressure between the two drives by means of a spring.
There are various methods in use for obtaining the three or four speeds
(16i, 33¼, 45 and 78 r.p.m.). A popular method, employed in B.S.R. changers
and players, makes use of a four-speed pulley on the motor spindle and an
adjustable jockey assembly. Speed changing is effected by raising or lowering
the jockey wheel by a set of levers actuated by a control knob. This arrange-
ment is shown in Fig. 7.14.
The jockey wheel is made of
hard rubber, and should
ciRcuP therefore be kept absolutely
free from oil or grease.
A frequent fault on this
type of unit, preventing
correct selection of the
various speeds, is caused by
JOCKEY PULLEY
ASSEMBLY the four-speed pulley slip-
ping down the motor
spindle as the result of the

F10. 7.14. Speed change is

effected by a four-speed pulley.
153
 THE PRACTICAL HI-FI HANDBOOK
FIG. 7.15. The Go/dring-Lenco method of
player-unit suspension.

retaining grub-screws loosening.

Resetting the pulley is a simple

'BASE
BOARD
matter,as will be seen from Fig. 7.14.
To avoid a "flat" appearing on
the rubber idler wheel, the wheel is
automatically retracted on some units
when the motor switches off at the
end of a record or series of records, or when the speed control is set to the
"neutral" position.

WOW, FLUTTER AND RUMBLE

Speed variations of the turntable below a frequency of about 20 c/s are
referred to as wow, since this is the effect given to the reproduction as the
result of this fault. The general causes are unbalance in the turntable, slip due
to oil on the drive pulleys, a distorted or off-centre record, slip of one record
upon another on an auto unit, and non-concentricity of the idler wheel.
Flutter refers to speed variations of the turntable above a frequency of
about 20 c/s. The causes are untrue motor bearings (due to wear in the bear-
ings), unbalance of the rotor, slight eccentricity of the idler wheel or in the
mechanical coupling between the motor and turntable, and the transmission
of vibrations from the motor affecting the speed of the turntable.
A low-frequency rumble is sometimes superimposed upon the reproduc-
tion also as the result of slight unbalance in the motor and drive assemblies, or
motor vibrations transmitted to the pick-up way of the idler wheel and turn-
table. In this respect, resilience of the motor mounting is important, as also
is the method of mounting the unit on the motor board. Fig. 7.15 shows the
spring mounting arrangement adopted on the Goldring-Lenco transcription
units.
Crystal pick-ups are usually more sensitive to motor rumble than their
magnetic counterparts, which suffer more from hum pick-up. This is because
crystal units are more sensitive to amplitude changes at low frequency
than are magnetic types. Sometimes, due to this cause, an aucostical feed-
back path is promoted between the pick-up and loudspeaker by way of the
stylus, record, motor unit and loudspeaker mounting. When this happens a
low-frequency howl is emitted from the loudspeaker which can be stopped
only by turning down the volume control or removing the pick-up from the
record. The trouble is characteristic especially of the portable record player
154
 PICK-UPS AND RECORD PLAYING EQUIPMENT
or radiogram in which the loudspeaker is mounted in the same case as the
record playing unit.
A microphone valve in the amplifier may evoke a similar symptom, but
this trouble can easily be determined by gently tapping each valve in turn
with the end of a pencil. Persistent acoustical feedback of this nature should
lead to investigation of the spring mounting of the motor unit on the motor
board. It should be ascertained that the clamping screws, which secure the
motor unit during transit, are either fully released or removed, depending
upon their nature, so that the unit is freely suspended upon the springs.
Sometimes it may be necessary to replace the crystal cartridge if acoustical
feedback cannot be cleared by other means.

RECORD CHANGERS
When servicing record changers it is always desirable to support the unit
at working level on two small boxes. The boxes or blocks should be situated
so that they are clear of the mechanism and permit normal operation of the
unit. The various operating cycles can be observed by placing a flat mirror
on the table or bench beneath the changer. Generally speaking, once the
sequence of operations has been carefully observed over several cycles, the
cause of the trouble becomes apparent and it requires only a logical approach
to apply a remedy. However, if the fault appears complex, or if the action of
the sections cannot be understood, the service sheet appropriate to the unit
should be studied.
With modern units, the turntable drive is similar to that associated with
single players. The intermediate idler wheel is used, and speed change is
effected by means of the stepped motor pulley, as already described. There is,
however, a secondary drive and cycling mechanism which serves to change
the record. This is brought into action by rapid movement of the pick-up
when tracing the play-out groove or by operation of the "start" control.
Several things happen as the result of the operation of the cycling mech-
anism. The next record in the pile on the turntable spindle drops on to the
turntable, the pick-up lifts and moves to its setting-down position-this often
being determined by the operation of a "feeler" as the record drops-and
the cycling action ceases until the pick-up is again moved rapidly by the play-
out groove or until the "reject" control is turned. When the last record is
played the unit switches off automatically, since then the control arm which
rests on the pile of records is at its lowest level. There are a number of varia-
tions of these actions, but the results are always the same.
The automatic trip which is brought into operation by the play-out
groove on the record is common to both auto units and single players, though
transcription players do not always employ it. There are two general types
operated by either a velocity trip or ratchet. The former type is the most
155
 THE PRACTICAL HI-FI HANDBOOK
popular and this takes the form of a quadrant which carries a trip lever by
way of a friction coupling. The quadrant is coupled to the pick-up arm and
moves with it as it traverses the record. At the end of the trip lever is a small
felt pad; on the turntable spindle is a striker which makes contact with the felt
pad when the pick-up nears the centre of the record.
As the record is played the trip lever moves very slowly inward, and at
the point where the felt pad makes contact with the striker the trip lever
is pushed back against the friction coupling on the quadrant. This action
continues at each revolution of the turntable until the trip lever is accelerated
by the pick-up stylus tracing the widely spaced play-out groove; the striker
then positively engages with the felt pad and pushes the trip lever in such a
way that the motor on/off switch is actuated. In some units a turntable break
is also brought into action.
The ratchet arrangement includes a lever connected to the pick-up arm
to which is attached a lightly loaded pawl. As the arm moves towards the
centre of the record the pawl rides over a lever-type ratchet which is mechan-
ically coupled to the on/off switch. When the stylus traces the play-out groove
the lever which is attached to the pick-up arm oscillates to and fro owing to
the eccentric nature of the groove spacing. The pawl thus engages with the
ratchet when the movement of the pick-up arm reverses, the associated lever
comes into play and the switch disconnects the motor from the mains supply.
With auto units, the trip mechanism starts the cycling action of the record-
changing mechanism.
These automatic functions are much of a compromise between the
mechanics of the unit and the mechanical and electrical nature of the pick-
up. Whilst modern units present little load on the stylus, the fact that a load
is presented to operate the mechanism puts the idea out of favour with hi-fl
types. There is, of course, the possibility of impaired tracking and groove
jumping as the result of the additional load of the auto-stop mechanism, and
a soft, though definite, knock is heard in some units at each revolution of the
turntable as the striker makes contact with the felt pad on the trip lever.
For these reasons transcription units use a manual on/off switch.
When servicing auto units, care should be taken to avoid the application
of too much oil as several of the functions rely upon friction. The motor
bearings themselves are usually of the self-oiling type, though occasionally a
few spots of very light oil are required on the felt pads at the seat of the
bearings.
With transcription units, the heavy turntable (the weight of the turntable
on the Collaro Model 4T200 unit is 8½ lb.) is fitted with a highly machined
shaft which runs in a bearing (sometimes self-lubricating) with a steel ball
pressed into its lower end. This ball takes the total thrust of the turntable, re-
duces friction to a remarkably low level and eases wow and rumble problems.
156
PICK-UPS AND RECORD PLAYING EQUIPMENT

FIG. 7.16. The Col/aro Conquest record-changer.

FIG. 7.17. The Col/aro 4T200 transcription unit.

157
 THE PRACTICAL HI-Fl HA-..DBOOK
F1G . 7.18. 11/wtrating tracking error.
The broken line sho.,.·s the path traced
by the recording stylus and the Jul/
line the path traced by the reproducing

0
stylw.

Other features sometimes found are a groove-location device (Goldring-

Lenco Models GL55 and GL56) which introduces the stylus to the start of
the groove, a stroboscopically-marked turntable to determine the accuracy
of the speed of the turntable, and a spring- or weight-loaded device on the
pick-up arm which can be adjusted to provide a range of stylus pressures.
Figs. 7.16 and 7. 17 show the Collaro Conquest record changer and Model
4T200 transcription unit respectively.

TRACKISG PROBLDtS
During the recording process the recording head is arranged to move
radially across the disk, as shown by the broken line in Fig. 7.18. On replay,
however, as the result of the pick-up swinging in an arc about the pivot of
the pick-up arm, the stylus follows the full-line curve in Fig. 7. 18. This means
that over most of the record the oscillations of the reproducing stylus, as the
result of the impressed modulation pattern, occur at an angle to those of the
recording stylus. The reproducing stylus deviates from the true lateral move-
ment by an amount depending upon this angle of error, usually referred to as
"tracking error".
Whilst this leads to even harmonic distortion, the stylus has a reduced
compliance away from the true lateral movement and greater energy is
required to cause it to vibrate, resulting also in greater wear on the sides
of the groove where the tracking error is greatest. The tracking error can be
kept at a very low figure by mounting the arm so that the tip of the stylus
overhangs the centre of the turntable by a small amount, by offsetting the
pick-up head on the arm at a slight angle, inclined towards the record centre
(this is catered for during manufacture), and by the use of an extra-long
pick-up arm-there is a limit to this, of course, though 16-in. arms are often
used for studio work.
The recommended position for the pick-up, giving the optimum over-
hang, is stipulated by the manufacturers, often in the form of a template
for the initial setting-up of the equipment. There are critical values for over-
158
 PICK-UPS AND RECORD PLAYING EQUIPMENT
hang and pick-up head displacement which result in two positions of zero
error. For example, an 8-in. arm can be set so that the maximum error at
two points is only 2½ deg., while by the use of a 16-in. arm the maximum
error can be reduced to I½ deg. On the other hand, a poorly mounted 8-in.
arm and pick-up can lead to anerrorofsome 18 deg., producing considerable
distortion at high recorded levels.
A method which virtually cancels the tracking error makes use of a
special type of pick-up arm on which the pick-up head carrier is pivoted.
As the stylus traces the groove over the radius of the record so the pick-up
swivels on its pivot in a way that counteracts the error. This idea is embodied
in the Burne-Jones (8-J) arm.
As already mentioned, the low-frequency resonance of the pick-up
system is somewhat governed by the pick-up arm. Hi-fi arms usually have a
very low resonance-well below the troublesome level. The Goldring Jubilee
arms, for instance, have a resonance in the region of 9 c/s. An average arm
has a resonance slightly above this figure-15 c/s to 30 c/s. Other spurious
resonances sometimes occur in less exacting arms due to torsional effects.
The bearing friction of pick-up arms should also be at a low level.
This friction is usually measured in terms of the weight required to overcome
the friction. A good arm may have a value equivalent to ½-gram, while 4 or
5 grams may represent the value if the bearing is tight or if the arm is poorly
designed. The average value is something like I ·5 grams. A high value of
bearing friction may promote groove-jumping, particularly if a low tracking
pressure is used. Other causes of this symptom are (I) a chipped or worn
stylus, (2) the use of a pick-up with too small a value of lateral compliance,
particularly when used with modern records of extended frequency range,
and (3) a faulty record-this should be suspected if groove-jumping
persistently occurs at a certain spot on the record.

159
 CHAPTER 8

Microphones and Mixers

A MICROPHONE performs two essential functions; it first

transforms the sound energy it receives into vibrations, and then transforms
the vibrations into a voltage whose pattern matches the original sound. A
microphone is, in fact, a generator of electricity, but is driven by sound
energy instead of mechanically. In the days when the author first began to
study microphones. he conceived an idea that he thought would help reduce
the cost of electricity. He proposed the installation of giant exponential
horns on noisy railway stations which would "capture" the wasted sound
energy and turn it into electricity through the medium of huge microphones
loaded to the throats of the horns! His tutor was pleased that the basic idea of
microphones had been grasped, even though he didn't seem very impressed
with the scheme!
Excluding the "lo-fi" carbon microphone, there are four basic types
which are employed in hi-fi work: (1) crystal, (2) moving-coil, (3J ribbon,
(4) condenser. With all types there is a diaphragm, or a moving member,
which is caused to vibrate by the sound energy and which actuates the
generating system.
There are two methods by which the sound energy can evoke sympathetic
vibrations of the diaphragm. Except in the case of the ribbon microphone,
which has a thin ribbon instead of a diaphragm, the sound is applied to one
side of the diaphragm only and the pressure component of the sound radia-
tions is utilized. The rear of the diaphragm is cut off from the pressure wave
by the microphone housing and is thus at normal atmospheric pressure, which
results in the diaphragm being deflected inwards and outwards by the pressure
variations in accordance with the pattern of the sound radiations. Micro-
phones adopting this principle are said to be pressure operated.
In addition to the pressure component of a sound wave, there is another
component called the particle velocity (see Chapter 1). Since it is not possible
to produce a ribbon of sufficiently slender dimensions to couple with high
efficiency to the velocity component of a sound wave, there can be no such
thing as a purely velocity-operated microphone, though this term is some-
160
 MICROPHONES AND MIXERS
times used in regard to ribbon microphones. Ribbon microphones represent
an approximation to velocity operation, since both sides of the ribbon are
exposed to the sound field, and movement of the ribbon is caused by the
sound pressure difference between the two sides. Such microphones are known
as pressure gradient types, and since the pressure gradient of a sound wave is
proportional to the particle velocity, there is some excuse for the term
..velocity operated".

CRYSTAL MICROPHONES

The generating system of a crystal microphone, as of the crystal pick-up,

is Rochelle salt crystal which produces an electrical potential difference when
subjected to changes of pressure. The crystal is cut from a big crystal at a
critical angle and placed between metallic plates arranged so that the pressure
on the crystal is varied in sympathy with the vibrations of the diaphragm.
The idea is shown in Fig. 8.1. The voltage thus generated varies in accordance
with the sound vibrations.
There is another type of crystal microphone, generally referred to as the
sound-cell microphone, in which the sound pressure operates the crystal
directly without a conventional diaphragm. This type is considerably less
sensitive than the diaphragm-actuated type, but its frequency response is
much better since there is no coloration from the diaphragm.
Crystal microphones are high-impedance devices and can, therefore,
be connected direct to the grid circuit of a valve. They are not affected by
magnetic hum fields, are fairly light in weight and, in the case of the dia-
phragm type, provide a fairly high output voltage, thereby avoiding the
necessity of high-gain microphone amplifiers. They are used extensively by
home tape-recordists, and are useful for a number of applications which do
not need long connecting cables
-these, because of the
necessary screening, would be
: ; ROCHELLE
likely to attenuate the higher ; / SALT
CRYSTAL
frequencies.

MOVING-COIL MICROPHONES
MICROf'HONE
This microphone is basi- AMPLIFIER

cally the same as the moving- 'GRID

RESISTOR
coil loudspeaker (see Fig. 8.2).
The diaphragm is mechanically
coupled to the coil which
operates in the air gap formed FIG. 8.1. Crystal micro-
by the poles of a permanent phone and its connexion to
magnet. The winding thus cuts ) a high-impedance circuit.
161
 THE PRACTICAL HI-Fl HANDBOOK

MOVING
COIL

MICROPHONE
AMPLIFIER

GRID
RESISTOI'.

FIG. 8.2. The moving-coil (dynamic) microphone. Its low impedance requires the use
of a matching transformer.

the lines of force of the magnet when it is actuated by the diaphragm. A

voltage is, therefore, produced whose magnitude is proportional to the rate
of cutting of the lines of force, or the velocity of the coil. Moving-coil units
are of low impedance (around 30 ohms), and usually give a smaller output
than the crystal unit. However, they lend themselves to operation at greater
distances from the amplifier than crystal types, since a low-impedance line
is less subject to losses of all kinds. A matching transformer is required,
and this is usually mounted in or near the amplifier.
RIBBON MICROPHONES

Fig. 8.3 shows the basic construction of the ribbon microphone. It

operates in the same way as the moving-coil unit, in that the ribbon represents
the moving conductor. Both the output voltage and the impedance are very
low, and to bring the impedance up to a reasonable figure a small transformer
is often incorporated in the stem of the microphone. The Reslo unit embodies
this feature.
CONDENSER MICROPHONES
As with the electrostatic loudspeaker, the condenser microphone (the
term electrostatic microphone is rarely used) is essentially a condenser formed
of two plates separated by the air as a dielectric. One plate is fixed, while the
other serves as the diaphragm and is caused to vibrate by the incident sound
pressure. The capacitance across the two terminals thus varies in accordance
with the sound pattern. A polarizing voltage is required, and is connected in
series with the microphone by way of a high-value resistor which, irrespective
of diaphragm movement, holds the charge on the microphone at a fairl:y
constant value. Thus, as the capacitance alters in value due to sound pressu1 e,
the potential across the capacitor varies accordingly, since this is equal to
the charge divided by the capacitance. This varying potential is applied to the
grid circuit of the microphone amplifier valve, as shown in Fig. 8.4.
162
 MICROPHONES AND MIXERS
,----------H.T. •
HIGH RESISTANCE

VERY
LOW MICROPHONE
TO LINE AMPLIFIER
IMPED· (MEDIUM
ANCE IMPEDANCE)

GRID
RESISTOR

POLE PIECES

(Left) FIG. 8.3. The ribbon microphone. Its very low output impedance usually re-
quires the use of a microphone transformer in the housing to produce a reasonable
line impedance. (Right) FIG. 8.4. The condenser microphone requires a polarizing
voltage, which is obtained from the h.t. line through a resistor.

This type of microphone is used with certain Continental tape recorders,

and also for laboratory tests and studio applications. Since the output
impedance is extremely high, and the output voltage is affected by cable
capacitance, a small pre-amplifier is sometimes built into the microphone
housing so that the line impedance can be reduced to a workable
value.
POLAR RESPONSES
Within the limitations of frequency, the pressure-operated microphone
is essentially omnidirectional, i.e., it is responsive to sounds arriving from
any direction within its range. Its polar response thus has a spherical distri-
bution. However, at frequencies where the wa,velength of the sound becomes
comparable with the size of the housing, it tends to become unidirectional,
and will have greater sensitivity to sound arriving at the front. This is illus-
tr,lted in Fig. 8.5.
The pressure-gradient microphone, on the other hand, has a figure-of-
eight polar response, as shown in Fig. 8.6. This kind of microphone
does not respond at all to sounds
SIDE arriving at the sides but has a
HIGH FREQUENCIES usable response over about 100
deg. both at the front and rear.
MICROPHONE
The polar response can, however,
o"FRONT REAR 1ao•
FIG. 8.5. The omnidirectional response
of pressure microphones tends to be-
come unidirectional when the wave-
length of the sound is comparable with
SIDE the microphone dimensions.
163
 THE PRACTICAL HI-Fl HANDBOOK
SIDE FIG. 8.6. The figure-of-eighr response of
MICROPHONE
pressure-gradient microphones.

FRONT REAR
o• ieo• be modified to suit the prevailing
acoustical conditions by partially
/ closing the rear of the microphone by
/ '
SIDE ' means of small acoustic filters (pads).
A cardioid (heart-shaped) polar
response can be obtained from a
microphone which combines the output from a pressure-operated unit
with the output from a pressure-gradient unit. Combined microphones of
this kind, known as cardioid microphones, are used extensively for
broadcasting work. The cardioid response diagram is given in Fig. 8. 7.
Because they combine a high-quality pressure-operated unit with a
pressure-gradient unit, whilst maintaining a sensitivity and acoustical
balance over the greater part of the sound spectrum, true cardioid
microphones are rather costly instruments, and are usually too expensive for
the average enthusiast. However, they are sometimes employed by the
"serious" amateur tape and disk recordist, and by organizations operating
sound-reinforcement services.
It is interesting to note that a semi-cardioid response can be obtained from
a pressure-gradient microphone by closing the back half of the ribbon with
an acoustical filter. Two responses are thus obtained, circular and figure-of-
eight, which combine to give the cardioid response. A cardioid response can
also be obtained from specially constructed condenser microphones, in which
two diaphragms are used, separated by a perforated electrode.

MICROPHONE SENSITIVITY
The sensitivity of a microphone is usually expressed in decibels relative
to a fixed reference level. The reference level chosen is invariably l volt
(equals O db) with a sound pressure of I dyne per square centimetre (I dyne/
cm2). Thus, a microphone quoted as having an output level 60 db below
l volt/dyne/cm 2 would generate about l
millivolt when subjected to a sound
SIDE
pressure of l dyne/cm 2 • A sound pres- MICROPHONE
sure or sound intensity of twice the
FRONT REAR
o• ---+--...------1eo•
FIG. 8.7. A cardioid or heart-shaped response
is obtained by combining the principles of
the pressure and pressure-gradient units. SIDE

164
 MICROPHONES AND MIXERS
value would increase the output voltage by a factor of 2, while a sound pres-
sure of half the value would decrease the output voltage by a factor of 0·5.
The overall sensitivity is somewhat governed by the output impedance.
For instance, the Lustraphone Full-Vision microphone is quoted as having
an output of - 88 db at 25 ohms, and an output of - 54 db at 50,000 ohms.
In the latter case an impedance step-up transformer is used, which also
increases the output voltage.

CHOICE OF MICROPHONE
No hard-and-fast rules can be given in this connexion, since the final
choice depends not only upon the particular application, but also upon
economic factors. Nevertheless, no one microphone does everything equally
well; the diversity of situations for which microphones are required calls for
different types if results of the highest order are desired. In this case, consider-
able knowledge of microphone techniques is essential; the amateur may well
obtain better results from the use of one versatile ribbon unit than by the
unskilled arrangement of an array of more specialized instruments.
Indeed, the ribbon microphone of modem design can be used for almost
all applications if reasonable thought is given to its positioning; the response
pattern can easily be varied to suit special conditions by the inclusion of
small acoustical filters, as already described. The modern unit has a good
sensitivity, and can thus be connected direct to most amplifiers and tape
recorders without the need for pre-amplification. The majority of commercial
units employ inbuilt transformers providing an output impedance sufficiently
low for connexion to long lines, whilst also providing a good match to the
input impedance of most amplifiers. The Resto Type RB miniature ribbon
microphone can be obtained with output impedances ranging from 30 ohms
to several thousands of ohms (i.e., high impedance). The Lustraphone range
of ribbon microphones are also available with various impedance values.
Ribbon microphones have excellent frequency-response characteristics,
often maintained substantially level up to 14-15 kc/s. This type of microphone
is therefore usually ideal for the recording or reproduction of music in all its
aspects. It is not always suitable for outdoor work, where the delicate ribbon
may be affected by wind pressure, though it is possible to employ so-called
windshields to minimize this disturbance which manifests itself in the form of
a roar from the loudspeaker.
Owing to the bi-directional characteristic, the orientation of the ribbon
microphone can be adjusted so as to discriminate against unwanted pick-up
off the main axis. This feature can be used to advantage to provide a fair
degree of balance when a single unit is employed for the reinforcement or
recording of an orchestra.
With its general freedom from response peaks, which are inherent in less
165
 THE PRACTICAL HI-Fl HANDBOOK
exacting instruments, the ribbon microphone can also be used for sound-
reinforcement applications in rooms which are liable to produce acoustic
feedback (the howl effect when the amplifier gain-control is advanced). In
spite of the inherently lower sensitivity as compared with, say, moving-coil
microphones, improved acoustical efficiency is often possible by the use of a
ribbon unit.
The recording operator or sound-reinforcement engineer should always
make a special point of instructing the artist or speaker in the use of the
microphone. A few minutes spent in the serious consideration of this point
is well worth while. Incidentally, considerable accentuation of the lower
frequencies, resulting in a "boominess" of reproduction, occurs if a ribbon
microphone is used too close to the sound source. If this type of microphone
is used closer than about 3 ft. a suitable degree of bass cut should be applied
at the amplifier. Perhaps this is the reason why crooners favour the ribbon
microphone!
The moving-coil or dynamic microphone is more sensitive than the
ribbon unit; it is also more robust, less expensive, and suitable for outdoor
as well as indoor functions. It is a popular unit with tape recordists generally
and with public-address operators (it should be observed that the term
"sound-reinforcement" has been taken throughout this book to mean "hi-fi
public address"). The average frequency response of this type of microphone
usually falls short of that of the ribbon unit and ranges about 8 kc/s. Its
omnidirectional characteristic makes it difficult to avoid acoustic feedback
effects in some applications.
The crystal microphone is also used by tape recordists, though it is
losing favour with public-address and sound-reinforcement operators because
of its high output impedance. It is usually less expensive than the other types
considered. The output voltage is a little higher than for moving-coil units,
and both its frequency response and response characteristics are rather like
those associated with moving-coil units, though they vary widely in different
designs. This microphone is also employed in office dictating machines.
The condenser microphone is rarely seen in amateur circles, but, as
already mentioned, is sometimes employed with Continental (i.e. Grundig)
tape recorders. It has an excellent frequency response, and certain specialized
types have been produced which respond to frequencies up to 100 kc/s!
There is a great diversity of designs of the three basic units. There are
microphone heads of various types for screwing to a floor or table stand,
microphones complete with table stand, hand microphones, so-called full-
vision microphones designed to avoid hiding the artist (these are often seen
on television), lapel microphones, noise-cancelling microphones and others.
It is outside the scope of this book to describe the merits and demerits of all
these types, but in all cases the functional units are similar to those described.
166
 MICROPHONES AND MIXERS
MICROPHONE MIXERS
There always comes a time when it is necessary to use more than one
microphone. Microphones can be connected in parallel and then to a common
microphone input socket on the amplifier, but this practice is not to be
recommended. It is far better to use a microphone mixer so as to maintain
optimum matching of the microphones to the input impedance, whilst at the
same time having full control over the gain setting of each microphone
channel.
A circuit of a microphone mixer (Pamphonic Sound Equipment) is
given in Fig. 8.8. It will be seen that each microphone is fed into its own pre-
amplifier valve, and the outputs are combined, at a level determined by the
setting of the appropriate "gain" or volume controls, and then fed to a
common voltage amplifier, and thence to the common output transformer.
The five input transformers and the output transformer ensure that the
correct load is presented to the microphones and the microphone input
channel of the main amplifier or pre-amplifier; which in turn results in the
maximum transfer of signal with the minimum generation of noise. whilst
exploiting the frequency-response characteristics of the equipment to the
full.
The 330k resistors connected to the sliders of the volume controls avoid
heavy loading on the grid circuit of the output triode when only one channel
is in operation; i.e., when four of the controls are backed right off. A degree
of frequency correction is also applied to this stage through frequency-
selective feedback being given by the 680k resistor and the 0·1 mF capacitor
connected between grid and anode. Further correction is applied across the
primary of the output transformer T6.
The mixer has its own power supply, which uses a Mullard EB91
(usually employed as a signal detector) as the h.t. rectifier. In order to keep
the valve within its limits of operation, the circuit is arranged in the form of
a voltage-doubler, and the potential between the heater and cathode of the
valve is reduced by the heater being connected to a point of positive potential.
The author has had frequent occasion to use this instrument, and it has always
proved reliable and has given virtually no trouble at all.
Another neat little four-channel mixer is the Grundig Type GMU3.
This is designed essentially for use with Grundig tape recorders, and two of
the channels cater for the Grundig condenser microphone by having the
necessary 100-volt polarizing voltage available to these circuits. Of the other
two channels, one is suitable for a low-impedance microphone, such as the
Grundig ribbon unit, and the other is intended to accept a fairly high-level
(approximately 300-mV) signal, such as that given by a radio receiver, ampli-
fier control unit or another tape recorder. A magic-eye signal-level indicator
is also included on the front panel.
167
 I0K

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 MICROPHONES AND MIXERS
TRANSISTOR UNITS
Although power transistors are now used in the output stages of public-
address amplifiers of the semi-portable variety, they have not yet found their
way into the comparable stages of hi-fi equipment. Whilst transistors are
capable of delivering some l0-20 watts or more of audio, the distortion con-
tent is above hi-fi acceptance (it is difficult to keep harmonic distortion
below about 5 per cent). However, in low-level audio stages the transistor is
now beginning to be exploited. One application is in microphone amplifiers
and mixers. Recent introductions in this field are the transistorized pre-ampli-
fier units by Lowther and the transistorized mixer unit by Lustraphone.
The mixer unit has four channels, two of high impedance and two of
low impedance. The output circuit is suitable for direct connexion to the
high-impedance socket of almost all hi-fi amplifiers or control units. The
power is provided by a single miniature mercury cell, which has an estimated
life of some 1,000 hours. The frequency response is substantially flat from
50 c/s to 14,000 c/s.
Since transistors are inherently free from hum and microphony trouble
that are always present to some degree with high-gain valve amplifiers, they
are ideally suited to high-gain "front-ends". Small transistor amplifiers can
easily be built into the housing of low-level microphones, including the
battery power supply. Since a signal of high level can thus be distributed
from the microphone circuit, the need for high-gain settings on the main
amplifier is precluded, and the danger of hum and noise pick-up on the micro-
phone cable is considerably alleviated.
Another advantage of the transistor is that its input impedance can be
arranged to match the low impedances of high-quality electromagn(;tic
pick-ups and microphones, without the need for a matching transformer.
The transistor thus serves admirably as an impedance-matching device.
A circuit of a pre-amplifier suitable for electromagnetic pick-ups or
microphones of from 100 to 1,000 ohms impedance is given in Fig. 8.9.
A transistor can be looked upon as two crystal diodes formed between the
emitter and base and the collector and base. In the circuit the letters B, C
and £ around the transistor symbol represent base, collector and emitter
respectively. These three points are often likened to the electrodes of a triode
valve as follows: collector = anode, base = grid, and emitter = cathode.
The transistor is biased in the forward direction in the emitter/base
circuit and in the reverse direction in the collector/base circuit. With the
emitter/base bias disconnected there is theoretically no current (a very small
amount in practice) in the collector/base circuit. When bias is applied to the
base/emitter circuit, however, current flows in the base/collector circuit,
and when the current in the base/emitter circuit increases, the current in the
base/collector circuit also increases, but to a greater extent. This action is
169
 THE PRACTICAL HI-FI HANDBOOK
promoted by the emission of so-called positive holes from the emitter to the
collector circuit with a consequent lowering of the resistance of the base/
collector circuit.
The signal is applied to the transistor so as to cause variation of the
negative current in the base/emitter circuit, which, depending upon how
the circuit is arranged, results in an equal or greater variation of current in the
base/collector circuit. Generally speaking, there occurs a power gain because
the applied signal promotes a current change in a low-resistance circuit (base/
emitter) while reflecting a similar or greater current change in a high-
resistance circuit, represented by the base/collector junction.
There are a number of methods by which the transistor can be connected
into the circuit, as with the triode valve. The arrangement in Fig. 8·9 is
usually referred to as a "common base" circuit (the SmF capacitor connected
to the base makes the base common to the input and output circuits, or
earthed base circuit, and it corresponds roughly to an earthed-grid valve
circuit. The input signal is fed into the emitter by way of the 25mF coupling
capacitor (such a large value being necessary to maintain a low-frequency
response in a low-impedance circuit), and the output signal is taken by way
of the O· l mF capacitor from between the collector and positive line (chassis).
The circuit is thus given a low input and a high output impedance, which is
ideal for feeding a signal from a low-impedance microphone or pick-up to
a high-impedance input circuit of an amplifier. There is no reversal of phase
between the input and output signal voltages.
The circuit is capable of delivering approximately 1 volt r.m.s. of signal
for an input of 16mV r.m.s., and thus has a voltage gain of some 62 times.
Power is derived from a 6-volt battery, and due to the very low current drain
(400 microamps quiescent) a very small battery is all that is required. The
response is about 3 db down at 100 c/s and 20,000 c/s relative to 1,000 c/s.
The 6·8k resistor connected to the collector can be considered as the
output load, while the resistor connected to the emitter and the two resistors
whose junction is connected to the base serve to stabilize the circuit from the
d.c. aspect.
One or two points 18K

regarding the servicing 0·11'

of transistorized equip- ~H
ment will be useful. IMPEDANCE
OUTPUT
Transistors are very
MICRO·
sensitive to heat, and PHONE '70

FIG. 8.9. Circuit diagram IK

6V
,~( a transistori:.ed micro-
phonl' pre-amplifier.
170
 MICROPHONES AND MIXERS
if overheated when operating are liable to be destroyed in a very short
time. They should therefore be kept clear of soldering irons and heat-
producing devices such as valves and resistors of the main amplifier.
Soldering in and out of the circuit should be performed as rapidly as
possible. A miniature low-power soldering iron is desirable, but even then a
heat "sink" should be produced by holding the transistor wires with long-
nose pliers while soldering is being done. The transistor wires should never
be bent close to the seal on the transistor.
Reversing the polarity of the supply voltage will almost certainly result
in immediate failure of the transistor. This must be borne in mind when
performing in situ voltage, current and resistance checks. It is worth remem-
bering that the negative terminal of most multi-range meters is usually in
connexion with the internal battery positive connector when used as
ohmmeters.
So as to avoid disturbing the balance of voltage in a transistor circuit,
voltage measurements are best made on a high-resistance instrument of
at least 1,000 ohms-per-volt. It is not a good thing to make or break a con-
nexion in a transistor circuit while the supply voltage is connected. If it is
required to perform a current test, the supply voltage should be disconnected
and then the milliammeter inserted where required. The power can then be
re-connected and the measurement taken. The same applies on removing
the milliammeter.

SERVICING MICROPHONES
Generally speaking, it pays to let the maker have the microphone back
if the need for servicing arises. Replacement diaphragms and ribbon elements
can, however, be obtained for most quality microphones, as can crystal
inserts for sound cell units. It is only a little more expensive to let the maker
replace the faulty parts, whilst at the same time ensuring that the performance
of the equipment will be up to the normal standard. In a number of cases, the
microphone fixing screws are sealed so as to avoid unnecessary tampering,
and if these seals are broken the manufacturer may charge for the correction
of a fault, even during the guarantee period.
With moving-coil and ribbon units, continuity should be registered
across the terminals, and a low-value resistance reading is obtained if the test
is directed across the moving coil or ribbon element. A higher resistance
reading will, of course, be obtained if the test is made across the secondary
of an internal transformer. A crackling is usually heard from the microphone
when this test is made, due to the battery in the testmeter causing the micro-
phone to act as a loudspeaker. Crackling noises are also heard from crystal
units when subjected to this test, in spite of the normal lack of continuity
as indicated on the ohmmeter.
171
 THE PRACTICAL HI-FI HANDBOOK
Microphone switches are a constant source of trouble on certain micro-
phones, but this is usually fairly easy to remedy. Broken conductors in micro-
phone cables also represent a frequent source of trouble, particularly if the
microphone is in constant employment and moved around a lot. These
faults are quickly located by means of simple continuity or resistance
checks.

MICROPHONE BALANCI:
As we have already seen, the choice of microphone is somewhat governed
by the polar characteristic and the frequency response, with due regard also
to such things as sensitivity, the type of material to be amplified or recorded
and (most important) the depth of one's pocket. The beginner invariably
commences operations with a relatively cheap crystal microphone-very
often the one supplied with the tape recorder or amplifier, though some
manufacturers are now wisely leaving the choice of microphone to the user.
With a little experience, the beginner soon realizes that something more
leaborate in the way of a microphone is desirable, if only to cut out the
squeak of the door, the tick of the clock or the crackle of the fire. It is
amazing how such noises assume prominence on a tape recording. While the
ear is able to discriminate against unwanted noises, since there are two of
these organs (stereo helps in this respect), the microphone responds to every
noise and brings both wanted and unwanted sounds into focus at the loud-
speaker. If one microphone is used at some distance from the sound source,
then the ambient sounds are going to be recorded at almost equal intensity.
The novice gradually discovers such things for himself, and possibly
experiments with various microphones, combinations and orientations.
This is a good thing, because it provides the necessary experience in micro-
phone technique for which words can never be used as a substitute.
When using more than one microphone, particularly if the microphones
are connected in parallel across the amplifier's common microphone input
socket, care must be taken to ensure that they are not placed equidistant
from the sound source. This is because there is the possibility of the micro-
phones being out-of-phase (there is an analogy with out-of-phase loud-
speakers), in which case serious distortion would occur as the result of
cancellation effects at certain frequencies. If this trouble is suspected, the
connexions on one of the microphones should be reversed, or one of the
microphones should be turned through 180 deg. if it is of the ribbon type with
a figure-of-eight response.
In general, however, even if the microphones are phased correctly,
it is not good policy to use them close together, because interference effects
of the nature described may result at certain frequencies. If the response
characteristics of the microphones are known (they can usually be estimated
172
 MICROPHONES AND MIXERS
fairly accurately), they should be orientated with regard to each other so
that their polar responses do not overlap to any large degree.
When acoustic feedback is troublesome (with, for example, sound-
reinforcement work), the placing of the microphone is of great importance.
If it is found that insufficient audio power can be obtained before the feed-
back point on the volume control, and it is impossible to re-position the
microphone, other microphones should be tried, such as the ribbon or
cardioid. Just before reaching the setting of the volume control which evokes
the characteristic howl, a slight ringing sound may be heard when the micro-
phone is being used. At this point distortion may also be at a high level, and
for this reason the volume-control setting must be retarded.
Intelligent use of the treble and bass controls may allow a greater volume
setting to be used, since the acoustic feedback is a function of the room
acoustics. A "live" room, for example, will reflect the higher-frequency
sounds and possibly promote feedback conditions, while a "dead" room will
tend to absorb certain frequencies of the sound and thus prevent it bouncing
back into the microphone. Line-source loudspeakers assist in this respect
also, as we have already seen, by concentrating the sound over the required
area of coverage and leaving little for spilling into the microphone.
The reverberation of a room has an appreciable effect on a recording. If
the microphone is placed a reasonable distance away from the sound source,
then it is going to pick up not only direct sound, but also quite a lot of reflec-
ted sound; the recording will be coloured by the room acoustics. If the micro-
phone, on the other hand, is placed fairly close to the sound source, the room
acoustics will have less influence since most of the sound will be picked up
direct, and only a small proportion will be reflected sound. It is, in fact,
possible to arrange the position of the microphones to secure almost any
required degree of recorded reverberation effect.
Too close a position of the microphone in relation to the sound source
should be avoided for most applications, however, since this tends to promote
a "bass heavy" effect, but is possibly useful for the recording of dance bands
and rhythm groups, where plenty of bass may be required. It should also be
remembered that the sound radiation from musical instruments varies
considerably with frequency. With a piano, for example, the maximum
treble occurs to the right-hand side of the keyboard, and diminishes pro-
gressively towards maximum bass in an arc towards the rear of the instru-
ment. With string instruments, the maximum treble is confined to a narrow
angle from the major dimension of the instrument.
It is obviously impossible to explore microphone balance in relation to
all musical instruments, and from the point of view of the home recordist
and enthusiast it often comes down to a matter of trial and error, aiming
for overall balance without introducing undue coloration.
173
 CHAPTER 9

The Use of Tape

Q NE major advantage that tape has over the disk is that it is

a recording medium which can be exploited with almost equally good results
by the amateur enthusiast as well as by the professional recording engineer.
A high-quality disk-recording outfit necessarily costs much more than the
average enthusiast can afford, whereas a hi-fi tape recorder complete is little
more expensive than, for example, a hi-fi amplifier and loudspeaker system
or a television set.
Interest in tape recording is growing rapidly, and its popularity has
undoubtedly been boosted by recent developments in stereophonic reproduc-
tion; one of its most attractive attributes is the comparative ease with which a
library of tape records can be built up by recording radio programmes,
preferably those broadcast on the VHF-FM system (see the author's "F.M.
Radio Servicing Handbook"). The quality of reproduction of a tape record
made by this means can be as good as that obtainable from long-playing
disk records. Tape records, like disk records, are also available commercially
and are made in both single-channel and dual-channel (stereophonic) versions.
Not only music lovers, but amateur dramatic societies, educational authori-
ties, business, science and industry generally are finding more and more uses
for tape recording.
Magnetic tape is also used for the recording of television pictures;
as with sound, pictures are broken down into component parts to form
electrical impulses, and the impulses are recorded on the tape. The pictures
are thus "stored" for future use, and can be reproduced whenever required
by playing the tape back through the same machine, which then serves to
change the impulses back into the original pictures.
There is little doubt that the enthusiast of the future will have available
a vision tape player which can be plugged into the video channel of a tele-
vision receiver for the showing of pre-recorded tapes. Vision tape records,
also carrying a sound accompaniment, will be available, and the enthusiast
will eventually be able to make his own vision records with the aid of a small
television camera and a tape recorder; immediate replay facilities will be
available by way of the television screen.
174
 THE USE OF TAPE
Tape will also store most of the programme material to be transmitted
over television networks. Hybrid s.h.f. and wired television networks,
will be extensively used in the future, and video tape machines will be
connected into such networks to facilitate the showing of items of local
interest, such as news and advertising, sporting events and even films.
Films, as we know them today, will disappear in favour of magnetic tape, and
the cinema industry will tie up closely with the electronics industry; cinemas
will close, and every household with a television set will be a potential box
office for the new cinema industry. The insertion of a coin into a small box by
the side of the television set will bring on to the domestic screen the very
latest film through the medium of magnetic tape. We have, in fact, only just
begun to exploit the potentialities of tape recording.

THE TAPE RECORDING PROCESS

Magnetic tape is a thin plastic material i in. in width and coated with
oxide of iron on one side; this is the "sensitive" side on which the recording
is magnetically impressed. With the highly developed processing of modern
tapes, it is often difficult to see which side of the tape actually contains the
oxide of iron. From both the mechanical and electronic points of view, this
is a good thing, because the smoothing of the coated surface as the result of
the "buffing" process during manufacture reduces wear on the recording,
playback and erase heads and ensures consistent mechanical contact of
the tape with the heads, which is desirable for extended high-frequency
response.
The recording head magnetizes each section of the tape as it passes over
the gap between the poles of the head. The recording head is, in fact, an
electromagnet which is energized by the signal current at the output of an
amplifier. Thus, if a person is speaking into a microphone connected at the
front of the amplifier, the recording head connected to the output is ener-
gized by the current which is varying in unison with the sound waves. This
means that the magnetic field across the poles of the recording head is also
varying both in polarity and in magnitude to the same pattern as the sound
waves. It is this varying magnetic field which is used to impress a magnetic
pattern on the coating of the tape as it passes over the gap of the recording
head.
The idea is illustrated in Fig. 9.1. Here the signal current is represented
by a sine wave which causes the polarity of the field across the recording
gap to reverse each half-cycle. Thus, as the tape passes steadily over the gap
small magnets are formed on the coating of the tape, as shown. The effect
would be the same if a more complex signal waveform, such as speech or
music, were used, but then the amplitude and the wavelength of the recorded
175
 THE PRACTICAL HI-Fl HANDBOOK
WAVELENGTH FIG. 9.1. As the record-
TAPE
i ing tape posses steadily
_,COATING
over the gap of the
DIRECTION OF TAPE - recording head, small
CORE magnets ore formed on
the coating, the charac-
ter of which conforms
to the signal waveform.
ALTERNATING CUP.RENT
IN COILS DUE TO
SIGNAL CURRENT
SHIM -~-i~K
WAVELENGTH
pattern would also be complex. The
---ort\Jo----
SIGNAL WAVEFORM
greater the amplitude of the signal wave-
form, the greater will be the current in
the recording coils and the greater the
strength of the magnetic field across the gap. Therefore the strength of the
magnets formed on the tape will also be greater.
The actual length of the magnets formed depends upon the wavelength
of the signal waveform. Two magnets go to make up one wavelength as the
result of the positive and negative half-cycles of the signal waveform (i.e.,
each half-cycle of signal produces one magnet). The length of the magnets
is also governed by the speed at which the tape passes over the gap-the
wavelength is equal to the speed of the tape divided by the frequency of the
signal current. Thus, at 7·5 in. per second, the wavelength representative of a
7·5 kc/s signal is 0·001 in., so the length of each magnet formed on the tape
is approximately half this value, or 0·0005 in. This is a very short magnet
indeed, but it is even shorter at higher frequencies.
We must now consider the hysteresis loop, sometimes referred to as the
cycle of magnetization. When an electromagnet is energized progressively
by the building-up of electric current in its winding there is built-up a mag-
netizing force around the coil or between the pole pieces of the magnet.
This magnetizing force is den0ted by the capital letter H, and if a ferrous
material (one which is affected by magnetism, such as iron or steel) is brought
within range of the field of this force it will have induced into it a magnetic
flux; i.e., the material will also be magnetized. Flux is denoted by the capital
letter B.
It follows that the greater the magnetizing force, the greater will be the
flux induced into the material and the stronger will be the magnetic symptom.
This action can be studied in more detail from the so-called hysteresis loop,
shown in Fig. 9.2. Here the magnetizing force, both positive and negative,
is represented by the horizontal line, while the flux is represented by the
vertical line. Let us assume that the ferrous material is in a magnetically
176
 THE USE OF TAPE
Fto. 9.2. The hysteresis loop.

neutral condition, and that the force H

is increased from zero (point a) in a
positive direction. This can be brought
about by increasing the current in the
electromagnet; if it is increased in the
opposite direction, then the value of H H- h•

will rise negatively.

Thus, as H increases so will the
value of B, and will trace the curve ab.
If now the force H is decreased (by
decreasing the current in the coil), the •
value of B will not fall to zero, but line
be will be traced and at zero H there will be an appreciable value of B in
the material as represented by point c on the B + line. This value of B is
representative of the residual magnetism held by the material. In other words,
the material has been magnetized by the magnetizing force which was applied.
However, if force H is now increased from zero in a negative direction,
the value of B will fall and line cd will be traced. At point d the material is
once again in a demagnetized state. The value of H which is required to
demagnetize the material is a measure of the coercivity of the material. As H
is further increased negatively, the line de is traced, and at e the material is
once again magnetized, but in reversed polarity. At point/His again at zero,
and then as H is increased positively once again, the remainder of the loop
fg and gb is traced. It should be noted that the initial path ah is traced only
when a material in a magnetically neutral condition is subjected to
force H.
From the foregoing, it should be clear how a tape is magnetized by the
changing magnetizing force across the gap of the recording head-bearing in
mind that the tape is passing the gap all the time.
Whilst the hysteresis loop shows the relationship of the flux induced
into a material to the force producing it over a complete cycle, it is not wholly
representative of the residual magnetism retained by the material owing to the
applied magnetizing force for a number of different loops, working from zero
H up to saturation, both in positive and negative values. (Saturation indicates
the point where an increase in H does not result in an increase in B.)
The residual magnetism retained by a material (usually referred to as
remanent induction and still denoted by B or sometimes Br) and the magnetiz-
ing force H required to produce it are plotted to form a transfer characteristic
or remanence curve. Such a curve is shown in Fig. 9.3, from which will be
seen that the remanent induction or intensity of magnetism impressed upon
177
 THE PRACTICAL HI-Fl HANDBOOK
111' • Fm. 9.3. The transfer characteristic of

I RECORDED
the remanence curve, showing how the
kink in the curve distorts the recorded
signal.
DISTORTED
SIGNAL
H• ------.~c......;........._.......,r-r---H •
the tape will not be proportional to
the force H or the current in the
APPLIED PURE recording head owing to the pro-
SIGNAL nounced curvature of the charac-
teristic.
Br• This "transfer distortion" can
be avoided by applying a bias in the
form of a high-frequency sine wave to the recording head along with the
actual signal. The frequency of the bias is usually in the region of 30-100 kc/s,
and therefore in the supersonic region and outside the range of hearing. The
bias signal is superimposed on the signal to be recorded and has the effect
of combating the kink in the centre of the transfer characteristic. How this
happens is somewhat complex, and there are several theories on the matter.
However, a working knowledge can be obtained from Fig. 9.4, which shows
the applied signal on which is superimposed the bias signal, and the resulting
distortion-free recorded signal. It appears that the applied signal is alternated
either side of the transfer characteristic because of the bias, and that an exact
replica of the applied signal is impressed in magnetic form upon the tape as
the result of an average of the remanent induction being "sampled" over the
two branches of the characteristic curve.

PLAYBACK
Having recorded a magnetic pattern on the tape as described, it is a
simple matter to re-convert the pattern back to electrical signals which can be
amplified and fed to a loudspeaker system, by passing the tape over a replay
head. The replay head is virtually
the same as the recording head, and
in most portable-type recorders the DISTORTION-
one head serves for both recording FREE RECX>RDEC
SIGNAL
and playback; the tape speed must
H•

FIG. 9.4. Showing how the application

of a high-frequency bias on the applied
signal has the effect of combating the
kink in the centre of the transfer curve BIAS SIGNAL SUPER·
IMPOSED ON APPi.JED
and gives a distortion-free recorded SIGNAL
signal. 81'•

178
 THE USE OF TAPE
be the same for both recording and playback if distortion is to be avoided.
As the tape passes over the replay head the magnetic pattern produces
corresponding variations of flux in the core, which in turn induces e.m.f.
variations in the windings matching the pattern of the original signal; these
variations in e.m.f. are applied to the grid of an amplifier, in the same way as
are the variations of e.m.f. produced by a pick-up or microphone.
The gap between the pole pieces of the replay head has a considerable
bearing on the response of the head to high frequencies, and in order to
secure optimum output voltage at high frequencies the length of the magnets
impressed upon the tape should not be much smaller than the size of the gap.
We have seen that two magnets represent one wavelength and that the speed
of the tape past the head as well as the frequency of the signal governs the
length of the recorded magnets. For example, if the tape speed is 7·5 in. per
second and the frequency of the recorded signal is 7,500 c/s, the wavelength
of the recorded signal will be 1/1000 in. (0·001 in.). This means that the length
of each recorded magnet is half this value, which is 0·0005 in. From theoretical
considerations, therefore, it would seem that in order to secure optimum
response at 7,500 c/s the gap size should be 0·0005 in.
In practice this is not strictly true, because there are other factors in-
volved. One is that the magnetic field across the gap due to the magnetized
tape tends to bulge outwards and cover a greater distance than that repre-
sented by the gap itself. This effect is known as "fringing", and tends to
reduce the high-frequency response for a gap size as computed above. The
effect is more apparent, and loss in high-frequency response becomes very
marked, if the tape is not making intimate contact with the head pole pieces.
Generally speaking, the maximum output with a gap size of 0·0005 in.
occurs in the region of 4-5 kc/sat a tape speed of 7·5 in. per second. This is
not the highest frequency which can be reproduced, however, since as the
frequency is increased above that representing optimum output, the output
does not suddenly fall to zero but falls off over several octaves, and when the
gap size is equal to the wavelength of the recorded signal (i.e., two magnets
in the gap) the output falls to zero owing to flux cancellation in the head.
At frequencies below that representing optimum output the output falls at
the rate of about 6 db per octave.
As a means of securing an improved high-frequency response, without
too great a tape speed, manufacturers are investigating the problems involved
in producing heads with very small gaps. Not so long ago, a gap size of
0·0005 in. was considered to be about the practical minimum. These days,
however, gaps smaller than 0·0002 in. are becoming commonplace. The
BBC's Vision Electronic Recording Apparatus (VERA) uses heads with gaps
of the order of 0·00002 in., and at a tape speed of 200 in. per second there is
only a 3 db fall in response at a frequency of 2·5 Mc/s. As a rough guide,
179
 THE PRACTICAL HI-Fl HANDBOOK
the effective gap size should be approximately 0·5 of the shortest half-
wavelength to be reproduced.
Mumetal, Permalloy and other high-permeability alloys are used for the
head pole pieces, and the gap is closed by a non-magnetic shim as a means
of keeping it clear of magnetic coating from the tape and of concentrating
the flux towards the tape. An additional gap is usually introduced diametri-
cally opposite the functional gap, whose purpose is to maintain a linear
relationship between the recording current and the flux at the gap. The rear
gap is also closed with a non-magnetic shim, and has a width approximately
ten times that of the functional gap.
It has already been mentioned that a common head is usually employed
for both recording and replay. Where a separate recording head is used,
however, the gap size is not so important as that on the replay head. Since the
recording does not occur until the tape section reaches the end of the gap, a
gap size of the order of0·OOOl-0·0015 in. is adequate.

ERASURE
One of the most attractive features of tape recording is that the material
recorded can be erased from the tape if it is not suitable or no longer required,
and the tape can be used over and over again almost indefinitely. The problem
is simply that of demagnetizing the tape. This is best done by passing the
tape through a strong alternating field; the field is produced across the gap of
an erase head and the winding is energized from the high-frequency oscillator
supplying the bias current. The construction of the erase head is slightly
different from that of the recording or replay head. The core material has a
higher saturation value, and a fairly wide gap in the region of 0·015 in. is
usually adopted. Some erase heads have twin gaps, or two separate erase
heads may be employed, as a means of ensuring complete erasure.
For successful erasure a high value of alternating current in the erase
head winding is necessary, and in order to help satisfy this condition the head
is often arranged to resonate at the frequency of the applied signal.
At one time home recorders employed a small permanent magnet
to erase an unrequired recording by magnetizing the tape fully in one
direction only. Although this simple method, by which the magnet is
brought into contact with the tape before it is presented to the recording
head, does result in complete elimination of the recording, distortion
and a poor signal-to-noise
ratio are inevitable when the

F10. 9.5. Dimensions of tape and

tracks.
180
 THE USE OF TAPE
tape is next brought into service, owing to the standing magnetism on the tape.
There are devices available which are capable of bulk erasure of a tape.
The whole tape is set up on the instrument, and the tape is subjected to a very
strong magnetic field for several seconds or more, while the spool is rotated
on a spindle. This is a very rapid and efficient method of erasure, is convenient
when it is required to remove a recording before the tape is re-issued, and
also ensures a very desirable high signal-to-noise ratio. A bulk erasure device
of this kind is marketed by Leevers-Rich Equipment, Ltd.

RECORDING TRACKS
In early models of home tape recorders almost the full width of the tape
was subjected to the recording flux and magnetized. This arrangement is still
adopted in professional equipment and machines employed for disk-dubbing
because of the ease of editing and the slightly improved quality obtained by
using the full width of the tape. With modern domestic tape recorders,
however, dual-track facilities are incorporated. Two separate recordings can
therefore be impressed side by side on one tape. The track dimensions for this
method are given in Fig. 9.5. Whilst these dimensions are almost standardized,
one or two very slight deviations are sometimes found, but these have little
or no adverse effect on the interchangeability of recorded tapes, provided the
direction of scanning of the tracks is as indicated by the arrows in Fig. 9.5,
that is with the coated side of the tape away from the observer.
Some machines employ two recording/replay heads, one for each track,
and the mechanism is arranged so that a button or switch changes from one
track to the other, whilst also reversing the direction of the tape travel. The
Simon Model SP4 tape recorder incorporates automatic tape reversal fo,
continuous recording or replay, without button-pressing or transposition of
spools. When one track runs out, the tape automatically reverses and the
other track is switched in.
With machines having a single recording/replay head, track change is
accomplished by reversing the spools and turning them over, so that the
unrecorded bottom half of the tape is scanned by the head.

TAPE DECKS
Generally speaking, the tape deck is a complete unit in itself, as a record
player is a complete unit in relation to a radiogram. There are one or two
variations in this respect, however, particularly with Continental equipment,
where the tape mechanism is an integral part of the recorder. Several makes
of British recorder have decks made by organizations specializing in this
field, such as Collaro, Truvox and Wearite. The recorders show many
variations in the style of cabinet, facilities provided, design of amplifier, type
of loudspeaker used, etc. However, to give some idea of the principles
181
 THE PRACTICAL HI-FI HANDBOOK
Fm. 9.6. General layout of tape deck.

involved, the layout of a typical

tape deck is illustrated in Fig. 9.6.
The tape deck must arrange
for a constant drive of the tape
past the various heads, whilst
maintaining control of the tape
on the spools. The actual drive RECOIIOING/ RELAY
HEAD
is obtained by means of a very
accurately machined capstan
which is driven at a constant speed against a pinch roller, the tape being
pressed between the two, as shown. The capstan is either driven by the motor
through a friction coupling or, in less elaborate machines, is mounted direct
on to the spindle of the motor. Whatever the method adopted, a weighty
flywheel is always employed on the capstan shaft to reduce any irregularity
in the bearings or motor, and thus assist in maintaining constant speed.
In some cases the tape speed is controlled by interchangeable capstans.
This is the case in the Truvox deck, where speeds of 3¾ in. per second and
7½ in. per second are available. The slow speed gives a playing time of 60
minutes per track for standard tape and 90 minutes per track for long-
playing tape, while the playing time is reduced by 50 per cent at 7! in. per
second. (That is for 7 in. diameter spools containing 1,200 ft. of standard
tape and 1,800 ft. of long playing tape, which is thinner than the standard.)
Various other arrangements are in use for speed change, including
stepped drive wheels, of similar arrangement to those incorporated in record
players, and control by switching the field windings in the motor (Grundig,
for example). Apart from the two speeds mentioned above, a speed of 15 in.
per second is also available on some models as a means of securing a really
first-class high-frequency response. A speed of l ·875 in. per second is some-
times used for the recording of material where quality of reproduction is not
a prime factor.
Whilst the tape is being driven past the heads the tape take-up spool is
slightly under load-sometimes through a mechanical or electromagnetic
clutch-to ensure that the recorded (or reproduced) tape is adequately
controlled. In some cases, the supply spool is also loaded lightly to avoid
spilling of the tape from the spool due to overdrive.
Facilities are available for fast rewind or wind-on. A separate motor or
motors are sometimes used in the more expensive equipment, while very
successful results are possible by employing for these functions the capstan
drive motor. The Truvox deck can re-spool a 1,200 ft. reel of tape in less than
182
 THE USE OF TAPE

FIG. 9.7. Col/aro Mk. Ill tape transcriptor.

one minute, using three B.T.H. shaded-pole motors. Two motors are
embodied in the Collaro deck, while the Grundig works very successfully
with one motor.
There are other refinements, and various combinations of switching and
mechanical functions, which need not be described here. It is now becoming
common practice to incorporate such things as "pause control" (for stopping
the transit of the tape past the heads and applying the breaks to the spools
whilst leaving all the switches and their mechanical functions in the selected
positions), "playing time indicator" (facilitating the location of any recorded
passage), and "instant track change", as already described.
The well-known Collaro Mk. III tapetranscriptor is illustrated in Fig. 9.7,
while the extensively used Truvox Mk. IV tape deck is shown in Fig. 9.8. Note
the three digit counters on both decks, which serve admirably as place
locators. The Collaro deck has facilities for instantaneous changes of track,
while the same facility is available on the Truvox deck by spool transposition,
183
 THE PRACTICAL HI-Fl HANDBOOK
FIG. 9.8. The Truvox
Mk. IV tape deck
with place locator.

FREQUENCY RESPONSE

If a tape record is made with an amplifier possessing a flat response.

and it is also replayed on a linear amplifier, the frequency /output curve of
the recording will be similar to that shown in Fig. 9.9 (a). Up to the point of
maximum response, the output rises steadily at the rate of about 6 db per
octave. This is because the voltage across the recording head, which is
essentially an inductive element fed through a resistor to provide a recording
current which is constant irrespective of frequency, doubles every time the
frequency is doubled.
Towards the peak of the response, however, the rate of rise tails off
slightly, and beyond maximum response it falls fairly quickly. This deviation
from the "natural" 6 db per octave rise is promoted by high-frequency losses
inherent in the system generally. For example, there is the effect of the gap,
the speed of the tape past the head, the coercivity of the tape and capacitive
losses in the amplifier proper. Also, the very short magnets which correspond
to high recorded frequencies tend to demagnetize themselves, though high-
coercivity tape is less prone to this trouble. A reduction of high-frequency loss
owing to this cause of some 8 db may be expected by the use of a good-
quality high-coercivity tape (it will be remembered that coercivity is a measure
of the ability of the tape to resist demagnetization).
184
 THE USE OF TAPE
It will be clear that the response relative to frequency is governed by the
speed at which the tape passes the heads. The curve given in Fig. 9.9 (a) is
for a speed of 7½ in. per second. From this curve clearly emerges the fact
that some form of equalization is called for on replay. In practice, equaliza-
tion is applied during the recording process as well as during replay. Whilst
recording, the recording head is invariably fed through a filter which provides
a treble lift as a means of combating some of the high-frequency losses. This
filter may be either a tuned device, including an inductance, or a relatively
simple resistance-capacitance network.
On replay there is a circuit which gives a rise in bass response and a
further circuit which gives a high-frequency lift to provide additional com-
pensation for losses in the replay head. Fortunately, there is some standardiza-
tion with regard to recording and reproducing characteristics. These are
dealt with in British Standard 2478:1954, which accepts the recommendations
of the Comite Consultatif International des Radio-communications (C .C.I. R.)
for programme interchange on tape. For tape speeds of 15 in. per second and
30 in. per second, the specified frequency-response curve represents a bass
rise equivalent to that secured by a series-connected resistor and capacitor
having a time-constant of 35 microseconds. For a tape speed of 7~ in. per
second a time-constant of 100 microseconds is recommended.

• 20

----- -----
+ 10
db
0
-10
1-----
(al '
-20

-30

+30

+25
" "-
"-.. 15 ANO JO IN PER SECOND
~
db
+20

+15
"'""--"-"- ... ~
+10

+S
7i IN PER SECOND - - -............
'-
r-,-..___
0 lbl
-5

NI 100 1000 10000

FREQUENCY c/s

FIG. 9.9. Frequency characteristics of tape recording.

)85
 R 250V
P220Y UO 3W 260V •

27K
R/P I 200/2l~yl
i
175~ 33K 195V SWm::H SWITCH

:;t;
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SOLEN• GANGED
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CONTROL\·' :c

-- l l sh ITI
JACK 2 330K
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50 ..,
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ALL SWITCHES SHOWN IN RECORD POSITION 22K

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RECORD/ REPLAY 2
HEAD -<:r-- OE.NOTES POWER PLUG PIN No
ALL VOLTAGE READINGS TAKEN WITH A
20000 OHM/VOLT METER

FIG. 9.10. Circuit diagram o.f the Truvox Type K tape-recorder amplifier.
 THE USE OF TAPE
The curves obtained by these time-constants, assuming an "ideal"
reproducing head, are shown in Fig. 9.9 (b). In practice, these time-constants
may have to be altered slightly to suit the particular head in use. It is usually
necessary to consider the equalized pre-amplifier and reproducing head as a
complete unit with regard to shaping the overall response. All tape records,
including "Stereosonic" versions, are recorded to suit the C.C.I.R. charac-
teristics, so it is as well for the recorder to maintain this replay response
within the tolerance of plus and minus 3 db, at least.

AMPLIFIERS
Fig. 9· IO shows the circuit of the Truvox Type K tape-recorder amplifier
which, apart from being used in certain Truvox tape recorders, can be
obtained as a separate unit if an enthusiast requires to make up his own
recorder, or if it is required to add another channel for the playing of
"Stereosonic" tape records.
It should first be noted that the various switches, representing record/
replay, are shown in the "record" position. The first stage, VI, serves on the
"record" position as a high-gain microphone amplifier. The microphone
signal is applied to the control grid of the valve, which is a low-noise pentode,
by way of jack I. It will be noted that this is a high-impedance input and
suitable for a crystal microphone or an electromagnetic type embodying an
impedance-matching transformer.
The amplified signal is fed to the grid of V2A (half of Mullard ECC83)
through the 0·05-mF coupling capacitor and 330k resistor. The resistor-
capacitor network between the anode and grid of V2A forms a parallel-T
feedback system which, being frequency-selective, gives a top lift to com-
pensate for high-frequency losses, as already described. The frequency-
compensated output from V2A is passed by way of the volume control-the
tone control being out of circuit on "record"-to a further amplifier V28
(the other section of the ECC83). This section also functions as the recording
output valve, the signal being fed through switch D, through a further
resistor-capacitor network for additional head compensation, to the record/
replay head through switch A.
The signal at the anode of V28 is also applied to a rectifier WX6, and
the d.c. voltage developed across the 4·7-megohm load resistor is applied to
the recording-level indicator valve V3, being brought into operation by
switch B.
During the recording process, valve V4 (EL84) serves as the high-
frequency oscillator. LI is the oscillator coil, which is resonated by the
parallel 0·002-mF capacitor. Switch I brings the oscillator into circuit by
completing the coupling between the grid and the anode. It will be seen that the
oscillator voltage is taken from the anode of the valve and is fed through a
187
 THE PRACTICAL HI-FI HANDBOOK
109V 200K 182V 10 K 190V IOK

0·1 IOOK
----.r--tif----wv---
MIC
INPUT

IM 2M
25
0·25

9A B -'W'--_J P~..f'R
200p
I IP C ~R rf
L-- --<r--.L-;;1/-L~- -'- _______ ----o-- I
lt£C0RD/ PLAYBACK HEADS

'" YS SA UT P/R
C 0-----------..-
!I,.
"L_g
6/S
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a ~ VI
l T P/R ~ 2
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HEAO

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!116<j> !113 2/S
116 21
o--......'..''_b&"o ,15 :ir.; 3!.--1 '

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0
"'3 ~

SWITCH CIRCUIT THE CONTACTS ARE SHOWN IN THE RELAXED (1 e STOP) POSITIONS

FIG. 9.11. Circuit diagram of the Sound Sales A20 tape recorder. Connexions shown
as dotted lines in the circuit diagram are simplifications of switch connexions, the true
connexions being given in the lower diagram.

capacitor of 100 pF to the recording/replay head by way of switch A. The

erase head is also energized by the oscillator current through the 0·01-mF
coupling capacitor and switch J. The bias voltage is fixed at 48 kc/s, and the
level of the signal is of the order of 150 volts at the erase head and 80 volts
at the recording head. Both heads are of the high-impedance type and thus
match directly into the circuits without transformers.
The amplifier is powered by means of a full-wave h.t. rectifier valve,
V5 (5Y3). The mains transformer carries two l.t. windings, one for the
heaters of the valves and the other for the heater of the rectifier. Although
there is no smoothing choke, a remarkably low level of hum (45 db down at
4 watts) is achieved by the use of four 40-mF electrolytic capacitors and
associated filter resistors.
188
 THE USE OF TAPE
212V

300K SOK

•01

V4
EM81 7

50
12K 47K 200K

ERASE HEADS

IOOOp

~
•01

EARTH OF r----U-----, EARTH OF

DECK ~ •• ~AMPLIFIER MOTORS

~
CIRCUIT SHOWN IN RECORD POSITION MAINS INPUT

The "record/replay" switch is actuated by a relay whose winding

-denoted R/P switch solenoid on the diagram-is energized by the instan-
taneous shorting together of the two wires labelled "brake" on the diagram.
This action trips the relay switch so that it returns to the "replay" position.
It is necessary to re-set the switch to the "record" position by pressing the
appropriate button on the amplifier control panel.
On the "replay" position the record/replay head is switched into the
grid circuit of VI, which now acts as the reproducing-head amplifier, by the
changing-over of switches F and A. Switch C alters the coupling response so
as to give a further lift of treble, switch H brings in the tone control and
removes the 0·01--mF capacitor from across the cathode resistor of V2B. The
oscillator is switched off by switch I and switch D disconnects the head
equalizer and couples the signal at the anode of V2B to the control grid of
V4 (V4 now acts as the sound output valve). Switch B disconnects the level
indicator valve, switch E brings in the output transformer and switch J
short-circuits the erase head which, of course, is not required on replay.
It will be seen that negative-voltage feedback is applied over two stages
from the output transformer to the cathode of V2B. Since the 0·01--mF
capacitor is disconnected from across the cathode resistor, the feedback is
not frequency-selective as it would be if the capacitor was left in circuit.
The output transformer provides for loudspeakers of either 3 ohms or
189
 THE PRACTICAL HI-Fl HANDBOOK
15 ohms impedance. The signal outlet at che anode of V2A by way of jack 3
permits monitoring of the recording or recorded signal. If required, the signal
at this point can be applied to the input circuits of a separate hi-fi amplifier.
The signal level at this point is approximately 0·5 volt at 500 ohms impedance.
Whilst it will usually be found necessary to employ VI as the microphone
amplifier during the recording process, the sensitivity at jack 1 being in the
region of 1-2 m V, a higher signal level should be applied by way of jack 2,
where the sensitivity is something like 500 m V; for example, a gramophone
pick-up signal or a radio signal can be introduced here. The Truvox radio
jack is useful for the recording of radio programmes, but this should be
inserted into jack I owing to the low output signal from the crystal detector.

SOUND TYPE A2O TAPE RECORDER

The circuit diagram in Fig. 9.11 of this recorder shows that it has much
in common with the Truvox circuit. Instead of a low-noise pentode, however,
two cascade-connected triodes VI and V2 (Mullard ECC83) provide the
necessary high gain for use as a microphone amplifier during recording and a
replay-head amplifier during replay, these functions being governed by the
setting of the changeover switches, as before.
The ECL80 valve is arranged so that the triode serves as recording
amplifier and the pentode as oscillator during the recording process, and as
voltage amplifier and output valve respectively during replay. The EM8l
acts as a recording-level indicator, the controlling signal being obtained from
the recording signal (in the anode of V3) and rectified by the metal rectifier
M3. There are two inputs, but these operate in the grid circuit of the first
triode.

ADDING A TAPE DECK

Many enthusiasts, already in possession of a good hi-fi amplifier, may
wish to add a tape deck without going to the expense of purchasing a com-
plete tape recorder. This can be done fairly easily by utilizing the existing
amplifier. However, it is not usually possible to connect the replay head
direct to one of the inputs of the amplifier, even though one may be marked
"tape input", owing to the lack of gain and need for special equalization.
From the replay point of view, an equalized-head amplifier is required, whose
equalization response can be switched to suit the various tape speeds in
accordance with the C.C.I.R. characteristics.
A single low-noise pentode, such as the Mullard EF86, can usually be
arranged to provide sufficient rise of signal from the replay head to drive a
control unit of most makes. The stage will incorporate an equalizing network,
preferably of the selective-feedback type already dealt with. Special attention
should be given to the layout of the components and to the common-negative
190
 THE USE OF TAPE
FIG. 9.12. Typical bias oscillator circuit.

connexions on the chassis, so as

to avoid circulating earth cur-
rents which become manifested RELAYro RECORD/
HEAO
...-ww-ff11--.
as hum from the loudspeaker.
The precise characteristics
of the equalization will depend ~~!~ASE ..._._ _ __,
to a large degree on the type of
head adopted, but manufac-
turers of tape decks and heads
invariably issue a suitable
circuit. Such an arrangement will permit the reproduction of tape records,
but if a facility for recording is required as well, a recording amplifier
and oscillator will also be necessary.
The reproducing-head amplifier can serve as the microphone amplifier
for recording. This will have sufficient gain, since the output from an average
reproducing head is little more than 1-3 mV, which is more or less the same
as the output from a microphone.
A recording current of the order of 100 microamps is usually required
and this can be obtained with little trouble from a simple triode stage, which
may be fed from the microphone amplifier. It will be necessary to have
provision on the amplifier to switch out the bass-lift equalizing and to intro-
duce a form of pre-emphasis. Alternatively, the amplifier can be switched to
have a linear response, and the pre-emphasis introduced in the network from
the recording output valve to the recording head. Additional pre-emphasis
can, if required, be introduced between the coupling from the microphone
amplifier and the recording output valve; this can take the form of a parallel-T
network.
The considerable amount of high-frequency power required for efficient
erasing can only be obtained from a fairly large power pentode or tetrode,
such as the 6V6 or EL84. Oscillators in general use employ either the series-
fed Hartley or the Colpitts circuit. A typical Hartley circuit is given in
Fig. 9.12. Here LI is the tapped oscillator winding, which i!- tuned within the
frequency range of 30-80 kc/s by the 500-pF shunt capacitor. L2 is the coupling
coil which feeds the oscillator current into the erase head through Cl and
into the record/replay head through C2 and RI. RI is variable so that the
optimum value of high-frequency bias can be selected for the type of tape
and the signal voltage employed.
For the purpose of erase the value of the bias in the erase head is not
critical, provided that the bias current is sufficient to secure complete erasure.
The current in the recording head is, however, much more critical. As the
191
 THE PRACTICAL HI-Fl HANDBOOK
bias current is increased so the response falls, but as the recording current
is increased so a greater level of bias current is required to alleviate disto1tion.
It is usually necessary to perform a series of tests in order to establish the
best setting for the bias level control. The aim should be to achieve best
signal-to-noise ratio consistent with the minimum of distortion and the best
high-frequency response.
A distorted bias signal will tend to impair the signal-to-noise ratio to a
large degree. The, bias oscillator should be capable of producing a pure sine
wave at full output, and to assist in this respect professional equipment often
uses a push-pull oscillator circuit.

SERVICING TAPE EQUIPMENT

Faults likely to develop in tape recorders are of two distinct types: there
are mechanical faults relating to the tape deck proper, and electrical faults
associated with the amplifiers, oscillator and various equalizing networks.
Faults of amplifiers-particularly those resulting in complete failure of
recording, reproducing or both-can be located with reasonable ease by
adopting the techniques described in Chapter 4.
It helps considerably if a circuit diagram of the defective instrument is
available, for then a study of the various stages will reveal the valves and
components which are common to both recording and reproducing, and if
the cause of failure of both services is being sought the components most
likely to be responsible can be examined in more detail.
For example, failure of VS in Fig. 9.11 will affect both recording and
reproducing, since this valve serves both as sound output on replay and bias
oscillator on record. The replay section will be totally dead, assuming that
VS has failed completely, but on the "record" position the signal-level
indicator will function normally. This may lead to a suspicion of trouble in
the sound amplifier or associated networks. If a trial on another machine of
the tape that has supposedly been recorded results in severe distortion owing
to the lack of recording bias, the diagnosis will be proved accurate. In practice,
it is usually the faults of a more subtle nature which cause difficulty, and these
can be of both electrical and mechanical origin.
A poor high-frequency response, particularly from tapes recorded on
another machine or from a tape record, is often caused by maladjustment of
the recording/replay head. It is essential that the angle that the gap of the
head makes with the direction of motion of the tape is the same beth on
record and replay. Whilst this angle will obviously remain consistent when
the tape is recorded and reproduced on the same machine, it is desirable to
know that a recording is going to be satisfactory when played on a different
machine.
For this reason the head is best adjusted for optimum high-frequency
192
 THE USE OF TAPE
response with a standard test tape or a tape record having plenty of "top".
Test records giving a tone of 10 kc/s or 12 kc/s are available for this purpose.
The azimuth adjusting screw in proximity to the head or head securing bracket
should be slowly turned first one way and then the other whilst the output
from the test tape is being observed on an output meter or a.c. voltmeter.
The point giving maximum output should be selected.
If the recording/replay head becomes slightly magnetized a marked
increase in background "hiss" on replay results. Theoretically, the head
should not become magnetized since a large-value time-constant is used in
the h.t. feed to the oscillator valve which causes a gradual decay of oscillator
current in the heads, and thus avoids surges which are likely to magnetize
the heads. However, in practice small current surges occur which promote
the trouble; there is also the possibility of the heads coming into contact with
a magnetized screwdriver or a magnetic field. Again, there is the possibility
that the head may be checked for continuity with an ohmmeter.
Whatever the cause, it is necessary to demagnetize the head to reduce
the "hiss" and to prevent a constant flux being impressed upon the tape. The
best way of securing this is by the use of a "defluxer". An ideal instrument
of this kind is the Wearite Defluxer (Fig. 9.13). This has a special shaped pole
piece which can easily be brought into contact with the head pole pieces.
The instrument is mains operated, and at the end removed from the pole
piece is a press-button control switch.
In operation, the pole piece is brought into contact with the head and
the button depressed. The pole piece is then gradually taken away from the
head, but the button should not be released until the head is outside the
influence of the field. In this way complete removal of residual magnetism
is achieved. The instrument is also useful for removing residual magnetism
from tape guides, pulleys, etc., which may fall in the path of the tape.
If the tape is not making close contact with the head pole pieces, the
effect is the same as an increase in the size of the gap, and a loss of the higher
frequencies results. If this trouble is suspected, the heads should be examined
for wear, and if very badly worn should be replaced. All traces of accumulated
particles of oxide coating from
the tape should be very carefully
removed from the head pole
pieces. A soft brush can be of
considerable assistance in this,
but cleaning fluid should not be
used unless specifically indicated

FIG. 9.13. The Wearite defluxer.

193
 THE PRACTICAL HI-Fl HANDBOOK
in the manufacturer's instructions. The tape guides and pulleys should be
examined for adjustment to ensure that the tape is being evenly transported
past the heads.
Excessive wow should lead to checking the pressure between the capstan
and the pressure roller, drag on the spools due to an incorrect clutch adjust-
ment, the regularity of the various drive faces (particularly the rubber
rollers and jockey pulleys), the lubrication of the bearings and the alignment
of the flywheel and motor spindle. Worn rubber or composition friction-
drives and oil on the surfaces of the drives are common causes of the trouble,
especially of the lower-frequency wow. Flutter, at a higher frequency, may
also be caused by an eccentric drive wheel, or by "snatching" of the tape
owing to irregular binding of the supply-spool bearing.
If a double-beam oscilloscope, an audio oscillator and a 1,000-c/s test
tape are available a useful test for wow can be made. The test tape should be
played on the recorder and the output signal monitored on one beam. The
other beam should be connected to the audio oscillator, which should be
adjusted for 1,000 c/s. Adjustments should be made to the controls of the
oscillator and oscilloscope until the two waveforms occur approximately in
step. Wow in the recorder will cause a regular movement back and forth of
the waveform from the recorder, while the waveform from the oscillator will
remain constant. The amount of shift relative to the stationary trace can
easily be observed and the percentage in speed variation ascertained. For
example, a shift of I c/s represents a wow content of O· I per cent.

RECORDING FROM RADIO

It will often be required to make a recording from a radio signal. One
way of doing this is to simply stand the recorder's microphone in front of the
receiver's loudspeaker. This is not a desirable method owing to the introduc-
tion of distortion from the receiver's output stage and loudspeaker; the
recording will also be coloured by the acoustics of the room.
Another method is to connect an input channel on the recorder to the
extension-loudspeaker sockets on the receiver. This also is undesirable
because the distortion in the receiver's output stage will be recorded. By far
Fm. 9.14. Method of
extracting a high-im-
,._-1t-.....,_.VIN..,...t"""""lt::::~ To HIGH IMPEDANCE pedance low-level signal
INPUT CHANNEL C'F from a receiver. (Not
RECORDER •
suuable for a.c./d.c. re-
RECEIVER'S IOOK
VOLUME ceivers, or where the
CONTROL chassis is connected to
RECEIVER'S
one side of the mains
CHASSIS supply.)
194
 THE USE OF TAPE

Fm. 9.15. Tape recorders and sound equipment installed for use at the Duke of
Edinburgh's Study Conference at Rhodes House, Oxford (1956).

the best method is to take a connexion from the detector circuit of the
receiver and feed the signal at this point into an input channel on the
recorder. The idea is shown in circuit form in Fig. 9.14.
To conclude this chapter a photograph is reproduced of a galaxy of tape
recorders and sound equipment (Fig. 9.15) which was employed at the Duke
of Edinburgh's Study Conference at Rhodes House, Oxford, in 1956. This
shows two Grundig Stenorette recorder dictating machines-a Grundig
TK820 and a Grundig TK819, both being high-quality home recorders-two
Pamphonic amplifiers, a B.S.R. amplifier, a Trix amplifier, an M.S.S. disk
recorder, a bulk tape erasure, and sundry other items of equipment. In the
centre, on the upper shelf, will be seen a microphone mixer unit which was
designed by the author to control the various signal inputs from the conference
microphones and BBC sound channels.
Tape recordings of every word of the conference were produced and
facilities were available for the immediate production of disk recordings
from the master tapes-which in overall length represented some 20 miles!
The Stenorette dictation recorders aided with the transcription of all the
principal speeches. All these facilities were provided by the Electronics
Division of Lowe and Oliver, Ltd., of Oxford, to whom the author is indebted
for permission to publish this photograph and also Fig. 5.24.
195
 CHAPTER 10

Stereophony

THE endeavours of hi-fi enthusiasts, designers and manufac-

turers of hi-fi equipment have resulted in a single-channel, or monaural,
reproducing system of extremely high standard. Harmonic distortion has been
cut to less than I per cent, the frequency response has been extended well
beyond the audible range as a means of improving the transient performance,
while the dynamic range and power-handling capabilities of modem hi-fi equip-
ment leave little to be desired. Nevertheless, there is something lacking!
With a monaural system, the sound which is picked up by the micro-
phone in a concert hall, for example, represents a sample of the sound
pressure which exists solely at the position of the microphone, and this sound
eventually arrives at the two ears of the listener at the reproducing end by
way of the loudspeaker. In other words, there is a complete, single channel
between the sound source and the listener, and this is so irrespective of
whether the connexion between the microphone and the loudspeaker consists
of a radio link, directly-connected cables or a recording system of some kind.
The idea is illustrated in Fig. I0.1.
This arrangement does not fit in with
things as they are in reality, for at a
concert performance the sound from the
various instruments embraces a wide area
in front of the listener and, even without
the use of his eyes, he finds no difficulty
in establishing the position of the instru-
MICROPHONE mentalists. The reason for this is that both
RJD10 LINK, ears collect the sounds coming from the
DIRECTLY CON- h t
NEC TED CABLES OR ore es ra a
t various
. ang Ies. The b ram
.
RECORDING SYSTEM compares several factors of the sounds

!LOUDSPEAKER
n'
0 LISTENER
FIG. 10.1. The monaural system of sound
reproduction.
196
 STEREOPHONY
FIG. 10.2. The stereophonic system of ORCHESTRA

sound reproduction. ,::::::::::3 c:::::J

~c::JCJ
picked up by the two ears, and from the
information thus abstracted brings into MICRO-
i !
PHONES
operation a form of sound location
TWO-CHANNEL
sense. Apart from the relative intensity RADIO LINK DIR-
of the sounds, this sound location ECTLY CONNECTED
CABLES OR TWO-
function is dependent also upon the CHANNEL RECORD-
ING SYSTEM
relative phase of the sounds at the two LOUD-
ears.lt will be appreciated that there is a SPEAKERS
time difference between the sounds, and
this may or may not be the same thing
as a phase difference; the former term, 0
LISTENER

however, applies essentially to sounds

of a random and transient nature, while the latter applies to repetitive sound
waves.
Sounds originating at the extreme right and left of a listener are dealt
with by the right and left ears respectively, while sounds originating between
these two points have their positions fixed (so far as the brain is concerned)
by one ear picking up more sound than the other and by comparison of
phase. It will be clear from this that a single-channel system cannot promote
operation of the sound location sense, and that as a result of this the illusion
of "perspective" and "spaciousness" of an orchestra is lost.
The problem is not solved by the use of two or more microphones
connected in series or parallel at the studio end and two or more loudspeakers
similarly connected at the reproducing end, since the individual signals,
though they may differ in phase and amplitude, lose their separate identity
in the single-channel network. With such a system, the sound simply appears
to come from the loudspeaker nearest to the listener or from the loudspeaker
which is radiating the greatest volume of sound; true perspective is not given
because all the loudspeakers reproduce in proportion the same sound. Two
or more loudspeakers in a single-channel system can help considerably,
however, in spreading the sound over a large area, provided that due attention
is given to their phasing.

THE BINAURAL SYSTEM

Correction of the loss of spaciousness and perspective is possible by the

use of two isolated channels, as shown in Fig. 10.2. The greater the number
of channels, the better the stereophonic effect, but economic considerations
limit the number of channels for domestic use to two (the cinema industry
197
 THE PRACTICAL HI-Fl HANDBOOK
is able to afford multi-channel systems). Fig. 10.2 shows that, in effect, the
listener's two ears have been extended into the studio or concert hall where
they are represented by microphones. Because the differences in the character
of the sound picked up by the microphones are conveyed to the loudspeakers,
the sound-location sense of the listener is brought into action, and he is given
the impression of the spaciousness and perspective of the orchestra as if
he were actually in the concert hall.
Consider a two-channel system with the microphones placed about 6 ft.
apart, in front of which are placed about six people in line. If the loudspeakers
are similarly placed in the listening room with regard to separation and
channel, and the people in line at the front of the microphones are instructed
to clap their hands in tum, starting with, say, the person nearest the right-
hand microphone, the listener at the loudspeakers will obtain the impression
of the sound starting at the right-hand loudspeaker, moving across between
the loudspeakers and finishing at the left-hand loudspeaker.
This is a stereophonic trick which can be used with great success in
demonstrating stereophonic sound reproduction; usually, however, some-
thing more elaborate than hand-clapping is featured-an express train
rushing through a station, a game of table tennis or something which illus-
trates vivid movement. The sound source will appear to be in the centre
between the two loudspeakers when the outputs are balanced. An announcer
standing in the centre between the two microphones would give this effect.
If he moved towards the left-hand microphone the sound would increase
from the left-hand loudspeaker and decrease from the right-hand one; the
converse effect would occur if he moved towards the other microphone.
This impression of movement, coupled with phase-comparison of the
sound from the loudspeakers, produces a two-channel stereophonic effect of
remarkably high standard. The illusion is somewhat enhanced by the
reverberant sounds also being put into their correct perspective, so that they
do not appear to come from exactly the same place as the original sound, as
is the case with a single-channel system.
Once stereophony has been experienced, it is extremely difficult to settle
down again to single-channel listening-a big attribute of stereophony is the
marked reduction in listening fatigue. The placing of the loudspeakers, their
phasing and the balance of sound are important factors which have to be
considered with the greatest care if the best results are to be secured.

MICROPHONE PLACING
The use of fairly widely spaced microphones, as shown in Fig. l 0.2, is
not always to be recommended. This is because the space between the two
loudspeakers may not be adequately balanced in relation to the level of
sound from them. The sound may appear to be concentrated either side of
198
 STEREO PHONY
the listener, with a distinct weakness in the centre. This may prove distressing
when the sound source is situated approximately in the centre between the
two microphones, since then the sound may appear to jump from one
loudspeaker to the other at the listening end, particularly if the sound source
happens to be someone moving about.
One method of overcoming this effect, developed by Philips, made use
of an artificial head with microphones instead of ears. The masking effect of
the head was relied upon to provide the amplitude ancl phase differences
of the sound signals passed over the two channels. This scheme provides a
stereophonic illusion at the higher frequencies only, since the masking given
by the head reduces in efficiency as the frequency is decreased and as the
wavelength of the sound becomes comparable with the head dimensions.
A system which is in current operation by E.M.I. for the production
of their "Stereosonic" tape records calls for the microphones to be situated
as close together as possible, one above the other. Use is made of pressure-
gradient microphones orientated so that their directions of maximum pick-up
are at right-angles, and positioned in relation to the centre of the sound
source so that the maximum pick-up axis of each microphone falls at an angle
of45 deg.
Another scheme which has received some attention both in America
and Great Britain is the placing of two microphones, one above the other,
over the orchestra. The lower microphone is arranged to respond to the
direct sounds, \\<hile the upper one is shielded from the direct sounds, but
responds to reverberant and indirect sounds. It has been claimed that
this method enhances the "presence" and atmosphere of an orchestral
reproduction.

LOUDSPEAKER PLACING
The subject of loudspeaker positioning is a cont1oversial one, and also
rather a problem if the most desirable listening position technically is not to
conflict with day-to-day household functions.
The arrangement shown in Fig. 10.3 has much to commend it, even

SPK.1 SPK.2 SPK.I SPK.2

(Left) FIG. 10.3. This

method of loud-
speaker positioning
has much to com-
mend it. (Right) FIG.
10.4. Another form
of loudspeaker set-
up.
199
 THE PRACTICAL HI-Fl HANDBOOK
though the area of stereophonic effect may be somewhat limited. The point
of optimum effect is shown at X, but movement from this ideal position
disturbs the balance of sound and gives the impression of movement of the
sound source. Within the stereo zone one can walk around without unduly
disturbing the effect of balance. This is because if one moves from, say, the
centre of the stereo zone to the left the resulting increase in distance from
loudspeaker 2 is offset by moving into its beam of greater sound level, while
the decrease in distance from loudspeaker I is corrected by moving away
from its beam of greater sound level. This effect was investigated in some
detail by G.E.C. engineers during the course of their experiments in stereo-
phony. (G.E.C. gave its first public demonstration of stereophonic sound
reproduction at the National Radio Show in 1951, magnetic tape being the
recording medium used at that time.)
The precedence effect also has some bearing on the balance and apparent
location of the sound source, relating sound intensity with relative time of
arrival of the sound at the two ears. For example, if the same sound is
radiated from two sources, the brain senses that the sound comes from the
source whose sound reaches the ears first. However, this effect can be balanced
by increasing the intensity of the delayed sound accordingly.
The set-up shown in Fig. 10.4 also serves reasonably well, particularly
in smaller rooms. Reflections from walls and furniture also disturb the stereo
balance when listening under domestic conditions. In order to combat this
trouble it is often necessary to experiment with the placing of the loud-
speakers, and best results may be secured with the loudspeakers arranged in
some odd pattern which appears to have no theoretical validity. There should
be no obstructions between the loudspeakers and the listener, however, and
in order to satisfy this condition it sometimes helps to mount the loud-
speakers above the level of the furniture.

LOUDSPEAKERS FOR STEREO

Experiments have been carried out with several makes of loudspeaker
and enclosure in different combinations, and in all cases the stereophonic
effect was secured once the sound levels were correctly balanced. In one test
a small battery portable receiver tuned to the BBC Home Service served as
the left-hand channel and a table television receiver as the right-hand channel
in connexion with the BBC's stereophonic experiments. The effect was truly
astounding; the inherent distortion of the non-hi-fl receivers appeared to
disappear, and the reproduction resulting from the two unmatched loud-
speakers was definitely of hi-fi quality.
This would appear to bear out the opinions of other workers in the field:
in fact, it has been suggested that a two-channel stereo system with a response
good to 6 kc/s sounds better than a single-channel system with a response
200
 STEREOPHONY
extended to 15 kc/s. However, the best results are undoubtedly gained by the
use of two hi-fi channels with matched loudspeakers and amplifiers. The
stereo effect is most marked in the middle and higher range of frequencies,
and at these frequencies it is desirable for the loudspeakers to have omni-
directional characteristics. Ralph West ("Hi-Fi Year Book", I 958) suggests
the use of reflecting cones serving as high-frequency diffusers, with the loud-
speaker enclosure preferably lying on its back. This idea has been tried, and
works admirably.
Loudspeaker phasing is not as critical in the author's opinion as is made
out by some authorities. Anti-phase conditions undoubtedly result in a loss
of heavy bass, as would be expected, but the overall stereo effect is not severely
affected, though there is a slight unevenness of the movement of the virtual
sound source between the loudspeakers. Ralph West says "out-of-phase
usually produces three sources-ahead, and one at each loudspeaker".
What can be most disconcerting is unbalance of hum levels from the
loudspeakers. During the course of recotding the BBC's stereo experiments
by way ofV.H.F.-F.M. channels, one channel suffered badly from modulation
hum due to the close proximity of e.h.t. cables. On playback (the stereo was
recorded remarkably well) the hum appeared to be removed from the actual
programme material and to exist close to one ear of the listener, that nearest
the humming loudspeaker.

EQUIPMENT REQUIRED FOR STEREO

In order to create the stereo effect in the home it is necessary to employ
two completely separate hi-fi systems. Starting from the listener and working
backwards towards the signal source, we require two loudspeaker systems,
two power amplifiers to drive them and two pre-amplifiers or control units
to feed the power amplifiers. Then there is the most important component
-the two separate signals representative of the programme material.
For use in the home, the programme signals will be derived from either
tape or disk records, and it will be shown later how the two signals are
maintained separately on a single length of tape and on a common groove of
a disk record. It is also possible to utilize two radio channels, one to carry
the right-hand signal and the other to carry the left-hand signal. This method
has been used on several occasions by the BBC in their experimental stereo-
phonic broadcasts. The television sound transmitters as well as the medium-
wave and V.H.F.-F.M. transmitters have been used on these tests in various
combinations so as to allow the maximum number of listeners to participate.
It is not the intention of the BBC to make a permanent feature of
stereophonic broadcasts at the present time, but radio-derived stereophonic
sound is available to listeners in New York by switching on both radio and
television receivers at the same time.
201
 THE PRACTICAL HI-FI HANDBOOK
Fm. 10.5. Pamphonic Model SI
loudspeaker, specially designed/or use
in pairs with the Pamphonic stereo
amplifier Model 3,000.

With regard to loudspeakers

for stereophonic applications, we
have already seen that these need
not be accurately matched, or
even of the same type, to secure
good-quality stereophony. Ob-
viously, the better the loud-
speaker systems, the better the quality of reproduction, but since difficulty
may be experienced in the housing of two large reflex enclosures in an average
living-room, it is likely that miniature reflex enclosures suitable for table
mounting or small console versions will become popular as stereophonic
systems become more widely used, and several manufacturers are concentra-
ting along this line. Fig. 10.5 shows the Pamphonic Model Sl stereo loud-
speaker, which uses a concentric cone unit which is adequately loaded by the
cabinet, and is specially designed for use in pairs with the Pamphonic Model
3,000 stereo amplifier.
Dual-channel, or stereophonic, amplifiers complete are now available,
and these represent an ideal choice for the enthusiast just entering the field
of hi-fi. Those already in possession of a single-channel hi-fi system are
catered for, however, as stereo control units are also available for handling
the two programme signals, derived from radio, tape or disk, and passing
them on to two (preferably matched) power amplifiers.

STEREO FROM DISK

There have been several attempts at producing stereo disk records, but
the system which is now in use dates back to A. D. Blumlein's experiments
around 1929, when a scheme for the simultaneous recording of two separate
sound tracks in a common groove was evolved. The idea was patented in
1931, but at that time it did not represent a commercial proposition and was
shelved. It was not until 1957 that the scheme was again brought to the notice
of the public: it was demonstrated at the London Audio Fair and the
B.S.R.A. exhibition by A. R. Sugden and in America by London Records (a
Decca associate) and Westrex. Its re-introduction coincided with the growing
interest in stereo reproduction, fostered by stereo tape records, and the
advanced state of development of the gramophone industry as a whole.
The system of stereophonic disk recording as first expounded by
202
 STEREOPHONY
Blumlein and now in current use requires only a single recording stylus and
recording head to produce recordings of both stereo channels in a common
groove, and a single pick-up with only one stylus to reproduce both channels
of the recording.
It will be remembered that a disk record can be made in two ways: the
conventional method whereby the modulation is imparted due to the
recording stylus oscillating to-and-fro in sympathy with the applied
modulation, and the hill-and-dale process whereby the modulation affects
the depth of the cut and the groove remains straight with no side-to-side
curves. Basically, stereo records are produced by combining both of these
methods, so that one channel modulates the groove laterally, as with an
ordinary record, and the other channel modulates the same groove vertically.
The stereo recording head is therefore arranged to have two electromagnetic
systems so that when energized by the applied signal one causes the recording
stylus to move laterally, while the other causes it to move vertically; the
movement of the stylus is thus in two planes, and corresponds to the complex
modulation pattern of the two stereo channels.
Similarly, the pick-up has two generating systems which respond
individually to the vertical undulations and the lateral displacement of the
recorded groove. The voltages generated by the two systems correspond to
the signals in the two channels and, since the pick-up is so designed that there
is very little interaction between the two systems, connexion can be made
direct to the appropriate amplifiers in the usual manner.
The patents of A. D. Blumlein allowed not only for the lateral-vertical
system of recording two channels in a common groove, but also for an
arrangement known as the 45/45 system. Whilst the modulation is imparted
into the groove by a slightly different method to that already described, the
45/45 system gives almost comparable results. With due attention given to
the phasing of the two electromagnetic systems in the recording head and of
the generating systems in the pick-up, either system can be used for recording
or reproduction on the same type of equipment. With the 45/45 system,
which is the one adopted, the two stereo channels are recorded at an angle of
45 deg. to the surface of the disk, and each channel is cut towards the opposite
wall of the groove, thus producing a single complex groove.
Let us investigate how a stereo record is cut, so as to secure a better
understanding of the 45/45 system. In Fig. l0.6 is shown, very much simplified,
the essential elements of an electromagnetic stereo recording head. This may
be of the moving-iron, moving-coil or even the crystal type, but it would
appear that the moving-coil type has much in its favour. The electromagnetic
type has two coils, one for each of the two stereo channels, which are denoted
A and Bin Fig. l0.6. Moving in the coils (or coupled to them in the case of
a moving-coil system) are two armature links C and D which are coupled
203
 THE PRACTICAL HI-FI HANDBOOK
FIG. l0.6. Simplified diagram of the
C 0
essential elements of an electromagnetic
stereo recording head.

to the stylus mounting shown at E.

It will be seen that the armature
,s• E ,s,• links diverge from the stylus mount-
__ f -
RECORDING
r- -, _t_ __ - -
I...._: DATUM LINES
ing at an angle of 90 deg. and that
STYLUS I each coil when energized would
promote a 45-deg. diagonal thrust
on the armature links in relation to the stylus tip or surface of the record.
Hence the term 45/45.
In Fig. 10.6 the stylus is shown at rest in relation to the dotted datum
lines, that is when neither coil is energized. The diagrams in Fig. 10. 7 illus-
trate how the stylus moves or oscillates under various conditions. Let us take
the case first of coil A (Fig. 10.6) being energized by an alternating current
of, say, pure waveform. Armature link C will oscillate up and down in the
coil in sympathy with the applied signal, rather like an engine piston, but
very much faster, and the stylus tip will follow the arrowed line as shown in
Fig. IO. 7 (a). Similarly, if the signal is removed from coil A and applied to
coil B, the movement of the stylus will follow the arrowed line in diagram (b),
this being opposite to that in (a).
However, when both of the coils are energized simultaneously the stylus
will follow a path governed by the relative phase of the two signals. For
example, if the signals are in phase, then both armature links will move up and
down in the coils together and, provided the signals are equal and the coils
are balanced, an up-and-down movement of the stylus will result, as shown by
the arrowed line in diagram (c), which is comparable to hill-and-dale record-
ing. If the signals are in anti-phase, one armature link will move up while the
other moves down. This will result in the stylus following the arrowed line in
diagram (d), which is comparable to ordinary lateral recording.
Now, ifit is borne in mind that while the stylus is oscillating in the various
modes described a groove is being cut, it will be realized that the impressed
modulation will be a combination of lateral and hill-and-dale, the same as
that produced by the lateral-vertical system of stereo recording. It will be
understood, therefore, that the two systems are basically identical. It comes
down to a matter of phasing of the two signals; the inclusion of a simple sum
and difference network in the recording head or pick-up circuit will permit the
pick-up or recording head of one system to operate with a recording made by
the other. It should be noted, however, that the 45/45 system has been
accepted as an international standard.
The recommendations in this connexion were withheld from publication
204
 STEREOPHONY
I I

-- j: ___ ----'t'_____ ___ Jl: ___ _

I

-- ~ - - -
I I
I I I I
I I I I
(a) ( b) (cl ldl

FIG. 10.7. The arrowed lines indicate how the recording stylus oscillates under various
conditions: (a) coil A (Fig. 10.6) energized; (b) coil B energized; (c) both coils
energized with in-phase signals; (d) both coils energized with anti-phase signals.

for a time by the European record manufacturers, who were represented at

the Ziirich Conference in November, 1957, until it was perfectly clear that
these harmonized with the views and recommendations of the Record
Industry Association of America. There has been close co-operation between
the recording companies regarding the standards for stereo disk records, and
no doubt this has helped in the rapid graduation of the stereo disk from the
laboratory, where it has been available for quite a long time, to the open
market.
With regard to other record parameters, such as the diameter of the
disk, speed and recording characteristics, the recommendations were that
they should follow wherever possible the existing standards for microgroove
records laid down by the International Electro-Technical Commission.
Thus it seems unlikely that a state of chaos will reign in relation to stereo
disks as it did at one time with regard to the parameters of monaural disks.
With regard to right and left channels-as defined by the loudspeaker
which supplies the sound to a listener in front of the loudspeakers-the
standard is that modulation normal to the surface of the groove wall which
faces the axis of the disk should supply the right-hand channel, while the
modulation on the groove wall which faces away from the axis of the disk
should supply the left-hand channel. This is really not very important from
the user's point of view, since it is usually a simple matter to change the
position of the loudspeakers or change over the pick-up connexions to
ensure that the right-hand modulation actuates the right-hand loud-
speaker.
Further, a maximum radius of 5 microns is recommended for the bottom
of the groove, but the included angle remains the same as that of single-
channel microgroove disks. However, in order to reduce pinch effect and
tracing distortion, the radius of the tip of the reproducing stylus requires to
be reduced to the region of 0·0005 in.; the recommended limits are 12·5
microns minimum and 15 microns maximum.
205
 THE PRACTICAL HI-FI HANDBOOK
COMPATIBILITY
There is another important standard relating to the phasing of the two
channels: it is stated that lateral movement of the reproducing stylus shall
provide equal, in-phase, acoustical signals from the loudspeakers. From this
it follows that pick-ups designed for stereo reproduction are suitable also
for the playing of normal single-channel microgroove records. This mode of
phasing also results in a theoretically "compatible" record, which will thus
provide a fully balanced single-channel signal when played with a single-
channel pick-up.
However, at the time of writing there is no monaural pick-up available
with sufficiently small mechanical impedance to the vertical movement of the
stylus to permit this concept of compatibility being exploited. Attempted
playing of a stereo disk with even the best of pick-ups currently available will,
without a doubt, wipe the stereo from the groove.
Apart from the high mechanical impedance of the vertical movement of
the stylus, it must also be remembered that single-channel pick-ups are fitted
with a larger stylus tip than is recommended for stereo disks.
This question of "phasing" the two stereo channels to secure a
"compatible" record is worth further investigation. Let us look again at Fig.
10.6. Here we see that when the signals applied to coils A and Bare in phase,
the stylus moves up-and-down as shown at Fig. 10.7 (c). This does not
follow the recommended standard which, for the in-phase condition, requires
the stylus to move parallel with the surface of the disk (i.e., laterally). This,
however, is easily remedied by changing the phase of the signal in one of the
channels so that it is shifted by 180 deg. Recording companies do, in fact,
phase the two channels in this way for reasons of compatibility, as explained
above.
Assuming in-phase signals from the stereo microphone system at the
studio, which, for example, would be obtained by an announcer speaking in
a position such as to provide sound waves of equal pressure and phase at the
microphones, the signals at the stereo recording head are applied to the two
electromagnetic systems in the manner which causes lateral modulation of
the groove.
On playing the record, the phasing of the generating systems of the
pick-up must also match, so that the corresponding lateral vibrations of the
reproducing stylus cause in-phase signals at the two loudspeakers. This
particular condition is, in effect, the same as that obtained by running two
correctly phased loudspeakers on a single channel, which would be expected
in view of the lateral modulation of the groove.
This illustrates the importance of maintaining correct phasing throughout
the system, from the microphones to the loudspeakers. Only when this
condition exists will the true stereophonic illusion be fully realized. The
206
 STEREOPHONY
actual phasing of the two channels, from the aspect of the recording head
serves essentially to modify the mechanical operation of the stylus to provide
a compatible record-the phase of one channel is purposely reversed here, it
will be remembered, to provide a lateral movement of the stylus for equal
in-phase signals. This method of phasing also helps to reduce the tracing
distortion and decreases the demands in relation to the vertical movement of
the reproducing stylus.

STEREO PICK-UPS

While two stereo signals are required to cause complex vibrations of the
recording stylus in accord with the waveforms of the signals, the vibrations
of the reproducing stylus as it traces the complex groove produce signal
voltages in the two generating systems of the pick-up of similar relative phase
and level as those applied to the recording head during the recording process.
Clearly, if the construction of the pick-up is similar to that of the
recording head then the movement of the armatures or coils in the pick-up
will follow the same movement as the armatures or coils in the recording
head, for a given mode of vibration of the stylus. Thus, the voltage produced
in each generating system of the pick-up will, within limits, duplicate the
voltage applied to each section of the recording head, through the medium
of the complex stereo groove.
In practice there is some slight interaction, or cross-talk, between
channels, caused by undesirable resonances in the pick-up and limitations
of the recording head and the recording itself. The desirable minimum of
cross-talk has been reckoned at 25 db: however, this may be rather an
optimistic value, and it has been suggested that the total value of cross-talk
may be as high as 10 db at high frequencies (the effect is more troublesome
as the frequency is raised). While there are a number of unknown factors,
tests indicate that a cross-talk value as high as 10 db has little adverse effect
on the stereo illusion.
Stereo pick-ups range from crystal and moving-coil to balanced-
armature and variable-reluctance types. With its relatively high output
voltage in relation to tracking weight, the crystal-cartridge has much to
commend it, even though its top response may not quite reach the standard
of low-level electromagnetic types.
The stereo crystal-cartridge element is formed of two slabs of crystal,
each with its own electrodes, cemented together to form a "bimorph". The
crystals are cut in such a manner that outputs are generated when one is
subjected to a torque and the other to a flexure, and they are known as the
"twister" and "bender" respectively. The vertical and lateral modulations
of the groove are transmitted to the two crystals through a form of mechanical
coupling so that an output is obtained from the bender crystal as the result
207
 THE PRACTICAL HI-Fl HANDBOOK
of vertical movement, and an output is obtained from the twister crystal as
the result of a lateral movement.
There is another arrangement whereby the crystals, instead of being
cemented together, are mounted so that their faces form a 90-deg. vee shape.
Stylus movement is mechanically coupled to the crystals, but in this case it is
possible to use two twister crystals, which require less stylus energy to provide
an output-which is desirable in view of the improved high-frequency
response.
As with crystal pick-ups, electromagnetic pick-ups for stereo must also
have two sensing elements operated from a common stylus. There are always
two coils, therefore, and these are invariably wound on a common magnetic
circuit. The moving armature is arranged in such a manner that stylus
movement in one direction induces an e.m.f. in one coil, while movement in
the other direction induces an e.m.f. in the other coil. Two output voltages
are thus obtained and, provided the phasing of the coils is correct, the
relative phase of the signals will be as required for true stereo repro-
duction.

STYLUS

Owing to the very small tip radius of the stereo reproducing stylus, the
rate of wear of the point is faster than for the 0·001-in. microgroove stylus.
For this reason, most stereo pick-ups are fitted with diamond styli as
standard. This puts up the cost, but it is really essential when it is considered
that a 0·0005-in. sapphire stylus tracking at 7 grammes does not last much
longer than about five hours! Most manufacturers aim at about 3-4 grammes
tracking weight, which increases the life of the sapphire stylus to about
70 hours; the diamond lasts some 20 to 30 times longer than this.
The reader should note that both pick-up channels require equalization
to conform to the standards given in Chapter 7.

STEREO FROM TAPE

Scereo tape records have been available for a number of years both in
Britain and America, having been introduced in this country by E.M.I., Ltd.
This company's "Stereosonic" tape records are becoming more popular as
more people are buying tape recorders and converting them for stereo
operation. Unlike the disk, however, which enjoys a considerably greater
degree of popularity, stereo tape records are still mainly of interest to the
specialist.
There is undoubtedly a great future for magnetic tape in stereophony,
but from the point of view of the general public it seems likely that the disk
will hold the field for some time to come. Of course, the disk has long been
established, whereas tape is a relative newcomer. Theie is also the question
208
 STEREO PHONY
of cost. The enthusiast already in possession
of a single-channel record-playing system
must buy a second amplifier and loud-
speaker and also a stereo pick-up in order
to exploit stereo disks to the full. If he
decides to start on tape, however, he will
require a complete set of equipment, in-
cluding a stereo tape deck (his original
loudspeaker will do for one channel, of
course).
With tape records, the two stereo
channels are recorded one above the other FIG. 10.8. The Truvox stereo
on standard ¼-in. tape. They are recorded head.
in conformity with the C.C.I.R. recom-
mended characteristic at a speed of 7½ in.
per second. E.M.I. "Stereosonic" tapes, which are the only stereo tapes
available in this country at the time of writing, are recorded with the
recording heads in line across the tape. This makes it necessary to employ a
stereophonic head for replay whose gaps are in line.
A stereophonic head of this kind is illustrated in Fig. 10.8. This is the
well-known Truvox head, which can easily be fitted to the Truvox Mark III
and Mark IV tape decks with very little trouble. By the inclusion of a Truvox
Type K amplifier to provide a second channel, a Truvox R2 recorder or any
Truvox tape deck can be used for the playing of "Stereosonic" tape records.
It is also possible to produce one's own stereophonic tape recordings, but a
high level of studio technique is necessary for good results, "a good half-
track recording being preferable to a poor stereophonic recording", accord-
ing to Truvox.
Other machines could be adapted to cater for the Truvox stereo head,
and to help experimenters in this respect, details of the head are given. The
gap is 0·00025 in. beryllium copper; output voltage 1-3 mV; impedance
approximately 50,000 ohms at JO kc/s; frequency response attainable with a
suitable amplifier 50 to 15,000 c/s; cross-talk better than 45 db; bias
for recording 120 V r.m.s. approximately; recording current O· l mA
approximately.
If it is required to play stereo tape records and one is already in possession
of a tape deck and a single-channel hi-fl outfit, there is usually no need to go
to the expense of adding a second complete reproducing/recording amplifier.
One scheme which can be adopted is to double-up on the hi-fi amplifier and
loudspeaker (or if just starting in the field to obtain a stereo amplifier
complete), replace the existing recording/replay head with a stereo model
and feed the second channel from the head to the second hi-fl amplifier,
209
 THE PRACTICAL HI-Fl HANDBOOK
or to the second channel of the stereo amplifier either direct or by way of
a head amplifier.
A head amplifier will not be required if the amplifier has a low-level
(l-3 mV) tape input channel, but a large number of amplifiers, although
making provision for a tape input channel, require an input of some
200-300 mV of tape signal, and need a head amplifier to raise the low-level
signal accordingly. The circuit of such an amplifier is given in Fig. 10.9. This
is based upon the Mullard Type C tape amplifier, with equalization fixed for
a tape speed of 7½ in. per second. The power requirements are quite modest,
and can be obtained with ease from almost any hi-fi power amplifier.
There are various commercial head, or sub-, amplifiers on the market
for those who do not wish to construct their own. A versatile unit is that by
Cape Electrophonics of Southampton. In basic form this uses a single-valve
circuit with a gain in the region of l 00, but it can easily be modified for
particular applications. For example, Model B is already set-up for use with
a tape deck, and embodies equalization to C.C.I.R. recommendations.
Now that stereo amplifiers are becoming popular, more manufacturers
are adding tape input channels with sufficient sensitivity to operate direct
from the replay head of a tape deck. This means that if one is interested
simply in playing stereo tapes, the head circuits can be connected direct to
the stereo amplifier, in the same way as can a pick-up. Once stereo amplifiers
have become well established, stereo tapes could very much be popularized
by a go-ahead manufacturer producing a reasonably-priced stereo tape
player on the lines of a record player, and suitable for playing monaural as

_ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _loo_v_ _ ___.H.r.,

I==~=,. OUTPUT
300mV

INPUT-

2·2M
50

FIG. 10.9. Circuit diagram of a tape-head amplifier. This is based upon the Mu/lard
Type C tape amplifier, with equalization for a tape speed o/1! in. per second.
2)0
 STEREO PHONY

FIG. IO.IO. Making a stereo tape recording of the BBC's stereo broadcast.

well as stereo tapes. Such an instrument would find a ready market among
hi-fi enthusiasts who are not interested in making tape recordings. Instru-
ments providing for both recording and reproduction are necessarily rather
expensive, and this may be the reason why tape records have not made more
rapid progress.
If a stereo head is added to a standard tape recorder or tape deck, single-
channel work can be carried on as usual, but if stereo recording is to be
attempted, it would pay to obtain either a full-track erase head or a bulk
erasure. It is very disconcerting to find that only one half of the tape has been
erased after expending considerable effort in the creation of a stereo tape
record.

STEREO FROM RADIO

This aspect of stereophonic reproduction is as yet in the experimental

stage, at least in Great Britain. There have been a number of experiments in
stereo radio broadcasts by the BBC, and these have proved remarkably
interesting. Whilst television, medium frequencies and V.H.F.-F.M. channels
have been utilized in these experiments, there is little doubt that the V.H.F.-
F.M. channels represent the ultimate solution to the problem of obtaining
stereo from radio signals. With this in mind, several manufacturers of stereo
211
 THE PRACTICAL HI-FI HANDBOOK

'h~~:,:J: BALANCE
. . ,. . . JP,1-{L~.
CHA,NH }
~I ..,I
- - - - - - - - CONTROL------. z, :t'
j1 31
'i:.'. ~~r.-J LEFT-HAND
CHANNEL lJ}i:~:
- I
I
I
I
L
~ -

(a) (b)

FIG. 10.11. Two balance-control systems: (a) circuit with variable loading; (b) with
balance control associated with ganged volume control.

amplifiers have incorporated two radio input channels. Further stereo

experiments by the BBC will most likely use two V.H.F.-F.M. channels, as
hitherto, so the enthusiast with two V.H.F.-F.M. tuners and a stereo ampli-
fier will really be able to make the best of the experiments.
Two V.H.F.-F.M. tuners (E.A.R. switched units) were used by the author
during the course of the experiments in May, 1958. Fig. 10.10 shows the set-up
employed, with the author observing the operation of the tape deck. The two
E.A.R. tuners are seen situated on the floor beneath the trolley-a handy
device for tests of this nature. V.H.F.-F.M. signals were fed to these from a
common aerial through a distribution system, and the stereo a.f. signals from
the Third and Home programmes were channelled to the amplifier in a
Truvox tape recorder and to a Truvox Type K recording amplifier, seen on
the second shelf of the trolley.
The recorder was fitted with a stereo head so as to record the stereo
signals obtained from the two channels. In addition, two hi-fi amplifiers (a
Pamphonic and a Pilot) received signals from the appropriate outlet sockets
of the recording amplifiers so as to provide immediate reproduction from the
two E.A.R. loudspeakers situated either side of the trolley. The set-up thus
permitted the immediate reproduction of the stereo sound as well as the
recording of the signals. The tests proved very successful from both aspects;
excellent live stereo was obtained during the transmission and, on playing
the tape recording afterwards, the performance was almost equal to the
original. The tests were performed some 60 miles from the V.H.F.-F.M.
transmitters, showing that stereo via radio is feasible in fringe areas and also
that it can be recorded with remarkable ease.
212
 STEREO PHONY
When making stereo tape records, the accepted practice is to record the
left-hand channel on the top track, that is when the tape is travelling from
left to right with the active side away from the observer.

BALANCE CONTROL
If two separate control units or hi-fi amplifiers are used for the playing
of stereophonic records or for reproducing stereo from the radio, it is
essential to maintain optimum balance by very carefully adjusting the two
volume controls. Tonal balance between the two channels is also important.
Stereo control units or amplifiers use ganged controls, so that operation of
the common control knob will vary the volume or tone of the two channels
in step. In addition to ganged controls, a balance control is often desirable
so that the volume of one channel in relation to the other can be altered
slightly to compensate for a difference between the sensitivities of the loud-
speakers, for example, or to counteract acoustical shortcomings of the
listening room.
Two methods for obtaining balance control are given in Fig. 10.11.
Diagram (a) is a straightforward arrangement which varies the loading, and
thus the applied signal, at the grids of the two triodes. An ordinary 500k
linear volume control serves as the balance control, and the two 130 k
resistors in series with it avoid excessive loading which would result in a
fall of top response.
A more commendable arrangement is given in Fig. IO.II (b). Here the
balance control is directly associated with the ganged volume control. The
balance control is also a ganged component, but is connected differentially,
so that as it is rotated the signal input is increased in one channel and
decreased in the other.
These controls can be installed in existing amplifiers if required, but

FIG. 10.12. The Pamphonic Type 3,000 stereophonic amplifier.

213
 155

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•

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n .. ~ ,.. --• __ ____ r>U CJJ 11 I A 1,0 · 150\'
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GRAM M010ll

FIG. I 0.13. Circuit diagram of the Pamphonic Type 3,000 stereophonic

amplifier. The component values are given on the opposite page.
 STEREOPHONY

Resistors Resistors Resistors Resistors

-· - - - - - -
I I I
R 1 IM Rl6 2·2K R31 2·2K R46 22K
R2 IO0K Rl7 IO0K RJ2 )()().Q R47 470K
R 3 lOOK RIS IOOK R33 22K R48 680.Q
R4 IM Rl9 470K RJ4 2·2K R49 470K
R S IOOK R20 22K R3S 22K RS0 27K
R 6 IOOK R21 IOOK R36 2·2K RSI 470K
R 7 3·3K R22 IOOK RJ7 IOOK R52 470K
R 8 6·8K R2J 470K RJS IOOK RSJ 27K
R9 3-JK R24 22K R39 IOOK RS4 470K
RIO 6·8K R2S 220K R40 IOOK RSS 22K (½W)
Rll IOOK R26 2·2K R41 470K RS6 400.Q (3W)
Rl2 IO0K R27 220K R42 22K RS7 22K (½W)
Rl3 220K R28 2·2K R43 470K RSS 400.Q (JW)
Rl4 2·2K R29 2·2K R44 680.Q RS9 220.Q (6W)
RIS 220K R30 100.Q R4S 470K !

Note: All resistors ¼W unless otherwise stated.

Capacitors Capacitors I Potentiometers

I
Cl S0pF S% Cl7 50µ.F 2SV Pl IM Jin.
C2 S0pF S% CIS S0µ.F 2SV P2 ½M Jin.
CJ 2SV
SOµ.F Cl9 50µ.F 2SV P3 IM lin.
C4 SOµ.F
25V C20 SOµ.F 25V P4 ½M lin.
CS 0·lµ.F JS0V C21 O·lµ.F JS0V PS ½M Jog.
C6 0·lµ.F JSV C22 0·1µ.F 350V P6 ½M log.
C 7 4,700pF 10% C23 3,000pF 10% P7 lK lin.
C 8 4,700pF 10% C24 0·lµ.F J50V PS lK Jin.
C9 lOOpF 20% C2S 0·l µ.F 350V
Valves, Rectifier
CIO 4,700pF 10% C26 J,OOOpF 10%
Cll 4,700pF 10% C27 32µ.F 450V ' VI l 2AX7 or ECCSJ
Cl2 IOOpF 20% C28 32µ.F 450V V2 I 2AX7 or ECCSJ
Cl3 0·lµ.F 3S0V C29 50µ.F 2SV VJ ECL82
V4 ECL82
Cl4
ClS·
SOµ.F 2SV
0·lµ.F 3S0V
C30
C31
SOµ.F
J2µ.F
2SV
450V I vs ECL82
Cl6 SOµ.F 2SV C32 32µ.F 4S0V V6 ECL82
1,
'
WI B2S0 ClS0

Component values/or Pamphonic Type 3,000 stereophonic amplifier (see Fig. 10.13).
215
 THE PRACTICAL HI-Fl HANDBOOK
care should be taken to position the circuit so that it does not affect the
frequency response or incite overloading of the low-level stages.

STEREO EQUIPMENT
Because of the enormous difference between single-channel hi-fi and
stereophony, there is little doubt that stereophony is destined to become the
hi-fi of the future. It will be available from tape, disk and radio; the gradual
development of the propagation of television and radio signals by way of
cables mstead of the ether will lend itself admirably to regular stereo broad-
casts on ordinary radio as well as television sound.
The change from single-channel sound to stereo is now taking place;
stereo attachments are available for existing equipment and complete stereo
amplifiers and control units. Stereo will not be adopted only by hi-fi enthusi-
asts, :is add-on units are already becoming available for the popular record
player and radiogram. B.S.R. have adapted two of their "Monarch" record
changers (Models UA8 and UAl2) to take stereo cartridges for stereo
reproduction, and these are being supplied to leading manufacturers. Com-
bined stereo-monaural pick-ups are also being developed. During the London
Audio Fair in 1958, R.G.D. demonstrated that the "Victoria" radiogram is
easily convertible for stereo operation by the fitting of a stereo pick-up
cartridge and an additional loudspeaker-amplifier unit. The "Victoria"
radiogram comes within the hi-fi category. of course; however. a number of
ordinary record-players and radiograms of reasonable quality which are now
on the market can readily be adapted to stereo operation.
Nevertheless, the hi-fi enthusiast will still wish to employ hi-fi amplifiers,
pick-ups and loudspeaker systems, as hitherto. For really high-quality stereo
reproduction, special attention will have to be paid to such things as turn-
tables and pick-up arms. Motor rumble will be found to be more trouble-
some with a stereo system than with a monaural system, since the stereo
pick-up responds almost equally in all directions of movement of the stylus.
Consequently, a motor which is virtually rumble-free on a monaural system
may exhibit a most disconcerting rumble when a stereo pick-up is incor-
porated.
There is also the question of the pick-up arm. If the pick-up is tracked
at the ideally low pressure of 2-3 grams, this may be found to have too
much friction in the main bearing. There may be a temptation to use a greater
tracking pressure and a stylus point radius exceeding the recommended
0·0005 in. as a means of keeping the stylus in the groove. Such expedients
should be avoided if possible, for a record once played under these conditions
will never give its best when later it is played under the correct conditions.
The Pamphonic Type 3,000 stereo amplifier (Fig. 10.12) is of interest.
As may be seen from the circuit in Fig. 10.13, it employs two independent
1

216
 STEREOPHONY
channels with ganged controls. There are three switched inputs catering f'or
pick-up, tape and radio. These are selected by switch SI, which has six
positions, giving three stereo positions and three monaural positions. The
pick-up channel is compensated for R.I.A.A. (British Standard 1928:1955
Fine Groove) recording characteristic and is adjusted for use with a stereo
crystal cartridge, while the radio and tape channels are substantially flat from
50 c/s to 15 kc/s and require signal inputs of l volt and 0·5 volt respectively
for 5 watts output. Each channel is rated at 7 ·5 watts, and on the monaural
positions the signal input is applied to both channels simultaneously, thus
giving a total power output of 15 watts.
Baxandall tone-control circuits are incorporated around stages V l and
V2 and provide 15 db variations at 50 c/s and IO kc/s. There is also an
interesting negative-feedback balance control, P7 /PS, which serves to vary
the gain of each channel differentially by 6 db from the level position. A

Fm. 10.14. The Sound Sales Tri-channel Mk. JV power amplifier.

channel-reversing switch, S2, is provided at the input of the amplifier, and is

useful in cases where the pick-up is connected wrong way round, or if the
pick-up wires are not known. There is also a phasing switch, S3, in the right-
hand channel loudspeaker circuit. This avoids having to fiddle with the
loudspeaker connexions.
The remainder of the amplifier follows conventional lines, details of
which have already been given. It is, however, interesting to see an ECL82
connected in the ultra-linear mode. A metal bridge rectifier supplies h.t. for
both of the channels, with excellent regulation.
Stereo tape equipment is marketed by E.M.I. for the playing of their
Stereosonic tape records, and the G.E.C. also make a stereo tape-playing
outfit. The latter incorporates the G.E.C. metal-cone loudspeaker system
217
 THE PRACTICAL HI-Fl HANDBOOK

FIG. 10.15. The Sound Sales Tri-channel loudspeaker system.

and presence unit, two BCS24 I 7A/ 18A 12-watt amplifiers and the Truvox
Type TR2 I I 2 tape deck.
Sound Sales, Ltd. market an excellent stereophonic version of their
"Tri-Channel" equipment. Each channel uses, in effect, three separate
amplifiers each covering limited sections of the frequency range and fed from
an infinitely variable electronic cross-over system. Three loudspeaker
systems are used on each channel, all being contained in a single cabinet of
special design.
This arrangement avoids the necessity of reactive cross-over units at the
loudspeakers themselves. The stereo Tri-Channel outfit comprises two Tri-
Channel main amplifiers, each rated at 50 watts, two Tri-Channel loud-
speaker systems and a Tri-Channel stereo control unit. The control unit has
a stereo/monaural changeover switch, giving parallel ( 100 watts) or stereo
connexion. Sound Sales also market a transistorized stereo control unit,
which has provision for direct replay from a low-level pick-up or tape head.
The Sound Sales Mk. IV power amplifier is illustrated in Fig. 10.14 and the
loudspeaker system in Fig. 10.15.
Especially worthy of mention is the stereo equipment by Cape Electro-
phonics, Ltd., comprising a stereo control unit of high standard, including a
facility for the direct connexion of a tape head, and associated power
amplifiers. For enthusiasts not interested in building their own equipment,
the control unit and amplifiers are available in ready-made and tested form,
218
 STEREOPHONY
Cape Electrophonics also manufacture and market the "Cape Audio
System", which includes facilities for recording and reproducing tape, as
well as control units and very high-quality power amplifiers. The equipment
is designed to very rigid specifications, thus ensuring results of professional
standard.
At the time of writing stereo pick-ups in ever-increasing numbers are
being made available which are suitable for playing the Nixa (Pye) stereo
records. Fig. 10.16 illustrates the "Acostereo" crystal cartridge (Cosmocord,
Ltd.), whose essential details are as follows. At a recorded level of I ·5 cm/sec

FIG. 10.16. The Acostereo crystal

cartridge by Cosmochord, Ltd.

an output of 200 mV is available; the frequency response is from 40-12,000

c/s and the separation between channels at l kc/s is better than 15 db. The
cartridge will track at a minimum of 2 grams, but since this very small
tracking pressure is dependent upon the lateral freedom of the tone arm, it is
suggested that pressures in excess of 2 grams may be required, especially
with autochangers. The recommended load is 2 megohms, and the radius of
the stylus tip is approximately 0·0005 in. to conform with the stereo standard
for disk records.
In order to avoid undue wear on the stereo record and stylus, the lowest
possible tracking weight should be used. A tracking weight in excess of
6-7 grams will ruin a 0·0005-in. stereo stylus after a few hours' playing,
while the use of a 0·001-in. stylus, particularly if it is tracking fairly heavily,
will ruin a stereo disk almost immediately.

219
 Index
Acos Black Shadow pick-up, 150 Cardioid response, 164
Acostereo crystal cartridge, 219 Cathode-coupled phase-splitter, 67
Acoustic curtains, 117 Cathode follower, 45
Acoustic lens, 123 Cathode loading, 64
Acoustical feedback, 155, 173 C.C.I.R. characteristics, 187
Acoustical labyrinth, 119 Check for:
Acoustical Quad II amplifier, 64 mains hum, 11
Acoustical resistance unit, 117 negative feedback, 92
Acoustics, poor, 22 output-stage balance, 89
Amplifier, complete failure of, 73 signal balance, 90
Armstrong unit, 41 Choice of microphone, 165
Atmosphere, 79, 80 Collaro Mk. III tape transcriptor, 183
Attack, in music, 13 Collaro Studio crystal cartridges, 143
Audible-frequency range, 12 Collaro Studio "O" crystal unit, 148
Automatic tape reversal, 181 Common-base circuit, 170
A vometer, model 40, 116 Common chassis point, 100
Axiom 80, I 04 Compatibility, 206
Azimuth adjustment, 192 Condenser microphone, 162
Cone break-up, 104
Background hiss, 193 Cones, metal, 104, 122
Baffles and vented enclosures, 114 Constant amplitude, 130
Balance control, 89 Constant velocity, 130
Bass correction, 40 Control circuits, 27
Bass-heavy effect, 173 Conversion of decibels to power ratios,
Baxandall negative-feedback system, 49 14
Beats, 18 Conversion of power ratios to decibels,
Bel, 13 15
Bender crystal, 207 Core saturation, 96
Bias current, 192 Cossor Amplifier Kit 562K, 55
Bias oscillator, 192 Crackling, 171
Binaural system, 197 Cross-over networks, I09
Blumlein, A. D., 202, 203 Cross-over unit, 105
Briggs, G. A., 84, 96, 118, 128 Cross-talk, 207, 209
British Standard Spec. 1928, 38, 133 Crystal microphone, 161, 166
British Standard Spec. 2478, 185 Crystal pick-ups, 144, 154
Buchman and Meyer pattern, 134 Crystal units, 148
Bulk erasure, I 81 Cushion effect, 57
Burne-Jones (B-J) arm, 159
Burne-Jones tweeter unit, 123 Damped oscillation, 68
Damping factor, 36, 67
Cabinets, 117 Decca FFR 25, 41
Cantilever-type stylus, 143 Decca test record LXT5346, 147
Cape Electrophonics, 210, 219 Decibel, 12
220
 INDEX
Decibel tables, 14 Frequency distortion, 80
Defluxing, I 92 Frequency-dividing network, 105
Dialomatic pick-up compensation, 41, Frequency-selective network, 22
94 Frequency/wavelength con version
Disembodied treble, 113 factor, 17
Disk recording, 127 Fringing, 179
Disk recording problems, 133
Distortion, 69, 79
Distortion, correction of, 88 Gain margin, 35
Distortion, non-linear, 82, 84 Gaps, 179, 180
Distortion, phase and transient, 85 Gauss, 102, 209
Disturbed load, 63 G.E.C. amplifier BCS 2317/8, 101
Doppler effect, 195 G.E.C. Periphonic loudspeaker system,
Dynamic microphone, 166 122
Dynamic range of a programme signal, Goldring Jubilee pick-up, 159
27 Goldring pick-up, type 600, 147, 152
Goldring variable-reluctance cartridge,
type 500, 140, 144
Ear, sensitivity of, 12 Goodman's Axiom 150 Mk. II loud-
Earthing, JOO speaker, 116
Effect of damping factor on loud- Goodman's Midax loudspeaker, 106
speaker, 68 Gramophone motor rumble, 44, 97
Effect of wind on sound waves, 18 Gramophone record, 24
Eigentones, 21 Grid current, 88
Electrical power, 51 Groove-jumping, 129
Electrical representation of sound, 21 Grundig mixer, type GMU3, 167
Electrolytic capacitors, 74 Grundig Stenorette recorder, 195
Electromagnetic induction, 29
Electromagnetic pick-ups, 138
Electromagnetic waves, 17 H (magnetizing force), 176
Electrostatic units, 107 Hangover effects, 114
Emptiness (result of incorrect phasing), Harmonic components, 12
113 Harmonic distortion, 82, 84, 92, 196
Enclosure construction, 11 7 Harmonics, 15
Equalizing for room acoustics, 22 Hearing, sensation of, JO
Equalizing networks, 144 Heat "sink", 171
Erase head, 180, 21 I Helmholtz resonator, 20
Exponential horn, 118 Hi-fl standards of judgment, 9
Hi-fl tape recorder, 174
Hi-fl transformers, 54
Faults, transformer, 89 High-coercivity tape, 184
Feedback, 32 High-frequency limit of audibility, 12
Feedback, application of, 67 High-permeability magnetic shields, 30
Feedback stability, 34 Hill-and-dale recording process, 26,
Filters, 43 202, 203, 205
Fletcher-Munson curves, 46, 72 Horn loading, 106
Floating paraphase, 62 Hot-stylus technique, 135
Flutter, 154, I 94 Hum caused by stray fields, 29
Forced vibration, 20 Hum troubles, 97
Frequency, 11 Humdinger control, 90
Frequency and power response, 184, 196 Hysteresis loop, 176
221
 INDEX
Infinite damping factor, 67, 69 Moving-coil pick-up, 138
Intermodulation analyser, 85 Mullard mixer, 167
Intermodulation distortion, 19, 84 Mullard tape amplifier, type C, 210
International Electrotechnical Commis-
sion, 205 Needle-talk, 143
Inverter stage, 62 Negative feedback, 36, 67
Nixa (Pye) Stereo records, 219
Kanthal, I06 Noise and hum, 28
Klein, S., 106
Output:
Leevers-Rich Equipment Ltd., 181 from pick-ups, 144
Longitudinal waves, 17 impedance, 27
Loop gain, 33 of microphone, 22
Loudness, 11 transformer, 51, 53
Loudness control, 14, 96 Overtones, 16
Loudness control circuits, 47
Loudspeakers: Pamphonic equipment:
adjustment of systems, 124 amplifier, model 1004, 38, 71
enclosures for, 102 power amplifier, 67
for stereophonic sound, 200 pre-amplifier, model 2001, 39, 61, 72
impedance measurement, 115 sound equipment, 167
line-source, 119, 173 stereo amplifier, type 3000, 216
matching, 113 stereo loudspeaker, model SI, 202
placing, 126 Victor loudspeaker system, 118
placing in stereophonic sound, 199 Parallel formation, 112
suspension, 103 Parallel-T circuit, 44
Low-impedance pick-ups, 151 Paraphase circuit, 62
Lowther high-frequency unit, type PM6, Parasitic oscillator, 92
123 Pause control, 183
Lustraphone full-vision microphone, 165 Pentode or triode in hi-fi reproduction,
56
Magic-eye signal-level indicator, 168 Periphonic loudspeaker system, 122
Magnetic recording, 26 Phase distortion, 85
Maxwells, 102 Phase margin, 35
Mercury cell, 169 Phase-shift tests, 87
Metal cones, 104, 122 Phase-splitting, 60
Micro-armatures, 140 Phasing stereo channels, 200
Microphone: Philips "Hi-Z" power amplifier, 65
balance, 172 Philips "needle clinic" microscope, 142
mixers, 167 Philips Novosonic pick-up, 139
placing, 198 Phon, 14
sensitivity, 164 Piano, range of, 173
types of, 160 Pick-up:
Microphony valve, 31 matching, 150
Miller effect, 151 mechanics, 139
Monarch record changers, 218 resonances, 141
Monaural system, 25, 196 Pinch-effect, 135, 144
Motor-boating, 93 Plastic foam, 103
Moving-coil loudspeaker, 102 Playing time, 182
Moving-coil microphone, 161 Plessey ionophone, I06
222
 INDEX
Polar responses, 163 Second-harmonic distortion, 16, 57
Poly-crystalline barium titanate, 139 Servicing microphones, 171
Power-amplifier: Servicing tape equipment, 192
failure, 75 Serviscope, 81
output, 69, 81 Signal detector, 167
supplies, 70 Simon tape recorder, 181
Pre-amplifier: Single-ended output stage, 55
failure, 78 Single-ended push-pull stage, 64
first stage, 39 Sound:
units, 169 columns, 119
Precedence effect, 200 distortion, 122
Presence, correct degree of, 118 radiation from musical instruments,
Pressure units, 106 173
Programme selection and equalizing, reflection, 21
37 reinforcement, 166
Programme-selector switch, 75 reproduction, 22
Push-pull output, 58 waves, 21
Pye HF25 pre-amplifier, 38, IOI Source impedance, 36, 67, 69
Pye HF25A pre-amplifier, 49 Square-wave tests, 86
Pye Mozart amplifier, 94 Squawker, 105
Pye Proctor pre-amplifier, 28, 72 Standing waves, 21, 126
Pye Provost power amplifier, 63, 72 Stereophonic experiments, 200
Stereophonic sound, 25
Quad electrostatic loudspeaker, 109 Stereophony, 196
Quality in sound reproduction, 11 balance control, 213
Quarter-section filter, 111 equipment, 201, 216
from disk, 202
Radius compensation, 134 from radio, 211
R.C.A. units, 41 from tape, 208
R.D. Junior control, 78 pick-ups, 207
Record changers, 155 tape records, 213
Recording characteristics, 129 Stringed instruments, 173
Recording from radio, 194 Styli, 141
Recording head, 24, 26, 178 Stylus, cantilever type, 143
Recording level, 131 Stylus, dimensions of, 128
Recording tracks, 181 Stylus replacement, 142
Remanent induction, 177
Reslo ribbon microphone, 165 Take-over frequency, 110
Resonance, 19 Tape decks, 181, I 90
Resonance frequency, of vent, 11 Tape recorder output socket, 50
Resonator, 20 Tape recorders, Continental, 166, 181
Reverberation of a room, 173 Tape recording process, 175, 180
R.l.A.A. recording characteristics, 38 Tape speed, 179, 185
Ribbon loudspeaker, 106 Tape, stereophonic, 208
Ribbon microphone, 160, 162, 165 Television interference, 31
Ringing, 68 Temperature, effect on sound waves, 18
R.M.A. rating, 22 Third-harmonic distortion, 57
Rochelle-salt crystal. 137 Time-constant, 146
Rumble, 154 Tone control, 23, 41
Rumble filters, 44, 97 Tone control and equalizing faults, 95
223
 INDEX
Tracing di1>tortion, 135 Unisil core material, 55
Tracing distortion through the ampli- Use of tape, 174
fier, 93 Use of treble anri bass controls, 173
Tracking error, 158
Tracking problems, 158
Transducer, 21, 24 Valve microphony, 31
Transfer distortion, 178 Variable damping control, 37
Transformer faults, 89 Variable-reluctance pick-up, 138
Transformer output, 53 Velocity, 17, 105
Transformerless operation, 64 Vent resonance frequency, 11
Transient distortion, 85 Vented enclosures, 114
Transients, 12 Vibration, 11, 31
Transistor units, 169 Victoria radiogram, 218
Transistorized mixer, 169 Violin, harmonics in 16, 19
Transmission of sound, 16 Vision tape records, 174
Transverse waves, 17 Vision Electrophonic recording appara-
Tri-channel equipment, 218 tus, 179
Truvox: Voltage amplifier, 27
Mk. IV tape deck, 183 Volume, change in, 112
stereo head, 209
Type K amplifier, 187, 209, 212
Walker, P. J., 109
T.S.L. Lorenz Type L.P. 312-2 loud-
Wavefronts, 12, 17
speaker system, 118
Wavelength, 17, 133
Tuned-pipe arrangement for loud-
Wavelength/frequency conversion fac-
speaker, 118
tor, 17
Turntable units, 152
WB Stentorian loudspeaker, 103
Tweeter, 56,105, I 10
Wearite defluxer, 193
Twin-cone and multiple units, 104 White noise, 29, 101, 103
Twister crystal, 207, 208 Williamson amplifier, 93, 147
Two-channel system, Blumlein 45/45, Woofer, 105
203, 205
Wow, 154, 194
Ultra-linear stage, 63
Unbalance of the rectifier valve, 98 Zurich Conference, November 1957, 205

224