the SOUND - American Radio History
588
SOUND
the
HIGH FIDELITY
of
by
1tOIIERT OAKES JORDAN
and
JA MES CUNNINGHAM
COPYRIGHT 1958
Artists:
Cover Artist:
DAVID FOSTER
ROBERT BARKER
ARTHUR LUTZ
ROBERT C. KORTA
Photographs by:
ROGER KIRKGASSER
POPULAR MECHANICS PRESS
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Printed in U.S.A.
ANY
of these superb
COLUMBIA and EPIC
BEETHOVEN:
EMPEROR CONCERTO
STRAUSS WALTZES
and OVERTURES
ROSSINI: WILUAM TELL AND
BEETHOVEN
BARBER OF SEVILLE OVERTURES
EROICA
CASADCSUS, Piano
MITROPOULOS
NEW YORK PHILHARMONIC
IIOEYZETYI: DAUGHTER OF THE
The Seasons
SYMPHONY
REGIMENT OVERTURE
SCHUBERT: MARCHE MILITAIRE
TCHAIKOYSKY: MARCHE SLAVE
CLEVELAND
r
STRAUSS. RADETZKY MARCH
ORCHESTRA
gw4
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BRUNO WALTER
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NEW WORLD
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@BRUNO WALTER
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GLENN GOULD
NANO
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WE$TM,NSTIR CHOIR
SIBELIUS
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PAGANINI:a,
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LEVANT PLAYS GERSHWIN
Pi1M OVNI SUM Wa.l
SWEDISH RHAPSODY
PATHÉTIQUE
SYMPHONY
cud 2unopeou cP.amtcaQ necevtds
watQdrs P.angW
CONCERTO tN
PAVANE
BOLERO. LA VALSE.
CHABRIER
A
need dub
PORTS OF CALL
RAVEL
.
ESPANA-
!BPI
:
ESCALES
DEBUSSY CLAIR DE LUNE
A
AN AMERICAN IN PARO
BACH
Toccata, Adagio
Fugue in A Mum
VIVALDI: Concedo for Two Violins
BACH: Violin Concertos I and 2
Philadelphia Orchestra Ormandy
RUDOLF SERKIN
Albert Schweitzer
and Fugue in C Minor
k
N
Mitropoulos,
New York
Philharmonic
Philadelphia Orchestra -Ormandy
DAVID OISTRAKH
ISAAC STERN violins
RHAPSODY
IN BLUE
Fantasia and Fugue
in G Minor
HAYDN
BEETHOVEN"'"''
"MOONLIGHT" Sonata
STRAVINSKY:°
"PATHE'IOUE" Sonata
FIREBIRD SUITE
"APPASSIONATA" Sonata
TCHAIKOVSKY:
ROMEO AND JULIET
LEONARD BERNSTEIN
Brandenburg
Concertos
Surprise.`
AND
~Sos.1,
,
PABLO CASALS
BEECHAM. Royal Philharmonic
SCHUBERT:
TLHAIHOYBRT serenade
BORnIH nocturne
"UNFINISHED" SYMPHONY
In
I NAIN
BARBER Adagio
í>
VAOGHN WILLIAMS
MENDELSSOHN:
MIDSUMMER NIGHT'S DREAM
lanlasia on greensleeres
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-1
2. Beethoven: Emperor Concerto
3. Beethoven: Quartets 9 and 11
4. Mozart: Requiem
5. Finlandia, Swan of Tuoneta, etc.
6. Beethoven: 3 piano sonatas
7. Beethoven: ''Eroica'' Symphony
8. Vivaldi: The Seasons
9. Tchaikovsky: "Pathetique" Symphony
10. Dvorak: "New World" Symphony
11. Bach: Goldberg Variations
12. Schweitzer Plays Bach, Vol.
13. Rossini: William Tell Overture, etc.
14. Strauss Waltzes and Overtures
15. Mendelssohn: Midsummer Night's Dream;
1
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1
Dimitri Mitropoulos
CIRCLE 5 NUMBERS BELOW:
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16.
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18.
19.
20.
21.
22.
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Vivaldi and Bach: Violin Concertos
Haydn: "Surprise", "Drum -Roll" Sym.
23.
24.
Paganini and Saint-Saens Concertos
Strings of Philadelphia Orchestra M-211
Brahms: Symphony No.
4
Ports of Call
Debussy: La Mer; Ravel:
to Valse, etc.
Bach: Brandenburg Cont. 1, 2, 3
3
CONTENTS
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ALLIED CATALOG
SOUND
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HEARING
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28
The authors would like to extend
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sible. Chief among these are:
Jory Nordland
Robert Nereim
Ruth Nereim
Charles Shipley
Edgar Vinyard
Mrs. Robert Oakes Jordan
Gee Reid
William Leonard
1
SOUND
THE NATURE OF SOUND
One man interested in the production of sound might describe it thusly:
Sound is produced by a body vibrating in air. Another might say: Sound is
a longitudinal vibration occurring as a mechanical disturbance of air particles. Yet another might say: Sound is a sensation received in the ear due
to slight changes in atmospheric pressure, which results in hearing.
All these answers are partly correct; all serve to offer an idea of what
really constitutes sound. However, an all-inclusive answer would involve a
combination of these responses, with attention paid to the vibrating object
producing the disturbance which is conveyed to the hearing mechanism, causing
the sensation of hearing in the mind. Such a definition might sound something
like this:
Sound is longitudinal vibrations set up in a surrounding medium (air,
water, iron, etc.) by a vibrating object, which, when communicated to
the brain by the ear, produce the sensation of hearing.
We are primarily concerned with sounds conveyed to us through air. The
actual connecting link between the air in a recording studio or concert hall
and that in our listening rooms may be a disc record or a recorded magnetic
tape. The definition offered previously will still hold true in these circumstances.
Though it differs from individual to individual, the range of human hearing
is now established as being from below 10 vibrations per second to above
18,000 vibrations per second. The rate of vibration of a sound is the frequency
of the sound, which is usually expressed in terms of cycles per second.
Sound energy is conveyed through air at the rate of approximately 1,100
feet per second. Even though sound energy may be transmitted over large
distances at this rate, no movement of the medium as a whole takes place.
The air particles oscillate along the direction of propagation, but do not move
very far from their original positions. Sound is a transfer of energy from
molecule to molecule of a medium, from its source to a receptor.
FREQUENCY
When an object vibrates-a tuning fork or a piano string, for example-it
actually moves to and fro in the air. The number of times the object vibrates
each second is governed by several physical factors. A tuning fork made of
5
TUNING FORK
AT MAXIMUM
POINT OF
COMPRESSION
CYCLE
As the
TUNING FORK
AT MAXIMUM
POINT OF
RAREFACTION
CYCLE
vibrating tines of
a tuning fork move away from each other, they compress the air around
them. As they move toward each other, the air is rarefied
lead would vibrate in a different way than an otherwise identical one made
of steel. The difference is caused by the variation in density between lead
and steel. The lead tuning fork, being denser, would have a much lower rate
of vibration than the steel fork.
Another point of difference might be the actual comparative sizes of two
vibrating objects. A small steel tuning fork would vibrate much more rapidly
than a large fork made of the same material. The relationship of such physical
factors to frequency of vibration is extremely important.
As an object vibrates, its motion is cyclical. Imagine an inverted tuning
fork vibrating. The two tines of the fork move back and forth in cyclic oscillation from the resting, or middle, position. To follow just one of the tines
through its path: if it were possible to see the tine move, the movement would
appear exactly like that of a clock pendulum. Once the tuning fork was set
in motion, the swing from side to side would always take the same amount
of time. Starting from the rest position the tine would swing to one side,
back through the middle, out to the other side, and finally back to the middle
again, in position to start another cycle.
The frequency of vibration of a clock pendulum is usually about one cycle
per second. The sound the clock makes comes from the escapement, a mechanism which restores the small amount of energy lost through friction in the
swing of the pendulum. The pendulum strokes are thus made to cover the
same distance each cycle, maintaining the accuracy of the clock. In a tuning
fork or other vibrating object, friction will in time cause the vibrations to die
away. Even though the intensity of the vibrations goes from maximum to
dead still, however, the frequency of vibration will remain the same, due to
those all-important physical factors of size and density.
WAVELENGTH
As a vibrating object moves from the rest position, from side to side, and
back, it is considered to have completed one cycle. It is the length of time
taken by the object to cover this distance which governs the wavelength of
any particular frequency of sound. Hence, the wavelength of any sound is
measured from two corresponding points in successive cycles of vibration.
If a tuning fork vibrates at a rate of 1,100 times per second, then the frequency
of the sound it produces in air is 1,100 cycles per second. As a further example:
6
the tuning fork has a frequency of 1,100 cycles per second; sound travels at
a rate of 1,100 feet per second in air. It can easily be calculated, by dividing
the speed, or velocity, of sound by the frequency, that the wavelength is 1
1100 _1
foot V or
1100
F
If the frequency under consideration were only half that of the preceding
example, 550 cycles per second, the wavelength would be 2 feet. The simple
formula given here will be most useful to the experimenter. It is easy to
remember that the product of wavelength and frequency is equal to the speed
of sound and that, in consequence, wavelength is inversely proportional to
frequency.
Where the source of a particular wavelength is small, the sound can be
considered to come from a point source. A large symphony orchestra is
composed of many of these small point sources, all combined into large wave
fronts of sound. A wave front may be visualized as a surface which passes
through all the points of equal density in one wave of a vibrating medium; in
sound, the points of initial disturbance of the medium. A wave front may
have a complex shape, indicating a complex frequency range, when coming
from an orchestra; it may have a very simple shape when coming from a
single small source such as a tuning fork.
The ever-expanding series of wave fronts coming from a tuning fork might
be conceived of as an infinite number of spheres of smoke particles being
A wave front of sound is an
area in which all the air in
one sound wave is of equal
density. A series of regular
wave fronts, as propagated
by a tuning fork, is shown at
right. As a wave front gets
farther and farther from the
source of initial propagation
it tends more and more to
resemble a straight line
STRAIGHT LINE
7
SHORT -SWING
PENDULUM
SINE WAVE
The simple harmonic motion of a swinging
pendulum may be used to inscribe a sine
wave pattern on a strip of paper moving
at a constant speed. The pendulum arm
must be very long and the arc swung must
be very short or the shape of the sine wave
will be distorted
PEN
MOVING AT
CONSTANT
EXTENT AND
DIRECTION OF
SPEED
SWING
generated inside each other, constantly expanding, each getting larger until it
fades in the air. If the spheres were able to expand sufficiently, they might
ultimately be observed as straight-line wave fronts.
The waves of sound emanating from a vibrating tuning fork are caused by
compression of air as the tines of the fork move away from each other and a
succeeding rarefaction of the air as the fork tines reach their maximum points
of travel and come back to the center position. In other words, the vibrating
sound generator changes its direction of travel during each cycle.
PITCH AND QUALITY
If an 1100 -cycle note from a tuning fork were followed immediately by a
note of the same frequency played on a piano, even the least musical of us
would be able to tell that the sounds did not come from the same instrument.
All the musicians in an orchestra may tune up on A, but a wide variety of
sounds will come from the group in practice. The two elements which give
each instrument its own characteristic sound are pitch and quality (timbre).
The pitch of a sound is determined by the fundamental frequency of vibration at the source of the sound. In musical scoring, the terms treble and bass
are used to indicate high-pitched and low-pitched tones respectively. In the
reproduction of sound by means of electronic instruments, bass and treble
are used to indicate the degree of emphasis on low- or high-pitched tones
desired by an operator at the controls.
Pitch can be described as the subjective effect of frequency of vibration;
that is, the effect of the quality of frequency on the human hearing system.
The intensity of any sound affects the pitch of the sound. In the case of
low -frequency sounds, intensity varies directly with pitch; just the opposite
is true at high frequencies, where pitch varies inversely with intensity. In
plain language, two sounds with the same frequency characteristics would be
different in pitch to a listener if the intensity levels were far enough apart.
If the character of a sound depended only upon pitch, two different instruments vibrating at the same frequency and intensity would produce identical
sounds. Earlier in this text, the to and fro vibrations of a single tine of a tuning
fork were compared with the movement of a clock pendulum. Both the fork
and the pendulum exhibit what is called simple harmonic motion. If a pen
were fixed to the bottom of the pendulum and the pendulum allowed to swing
naturally, it would draw a straight line on a piece of paper placed under it. If,
8
however, the paper was moved at right angles to the swing of the pen, the picture
on the paper would be quite different. Instead of a straight line, a perfectly
symmetrical sinusoidal wave would be drawn on the paper. Using this curve
as a basis for comparison, and employing a writing oscillograph to plot a twodimensional picture of a wave of the sound from a piano string, it would be
found that the latter curve was not nearly as smooth and symmetrical as the
former. The piano-string vibration is composed of a fundamental frequency
just like the one from a tuning fork, with the addition of certain other frequencies known as harmonics. These harmonic frequencies are multiples of
the fundamental frequency of the piano note. These combinations of different
harmonics at different intensities with the fundamental frequency of a note
from any instrument are what gives the instrument its own particular quality,
or timbre.
A sound from any musical instrument may contain four or five harmonic
frequencies in addition to the fundamental frequency. Some instruments may
exhibit strong audible harmonics of the 2nd, 3rd, 5th, 9th, etc., order, while
others may be strong in only the 3rd, 5th, and 7th harmonics. It is the relationship of these harmonics in order, phase, and amplitude that is important
in establishing the character of an instrument.
Even though some of the harmonics of the instruments in an orchestra may
lie beyond the normal range of human hearing, they may combine to produce
audible tones which are most important to the over-all quality of the piece
of music being performed. In recording and reproducing sounds by means
of electronic equipment, these harmonics must be handled in proper proportion to the fundamental tones, without change or distortion, or loss of fidelity
will result.
INTENSITY AND LOUDNESS
When speaking of the physical properties of sound, people are often heard
using the terms intensity and loudness interchangeably. This is incorrect
usage. It is true that these characteristics of sound are closely related, but
they are not the same. The intensity of a sound in air is directly proportional,
A sound wave from a piano may be recorded
BASS NOTE
PIANO STRING
for visual inspection by means of an oscillograph
RECORDING
OSCILLOGRAPH
PEN
DAMPER
MOTION
PEN
HAMMER
PIANO
KEY
PURE SINE WAVE
(2 CYCLES)
SOUND WAVE
CONTAINING
HARMONICS
12
GRAPHIC REPRESENTATION
OF PIANO STRING SOUND
WITH ALL HARMONICS
CYCLES)
9
If the molecules of air were large enough to be viewed as black dots, sound frozen in time might
appear as illustrated here. Above: a segmented sphere of frozen sound. The sound source is in the
center. Below, left: a partial section of the sphere above. Below, right: a variable -density motion
picture sound track. These illustrations are repesen'ations of sound in terms of density
Left: a linear representation of sound in terms of
air pressure and distance
from the source. Right:
78 -RPM phonograph record grooves
DISTANCE
Different
ways of
1
CYCLE
visualizing
SO II Ill
Right: a linear representation
of sound in terms of air pressure and time, as measured
from a fixed position
TIME SCALE
Below: sound represented as a series of wave fronts
expanding from a central source
Left: a variable -area motion
picture sound track
FUNDAMENTAL
NOTE
(FROM TUNING
FORK)
FUNDAMENTAL
2ND HARMONIC
3RD
FUNDAMENTAL
HARMONIC
4TH HARMONIC
2ND HARMONIC
RESULTANT
NOTE
RESULTANT
NOTE
The harmonic frequencies which accompany any
fundamental note
from a musical instrument give the note a different character from
the fundamental tone. Instruments are identified by their harmonics
in terms of actual air pressure, to the power radiated at the source of the
sound. Intensity can be measured easily on laboratory instruments, and is a
fixed quantity under equivalent circumstances. The measure of loudness, on
the other hand, calls for the subjective analysis of a human being. Loudness
concerns the ways in which changes in intensity at the source of a sound affect
the sensation of hearing. Going back once again to the illustration of spheres
of smoke representing the expanding wave fronts of sound radiating in all
directions: it is obvious that equivalent portions of these spheres must contain
less and less smoke as they move farther and farther from the source which
generated them. In effect, the smoke becomes thinner and thinner with increasing distance. Just so, the condensations and rarefactions of air in a
sound wave become less and less extreme with increasing distance. In the
case of the smoke spheres, the amount of smoke in each sphere- remains the
same no matter how great the distance from the source. Just- so, the amount
of energy remains the same in each sound wave, no matter how great the
12
FUNDAMENTAL
2ND HARMONIC
HARMONIC
3RD
FUNDAMENTAL
2ND I-I;..^,MONIC
3RD HARMONIC
4TH HARMONIC
7\
5TH HARMONIC
RESULTANT
NOTE
RESULTANT
NOTE
The bottom notes in the graphs above are obtained by adding the
values of different accompanying harmonic frequencies to the same
fundamental note. Pitch remains the same in each case
distance from the source. It is intensity which diminishes with increasing distance in each case, not amount.
A geometric law holds that the surface area of a sphere increases as the
square of its radius. Relating this law to the spherical wave fronts of sound,
it is apparent that the intensity of a sound decreases inversely with the square
of the distance from the source of the sound. In other words, if a sound has
an intensity of 1 at a two -foot distance from the source, the intensity will be
Y4 at a four -foot distance, and 1/16 at an eight -foot distance.
The effect of intensity on the human hearing system is the subjective value
called loudness. The human hearing system responds to very wide ranges of
intensity and frequency, and functions well under complex sound conditions
that make the electric microphone inoperative and useless. It is possible to
turn down the volume of a sound coming from an electronic instrument until
it is at a point of intensity where the ear no longer responds to it. This lower
limit is at the very threshold of hearing; it is called just that: the threshold
of hearing. Conversely, as the intensity of a sound increases, its loudnessthat is, the effect of the sound to the ear-can increase to the point where
listening is painful. This upper limit is called the threshold of pain.
In dealing with loudness, it is well to remember that no two people hear
with the same relative sensitivity. Present-day standards of measurement of
loudness are derived from thousands and thousands of individual tests given
13
L
Relative energy distribution
in an expanding sound
wave. Power diminishes with
the square of the distance
the sound travels from its
source
to humans under standard laboratory conditions. From these findings it is
possible to make some general statements about the nature of the relationships between intensity and its subjective counterpart, loudness.
It has been found that although the intensity of sound may be increased
by equal increments, the human ear will not necessarily register equal increases in loudness. Messrs. Fletcher, Munson, and Weigal, in their work at
Bell Laboratories, have demonstrated that variations in the thresholds of
hearing and pain exist; these are governed by the variations in the particular
frequencies involved.
If the intensity of the sound of a single instrument in an orchestra were
measured, the ratio of its intensity to that of the full orchestra might be
found to be several million times. Imagine how painful it would be if, when
the orchestral sound reached its peak intensity, the loudness registered at the
human ear also increased by several million times. It is at this point that
the importance of the relationship between intensity and loudness becomes
apparent. As the source intensity of a sound increases in multiples of 10,
starting at 10 and going from there to 100, then 1,000, then 10,000, 100,000,
Manmade noises encompass and exceed the total range of human hearing
NOISE LEVEL
IN DECIBELS
NOISE OUT-OF-DOORS
NOISE IN BUILDING
THRESHOLD OF PAINFUL SOUND
AIRPLANE, 1600 RPM, 18 FEET
RIVETER, 35 FEET
ELEVATED TRAIN, 15 FEET
NOISIEST SPOT AT NIAGARA FALLS
VERY HEAVY STREET TRAFFIC, 15 FEET
AVERAGE MOTOR TRUCK, 15
FEET
AVERAGE AUTOMOBILE, 15
FEET
BOILER FACTORY
SUBWAY, LOCAL STATION WITH EXPRESS PASSING
LION'S ROAR, BRONX ZOO HOUSE, 18 FEET
AVERAGE OF 6 FACTORY LOCATIONS
DEPARTMENT STORE
QUIET RESIDENTIAL STREET, NEW YORK CITY,
15 TO 300 FEET
MINIMUM
STREET NOISES, MIDTOWN NEW
YORK CITY, 50 TO 500 FEET
QUIET GARDEN, LONDON
RUSTLE OF LEAVES IN A GENTLE BREEZE
AVERAGE OFFICE
QUIET OFFICE
AVERAGE RESIDENCE
QUIETEST RESIDENCE MEASURED
QUIET WHISPER, 5 FEET
THRESHOLD OF HEARING OF STREET NOISE
REFERENCE LEVEL
ZERO LEVEL
1!
-10-16
WATTS
PER
SQUARE CENTIMETER
150
UPPER LIMIT OF HEARING
140
130
120
110
100
90
80
70
60
50
40
30
20
10
o
10
LOWER LIMIT OF AUDIBILITY
II1
20
20
100
1K
10K
20K
of Fletcher and Munson, indicates the response of the
average human ear to sounds of different frequency at various loudness levels
The chart above, based on the studies
and so on, the sensation of loudness increases in multiples corresponding to
the logarithms (to the base ten) of the numbers given as examples, i. e.,
1, 2, 3, 4, etc. Thus, an intensity increase of 1,000 to 100 corresponds to a
loudness increase of only 2 to 1.
INTENSITY IN HIGH FIDELITY
The total range of human hearing, that is, the range between the threshold
of audibility and the threshold of pain, is about 130 or 140 decibels (a decibel
is a unit on a logarithmic intensity-measurement scale, as discussed previously).
The loudest noise from an orchestra may be 15 to 20 million times as great
as the noise from a single instrument. The range of hearing sensation in an
orchestra may thus reach 75 decibels. In modern electronic -magnetic recording equipment, the maximum reproducible range may be somewhat lower
than this full orchestral range of 75 decibels; methods of compressing this
range onto the tape medium, and later expanding it, have consequently been
developed.
In electronic equipment, inherent factors of noise, both mechanical and
thermal, tend to reduce the maximum possible dynamic range by a background masking effect. Masking is a condition wherein a continuous, or ambient
sound of low intensity, or any irrelevant sound, serves to limit the total loudness
range out of all proportion to its own intensity. The surface scratch of a disc
record is an example of such a masking noise. Later in the text these factors
of dynamic range and signal-to-noise ratios will be mentioned where they
apply to the equipment under consideration.
15
HARD REFLECTIVE SURFACES
SOFT ABSORPTIVE SURFACES
reflection characteristics of different wall and
surface areas hove a great
effect on a sound as heard
by a listener
The sound
REVERBERATION AND EXCLUSION
Sound rebounds from solid objects. The familiar echoes in mountain areas,
the echo of a public address system, the voices of a crowd in a football
stadium; all are part of the phenomena of sound. Almost everyone has had
the experience of entering a room with very hard floor, wall, and ceiling surfaces. A spoken word, or another sound, will bounce off such a surface in a
complex fashion until it is hard to recognize the original, or direct, sound.
And, of course, most of us can recall rooms with heavily draped walls, deep
pile carpets, and acoustically treated ceilings, where most direct sound energy
has been absorbed and little reflected.
Apart from the degree of sound energy absorption, a mountain echo
differs only in time relationship from the reverberation in a "live" room. A
point exists where the ear does not register a sensation of reverberation on
the conscious mind; a short -time echo, or a sound reflection from a nearby
surface, will fulfill this condition.
Imagine that you are seated in a small room with hard-surfaced walls. The
time it would take for a sound to leave your lips, bounce off a wall, and return
to your ears would be governed by your distance from the wall. In the small
room under consideration, sound will not only bounce directly from adjacent
wall areas but will bounce in angles, much like a tennis ball thrown obliquely
at the wall. Let us expand this analogy and think of sound as a cluster of
tennis balls. Uttering a sound in the small room would now be analogous to
exploding a cluster of tennis balls in all directions. If the force of the explosion
was sufficient, all the tennis balls would ultimately rebound to a basket at
your feet; similarly, all the sound would ultimately rebound to your ears. As
one first ball-representing direct sound-was catapulted directly into the
basket, many others would go off in all different directions. Some would make
many bounces, angling from surface to surface before landing in the basket.
Each ball would take a different length of time to come to rest, depending
on distance traveled. Some balls, rebounding only once and then from very
near surfaces, would follow the first ball into the basket with only a few
thousandths of a second delay. If you were to attempt to distinguish the first
ball from the other early arrivals, you would find your choice very difficult.
So it is with sound in our everyday world of audial complexity. Nature
has designed the human hearing system in such a way that it is able to exclude
from recognition those reflected sounds which return quickly, on the heels
of the direct sound from a source. Since the approximate speed of sound in
air is 1,100 feet per second, it is easy to see that the average living -room would
16
140
120
100
80
60
40
20
o
-20
tympani
Ì
snare drum
cymbals
bass viol
piano
violin
bass tuba
french horn
trumpet
«I»
bassoon
clarinet
oboe
flute
piccolo
,. .- ,
9-de--20
50
500
100
1000
5000
10000
20000
CYCLES PER SECOND
The
intensity
levels of
music and
the frequency
ranges
of different orchestra instruments
have many disturbing reverberations or short-time echoes. The important
exclusion period, that is, the period of time a reflected sound may take to
travel to the ear without interfering with the recognition of the direct sound
from a source, has been established to be between °hoop second and 60h000
second. Sound travels 1.1 feet in 1/00o second. Hence, in 5/i000 second it will
17
0
iz)(
An explosion in a cluster of tennis balls. One ball is shot directly into the container. Another
ball reaches the container later, after a series of rebounds, at considerably reduced velocity
travel 5.5 feet. The sound could thus travel to a surface 2.75 feet away and
return without registering an impression on the mind. In the maximum case,
the same could occur with a wall 27.5 feet away, in which case the sound
would travel a distance of 55 feet in 60h000 second. If the human hearing system
did not function in this manner, it would be most difficult to hear even ordinary
conversation in the average room. In larger areas, most sound reflections will
take even longer than 6%000 second, and will thus be heard as ordinary reverberation, adding to the "liveness" of the room.
One very useful characteristic of the human hearing system is its directional
quality. We are able to concentrate our hearing mechanisms, both mentally
and physically, on any desired sound, partially excluding those sounds deemed
undesirable. The human hearing system is highly adaptable, being able to
adjust itself to a tremendous range of audial frequencies and ambient noise
levels.
One illustration of the ability of the human hearing system to exclude
unimportant sounds from consideration is its responses during periods of sleep.
We are jarred into wakefulness only by strange sounds; familiar sounds lose
their ability to capture the attention of the sleeping human.
18
2
HEARING
If we are to believe sound has a history, we must go back in thought to a
time when early man relied only on his senses to protect himself, many
thousands of years before a mechanical means of recording and reproducing
sound existed. In this period of time, the human hearing system probably
evolved by adaptation to man's cultural and natural environment.
Man's sense of hearing at this time was, in effect, an acoustical radar system
warning him against his natural enemies. The vision of primitive man was
probably often obscured by disease and by the dense undergrowth and trees
of his surroundings, which would have limited even perfect vision. His hearing
system, however, protected by long hair and by the placement of his ears,
would have been a protective warning device with a relatively long range.
Animals with two -channel, or binaural, hearing are able to locate the immediate source of a sound quickly, and with fair accuracy. Man must have
combined his directional sense of hearing with his ability to reason in order
to survive attacks by stronger, faster, hungrier animals.
Until comparatively recent times, there had probably always been a functional reason for the production of a particular sound; that is, sounds were
in congruity with their surroundings. Jungle animal sounds were heard by
man only in specific places. Thunder, and other sounds of nature, accompanied
visible physical phenomena. Art forms such as dance and music would have
been specific occurrences; the accompanying sounds were integral parts of those
occurrences. Man's growing intellectual ability must have given him reasons
to differentiate between the various sounds recurring in the small circles of
his activity. New sounds once experienced, however fearful, could be instantly
associated with their sources. Hence, the sound of a child at play would not
cause alarm, as might the roar of a tiger.
Placement and localization of sound sources must have begun very early
in man's development. The snarl of a wild animal or the footstep of another
man could be positioned instantly in relationship to the listener. In those
dangerous times, when a sound was heard the mind of man simply activated
the muscles on the basis of recognition of friend or foe. Defensive acts or
friendly recognition, depending on the circumstances, then took place.
A man can locate a source of sound in relation to his own position in
several ways. A person with defective hearing in one ear, for example, can
locate a sound by virtue of the differential directional sensitivity of his good
19
Above: loudness varies with
direction. A "white," or
mixed -frequency, sound of
constant intensity will appear equally loud to the left
ear at any point on the
periphery of the colored
area. Left: locating the direction of a sound source
monaurally. The head
is
turned until the single ear
receives the sound at maximum loudness. Since turning the head takes time,
sounds of short duration
cannot readily be located
monaurally
20
LOUDNESS
(íEQUAL
PHASE IS UNEQUAL
LOUDNESS IS EQUAL
PHASE IS EQUAL
Locating the direction of a sound source binaurally. When a sound is at an angle to the head,
right, loudness and phase are different at the two ears. Turring the head to make them equivalent,
left, places the eyes in position to observe the source
ear. To the one -eared, or monaural listener, the loudness of a sound of constant
intensity at a fixed distance varies with its direction in relation to the ear.
The position of maximum loudness of a sound is usually about eighty degrees
toward the ear from the center of the nose. Thus, by turning his head until
loudness is at maximum, a one-eared man can judge the direction of a
source of sound fairly accurately.
The chief objections to the one -eared method of locating a source of
sound are, first, the sound must be fairly constant and of long duration, and
second, a good deal of time must be spent in turning the head back and forth
to bracket the position of maximum loudness. The monaural listener of primeval
times could thus be attacked and half devoured while he was still trying to
locate the source of impending danger.
The two-eared, or binaural listener is at a considerable advantage in this
respect. Sounds generated at an angle to a line bisecting the head, between
the two ears, will be of unequal loudness to the two ears and, because of the
physical distance between the ears, will be out of phase as well; i. e., a time
lag will exist between the responses of the two ears. Thus, sounds of very
short duration may be located quickly by the two-eared listener through
a practically instantaneous process of mentally comparing the differences in
the sounds received by the two ears.
Binaural hearing must have been very important to primitive man in that
many of the sounds of danger were probably of very short duration, preceding
attack only by seconds. Man's-rapid perception of audial direction may have
provided him many narrow escapes.
21
Locating the distance of a source of sound in an open, non -reverberant area is difficult unless
the sound is a familiar one of known loudness or unless it is very near. Loudness and phase
relationships remain nearly the same in both ears regardless of distance. Angling the head,
right, alters the relationships slightly and may help
Since practically all the sounds of prehistoric times would have had a
direct bearing on the people who heard them, it would have been natural for
people to support the evidences of their senses of hearing with their vision.
Thus, people probably turned to face each and every sound, bringing their
eyes to bear on the sources. One reason for this assumption is this: while
man's audial perception of direction is good, his audial perception of distance
is rather poor, unlike that of some animals, at least in open spaces without
reverberations. Thus, man may have relied on his ears for initial perception
and for perception of direction, and then on his eyes for final recognition and
for perception of distance.
As man's intelligence grew, mental sorting of the many sounds reaching
his ears must have taken place. Surrounded by the harmless sounds of nature,
domestic animals, family, and neighbors, man doubtless began to concentrate
on his progressive endeavors without feeling the need for constant head motion. Man's adaptation to a world of dimensional sound began to take place,
and it became natural for him to hear many sounds simultaneously.
In part, progress in religious ceremonies and various forms of entertainment would force man to develop a new manner of using his hearing.
Whenever he was a spectator and heard many sounds at one time, or in quick
sequence, he would have had to overcome his instinctive impulse to move his
head, orienting himself to each sound. We can only speculate that this is what
happened, but the facts at hand seem to lead to this conclusion. Man must
have grown more and more accustomed to these multiple, complex sound
situations as his role as spectator became increasingly important. It was then
that his cultural conditioning must have overcome his instinctive listening
habits.
As a modern example of a spectator situation, let us visualize a man attending a musical concert. In front of the man are the hundred members of a
22
LEFT
EAR
CULTURAL\
CONDITIONING
TO NO
ACTION
/
HEAD
TURNING
ACTION
CONTROL
CHEARING
EVALUATION
CENTER
CENTER
RIGHT
EAR
/
>
\ /
o
9,
--
Cultural conditioning can overcome the normal instinct to move the head to face a source of sound.
One path is followed when protective head movement is called for. The other path is followed
when identification is unnecessary
symphony orchestra, each with an instrument capable of making noise. During
the course of the concert, some of these instruments will function together,
some singly, and some in quick sequence. Though by instinct the listening
man may wish to locate each sound source, he realizes that it would be impossible for him to do so. As a passive spectator, the listener's cultural conditioning overcomes his instinct. His instinctive head movements, formerly
catalyzed by signals from the part of his brain controlling his actions, are
inhibited by his cultural conditioning, which interrupts these signals if no
action is necessary. It is the inner tension caused by this series of action
signals and reaction cues that gives a feeling of dimension and spatiality as
we listen to music. It must be remembered that this discussion has not dealt
with the acoustical reasons for the feeling of dimension caused by direct
and reverberant sounds; the authors have merely conjectured about the psycho acoustical development of human hearing as accurately as they can.
23
Abbre'dations
Multipliers: For the most part, the following are
= wavelength
%
used to denote numerical quantities less than 1;
= impedance
when capital letters are used, the reverse is true.
Z
d=deci=1/10
a.c.
= centi = 1/100
m = milli = 1/1,000
Fc = micro = 1/1,000,000
u µ = micromicro = 1/1,000,000,000,000
(k) = kilo = 1,000
M = meg = 1,000,000
t
= alternating current
d.c. = direct current
of
=
if
= intermediate
rf
= radio frequency
vhf
K
Units of measure:
A (a)
= ampere
=
MA (ma)
p,
A
(»a)
measures of
milliampere
= microampere
current flow
1
= farad
= microfarad
f) = micromicrofarad
N -M(f)
F
F(µ f)
µ
1-4
F (
(h)
H
= henry
mH (mh)
=millihenry
p,
h)
H
(,u
=
(
units of
capacity
1
units of
inductance
microhenry
units of
= ohm
M SL = megohm f resistance
units of
V = volt
electromotive
MV (mv) = millivolt
force
KV (kv) = kilovolt
W = watt
units of power
MW (mw) = milliwatt f
m = meter
iL
= centimeter
mm = millimeter
audio frequency
= very
high frequency
= ultra high frequency
r.m.s. = root mean square
db = decibel
dbm = decibels of power referred to milliwatt
cps (c/s) = cycles per second
Kcs (Kc/s) (kcs) = kilocycles per second
Mcs (Mc/s) (mcs) = megacycles per second
uhf
1
A.V.C.
= automatic volume
A.F.C.
= automatic frequency
control
AM
= amplitude modulation
FM
= frequency
ips
=
AES
=
MRIA
control
modulation
inches per second
Audio Engineering Society
= Magnetic Recording Industry
Association
NARTB (NAB)
= National
Association of Radio
and Television Broadcasters
cm
24
frequency
RIAA
=
Record Industry Association of America
3
.tieII'LIFICA'l'IU \í
The first time someone said to another person "What did you say?" the
need for amplification was established. How many thousands of years ago
this may have been is anybody's guess. Little indication exists that man ever
had any desire to send the sounds of speech over extremely long land distances.
He seems to have been content with cupping his hands, or, in later seafaring
days, with using a speaking trumpet to communicate over a distance just
greater than cannon range. When a man was beset with deafness, a hearing
trumpet was his only recourse in collecting enough sound to hear. Strictly
speaking, none of these expedients really amounted to what moderns would
call amplification, although by different means they accomplished much the
same thing. When the term amplification is applied to any of our physical
sources of energy, the addition of an external source of energy is implied.
The difference between shouting and whispering is not amplification as we
use the word. (In raising the intensity of his voice, a man simply moves more
air than before.) As a man speaks into a modern microphone, the voice signal
is fed into a transistor, or another electronic device which operates on some
source of electrical power; this voice signal is then fed to a loudspeaker,
which duplicates the words aloud with an intensity a thousand times greater
than the original voice power. This has been accomplished through amplification; the increased power of the voice was not caused by shouting, but by the
external electrical power applied to the transistor or other amplifier system.
It was not always so; until the early part of the 19th century no means of
amplifying the human voice existed; man had to be content with raising his
voice or using an ear trumpet. By the end of the 19th century men had found a
method of amplifying sound by the use of externally introduced compressed
air. It was the hope of these men that compressed -air amplifiers would provide a means of increasing voice and music volumes for large gatherings.
Thomas Edison and Dr. Chichester Bell, working independently, both devised systems of this type. Edison called his machine the aerophone. Later,
Charles A. Parsons, English inventor of the compound steam turbine and of
non-skid automobile chains, perfected a device he called the auxetophone, working on the same principle of sound amplification through the addition of compressed air. Robert L. Gibson and W. M. Dennison, both of the Victor company, adapted this amplification system to the disc phonograph. The French
Pathé company tried to couple this device with a new medium, motion pictures,
in order to turn out talking pictures in 1904.
HUMAN LUNGS SUPPLY
COMPRESSED AIR
SPEAKER WHOSE VOICE
IS TO 8E AMPLIFIED
MEGAPHONE
MOUTHPIECE
VOICE -OPERATED AIR VALVE
Above: the compressed air
amplifier called the aero phone by its inventor, Thomas Edison. Right: a modern
compressed air loudspeaker
operating on the same principle, but with an electrically
operated valve
SIGNAL -OPERATED
AIR VALVE
ELECTRICAL
COMPRESSED AIR INPUT
SIGNAL INPUT
All these compressed air devices functioned along the same lines. A supply
of compressed air was forced through a tube and against a small valve. This
valve, which interfered with the passage of the air, was operated by the voice
power of a speaker. As he talked into a mouthpiece, the valve opened and
closed according to his voice modulations. The force of the compressed air
was the power amplifying the sound. When such a compressed -air amplifier
was connected to a phonograph, the vibrating stylus opened and closed the
air valve, and the compressed air took the place of the air-pushing diaphragm.
With a pressure of only 3 pounds per square inch in the sound box, the increase in volume was considerable. However, due to poorly conceived mechanisms, the compressed-air amplifier of the early 1900's was not commercially
successful. The compressed -air amplifier is still used today in situations where
a limited range of frequencies needs to be amplified with an extraordinary increase in power. The most effective air raid warning sirens in existence today
are compressed -air amplifiers, rather than the conventional motor -driven rotary
sirens.
The wonderful 20th century began, and man's technical knowledge was
on the upsurge. It was the beginning of the air age and the electronics age;
radio and television, though not yet practical, had at least been conceived.
In 1904 Sir John Ambrose Fleming announced to the world the invention of the
two -element vacuum tube, with its uncontrolled flow of electrons. Fleming
found that if you could electrically heat a cathode element in an evacuated
glass bottle, electrons would bubble off and be attracted by a plate element in
the same bottle. He could control the current flow to a limited extent by
varying the voltage relationship between the two. The major discovery in
26
this invention was that electrons could flow across the distance between the
cathode (negative element) and the plate (positive element). In effect, this
device of Fleming's was a valve for electrons, and current flow was either on
or off, without much control in between. A few years later, Dr. Lee De Forest
took advantage of Fleming's device by inserting a control element between
the two basic elements. This third element De Forest called a grid; in physical
configuration it was just that, a grid of fine wire mesh which could be electrically charged, and used to control the flow of electrons. He found that by negatively charging this grid in relation to the cathode element he could place a very
small electric signal on the grid and cause it to be amplified into a much larger
signal in the output of the tube. This first tube, called the De Forest Audion,
marked the beginning of the electronics age.
The invention of the vacuum tube did not bring about startling changes in
sound amplification techniques; in fact, it took almost another 20 years before
amplified audio systems were recognized as an important part of radio broadcasting. Until this time the vacuum tube was considered only as a source of
high frequency vibrations (radio waves), much as the tuning fork was its
counterpart in sound vibrations. The tube provided an easy non -mechanical
means for producing these electromagnetic wave motions and amplifying them
to the great power necessary to broadcast. Later, their use was necessary
for home radio reception; still, the audio amplifier was the least part of a radio
of that day. Many who are working in the field of electronics at the time of
publication of this text can review with ease the life span of the conventional
vacuum tube. It came to life in the late 20's and 30's, enjoyed widespread use
in World War II, and now we see its gradual decline in use, supplanted by
the transistor, magnistor, and other devices designed to control the flow of
electrons.
HEATED FILAMENT
PLATE
ELECTRON
FLOW
<,
i
LOAD
RESISTOR
Left:
sketch and schematic
diagram of Flemings electron valve
GLASS
ENVELOPE
I''IiliI
CONNECTING
WIRES
FILAMENT
PLATE
BATTERY
BATTERY
GRID
FILAMENT
PLATE
Right: the De Forest Audion
tube, which employed a grid
element in addition to the
other two
GLASS
ENVELOPE
27
6J
lQJl9A4J
römi
.:»
d
4c
3
°1
8 c
c
;
«
CO
C.a
F°
fo
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ú m
e
"
`O1
ÿca
cÑ
a
<
c
CT
oe
V
v
e
c
oÿv
c
a'v=o
c«
.a
S
áó
Wa
01oa-.
E
M
>
3i
V
v
1o
3
vv
ÿ-
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<
D
-jQ
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a
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CHART OF ELECTRONIC SYMBOLS: some of the common electronic devices pictured in the photograph at right are illustrated here in schematic form. The key numbers refer to the photograph
28
Vacuum tubes:
Semiconductors (transistors, etc.):
Diode (2 -element)
P -N
Dual diode (rectifier)
N -P -N transistor
Triode (3 -element)
P
Dual triode
N -type uni junction transistor
Pentode (5 -element)
P
Multiple section
N -P -N tetrode transistor
Beam power amplifier
PI-N triode transistor
P
Toroidal coil:
-P
transistor
-type uni junction transistor
-N -P tetrode transistor
-1-N
diode transistor
24
Loudspeakers:
Rotary selector switch:
25
Permanent magnet
Electromagnetic
29
ELECTRICAL SIGNAL TO AMPLIFIER
REPRESENTED BY SINE WAVE
+MAXIMUM
r'
/r
Above: actual placement of
electronic parts around a
tube socket in an amplifier.
Right: conversion of mechanical stylus motion into electrical signal energy
i
-MAXIMUM
STYLUS MOVEMENT
FLEXES CRYSTAL
A MAXIMUM
STYLUS MOTION
REPRESENTED BY SINE WAVE
, \'\\WOMBS.
B
MAXIMUM
RECORD GROOVES
The new types of amplifiers which incorporate the use of transistors,
magnistors, and nuclear- and solar -powered elements accomplish the same feats
as the vacuum tube amplifiers, but eliminate the necessity for special filament
heating supplies, unreliable construction, and over-all bulk.
HOW A VACUUM TUBE AMPLIFIER WORKS: The purpose of any amplifier is to increase the energy level of the signal applied to its input circuit.
In high fidelity sound reproduction, we are concerned with several different
input signals. These signals may come from a phonograph cartridge, an FM
tuner, a microphone, a tape recorder, or a television set. If these signals could
be fed directly to a loudspeaker there would be no need for audio amplifiers.
Unfortunately, these signals are much too weak to power a loudspeaker. The
only requirement of an audio amplifier is that it increase the signal level, without changing the character of the signal. If the amplifying circuit changes
the wave form of the signal, and/or adds noise to the sound, then it fails to
do its job.
Let's see what happens to the small signal from a phonograph cartridge
as it is amplified to an audible volume. In tracing this signal from the input
circuit of an amplifier we will see what undesirable things may happen if the
circuit does not operate correctly. Let's begin with the operation of a simple
audio amplifier using triode (3 -element) tubes. Many of the tubes used in
The input vacuum tube
circuit with its component parts in schematic form.
signal relationship is indicated
The
input-output
INTERSTAGE COUPLING
CAPACITOR
INPUT SIGNAL
V
-
r
TRIODE TUBE
.
,)
[-I
SIGNAI OUTPUT
TO NEXT TUBE
PLATE
GRID
TONE ARM
PLATE LOAD
.RESISTOR
CATHODE
GRID LOAD
10V
HEATER
RESISTOR
INPUT CABLE
TO FILAMENT
.-SUPPLY
TO B + PLATE
CHASSIS GROUND
CATHODE RESISTOR
30
--I
RELATIVE
OUTPUT SIGNAL o
I VOLTAGE SUPPLY
`
today's modern amplifiers have more than three elements for special purposes.
For ease of understanding, the basic diagrams here represent the operation
of all vacuum -tube circuits. Later in this chapter the operation of other types
of amplifiers-those employing semi-conductors (transistors, etc.) and magnetic
elements-will be covered briefly. In essence, these devices function much
as do vacuum tubes.
The phonograph cartridge produces a minute electrical signal voltage
which varies directly with the mechanical variation in the record grooves. The
coupling device between the disc and the cartridge unit is the stylus. Just as
a page of sheet music can represent, in a passive way, the music on the disc,
so the music on the disc can be converted to a representative electric signal
voltage. This voltage would be too small to be heard if it were directly connected to a loudspeaker. It is the job of the amplifier to increase this electrical
signal level to a point where it will be able to "drive" the loudspeaker. Though
it would seem logical simply to fit together any cartridge, amplifier, and loudspeaker units, it is not. Certain problems of electrical matching occur. Much
as it is the job of a plumber to fit various sizes of water pipe together with
an understanding of specific plumbing requirements, matching is one of the
jobs of an electronics engineer. It is his responsibility to design and construct
these audio units in such a way that they will fit together or match, so that
they will accomplish their purpose most efficiently. This operation involves,
primarily, a matter of matching the opposition to the flow of current, or
impedance, of the various components of a high-fidelity system. One problem
which often arises in matching is that impedance usually varies with frequency.
The manufacturers of high-fidelity equipment, taking their cues from their
engineers, label all their equipment in such a way that the interested novice
can make his connections with as little technical knowledge as possible. While
the various stages of amplification may be discussed here as though they involved individual tube units, such is not the case. These various circuits are
combined into one or more chassis provided with accessible standard jacks and
terminal strips for easy connection of the components.
In the sketches and diagrams given here you will find standard symbols
used to designate specific electronic parts. Circuit diagrams for engineers are
composed entirely of these symbols; while the circuits themselves may be
complex, the symbols are not. A chart of most of the symbols designating
electronic components found in the average high-fidelity system is given here.
First comes the symbol, then a sketch of the part as it might be used in pictorial do-it-yourself kit diagrams, and in some cases a photograph of the part
itself.
From our earlier discussions on sound we know that a sine wave can be
used to designate an audio signal. As a turntable rotates with a stylus in the
groove of a record, the physical variations in the groove cause the stylus to
move back and forth laterally. It is this movement of the stylus in the phonograph cartridge that produces the audio signal voltage we have represented
with a sine wave. Cartridges can be obtained in a wide variety of styles
and types.
Output signals vary widely. The crystal or ceramic units have high output
signal voltages, while the magnetic varieties have lower output voltages and
often require additional amplification. Let's assume that the unit used in this
explanation has an output signal voltage which varies from 1/10 -volt in quiet
musical passages to a maximum of 1/2 -volt in the loud sections, and ignore
the range of frequency response at this time. The signal is being fed to the
grid of a triode vacuum tube. In operation this tube is supplied with a small
a.c. filament voltage. This filament voltage is sufficient to heat up the negative
element of the tube: the cathode. In many tubes the red glow of this heat is
visible and during operation many tubes become hot enough to scorch paper.
Along with the filament or heater supply, the plate or positive element of the
31
THIN METALLIC CATHODE BASE
WITH THORIATED TUNGSTEN COATING
°
°9
A stylized sketch of the
e
internal operation
0
e
e
eee
e
°
e
FLOW
'e
oe e e
e e
e
e9
eé e
;
°
o e
ee e e
000
e
emitted from a hot cathode are attracted to a
plate element across a
partly evacuated space
a e
ELECTRON G
.,
of a
diode tube. Electrons
e
e
e
CERAMIC
FILLING
POSITIVELY
.
CHARGED PLATE
EMBEDDED HEATER
FILAMENT WIRES
tube is supplied with a high positive voltage; several hundred volts d.c. In
the construction of the vacuum tube the manufacturer has coated the cathode
with a material called thoriated tungsten which, when heated, gives off many
electrons in a random manner, much as grease bubbling off a very hot frying
pan. In their natural state the negative electrons are attracted to the highly
positive plate in the same fashion as the grease from a frying pan might
collect on a blotter held above it. This movement of electrons across the
space between the cathode and the plate causes current to flow in the circuit
connected to these two elements. Without a grid, this flow would go on at a
steady rate, dependent upon the supply voltages and the configurations of
the tube elements. The flow can be controlled simply by placing an electrical
charge on the grid. Negatively charged objects repel each other. From this
it is easy to see that if the grid were charged negatively, the negatively charged
electrons coming off the cathode would be repelled right back to the cathode,
leaving the vacuum tube in a condition of inactivity (called cutoff). By placing
The
internal operation of a triode tube. Some of the electrons which would otherwise be
attracted to the plate are repelled by the varying negative charge of the grid
o
GRID CHARGE VARIES
IP
1
0
.,,.00 o 0
o
o
.0
..o°°
oo
° 0
Q0
0 0'_
dI
°"ooó ó 0
'
GRID
WIRE
,
dI1111
0 ó
MAXIMUM °.
FLOW
o
o
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0 ° °,I
O°`'
°
0
O
32
0 0
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FLOW
oo0oe
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ACTUAL POSITIONS
OF TUBE ELEMENTS
o
SEEN FROM TOP
o
OUTPUT SIGNAL
-
GRID BIAS CENTERED
ON LINEAR SECTION OF
CHARACTERISTIC
GRID BIAS NOT
CENTERED ON
LINEAR PART OF
CURVE
-
1
__
- -T
-DISTORTED 1-OUTPUT _\_
SIGNAL
V
SIGNAL CUT OFF
GRID BIAS
VOLTAGE AXIS
INPUT SIGNAL
LESS
MORE NEGATIVE
INPUT SIGNAL TOO LARGE
FOR TUBE CHARACTERISTICS
NEGATIVE
I
The operation of a linear
amplifier vacuum tube. Left, proper operation. Right, improper usage
a fixed balancing voltage, called bias, between the cathode and the grid connections of the tube, the degree of this negative charge on the grid may be
varied. It is necessary to keep the grid always at some degree of negative charge,
since in swinging into a positive charge the grid would attract electrons, losing
its purpose in the process. The grid of the tube is biased with a fixed negative
voltage about half -way between zero voltage and the lowest negative grid
voltage at which the tube would cease to conduct, that is, be at cutoff. As
the signal voltage output from the phonograph cartridge is allied to the grid,
the alternating -current nature of the signal causes a constant variation of the
negative charge on the grid. As the signal voltage becomes positive, it decreases the negative charge on the grid; then, as the signal voltage swings
into the negative portion of its cycle, it adds to the negative charge on the
grid. As these changes take place, more or fewer electrons are allowed to
flow between the cathode and the plate, in direct relationship to the variations
caused by the very small signal voltage on the grid. In this manner, we are
able to control a very large flow of electrons with a very small grid signal.
These amplified signals are passed on to the next stage of amplification
through the resistor-capacitor circuits connecting the two stages. While it is
possible to gain tremendous increases in signal level, it is sometimes necessary
to add several tubes together to effect the maximum level needed for the
operation of a loudspeaker system.
ARROWS INDICATE DIRECTION
OF CURRENT FLOW
How a signal is transferred from tube to tube
resistor -capacitor
in
a
coupled audio amplifier.
As output signal current
varies through the plate
load resistor a corresponding change in voltage drop occurs across
the resistor. This changes
the charge of the coupling capacitor, producing a signal effect on the
grid of V 2
INPUT SIGNAL
V
I
±
PLATE LOAD
RESISTOR
AMPLIFIED
OUTPUT
SIGNAL
GRID
RESISTOR
-F
GH -VOLTAGE
POWER SUPPLY
H
=
-
H GH -VOLTAGE
POWER SUPPLY
33
While many specific details have been left out of this simplified explanation of vacuum tube amplification it is sufficient to explain the operation of
most vacuum tubes used in the reproduction of high-fidelity sound. The application and arrangement of tubes and circuit components for specific jobs
in high-fidelity sound are grouped into several general amplifier classifications:
control or preamplifiers, equalizer or tone -control amplifiers, power amplifiers,
etc.
DISTORTION IN AMPLIFIERS: We have traced a signal through a single
triode vacuum tube stage of amplification, finding that the output signal is
identical to the input signal except for the increased magnitude. It would be
very nice indeed if such high-fidelity amplification took place in every circuit,
but this is not the case. Many disagreeable changes and additions to the signal
take place in practice. Most listeners are familiar with a 60 -cycle hum sound
coming from their loudspeakers, especially noticeable during soft musical
passages. Other noises are often audible; some are caused by faulty parts,
loose connections, or by the tubes themselves. All these distractions, whether
mechanical or electronic, cause small signal voltages of their own to be added
to the desired signal. In the course of normal amplification these noises are
amplified right along with the signal.
Every high-fidelity component can be plagued with unforeseen part failures,
causing noise, but these noises are inherent in some systems due to poor design. While noise and hum add small disturbing signals of their own to the
desired signal, distortion in its several forms changes the desired signal on its
way through the amplifier. As distortion increases in an amplifier, it lessens
fidelity in the output signal. Distortion can be classified as to type. The major
types of distortion are: harmonic, intermodulation, frequency, transient, and
phase.
HARMONIC DISTORTION: Earlier in the text, the nature of sound from
musical instruments was mentioned. Even though two instruments may play
the same note, the resulting sounds will be different. Each instrument has its
own sound characteristics because of its harmonic content. Harmonic distortion exists to some degree in every high-fidelity amplifier. It is as if each amplifier had its own harmonic structure. Harmonic distortion occurs after a
signal is put into an amplifier; it is the amount of harmonic energy formed as
the signal passes through the various stages of the amplifier. This type of
distortion takes place because of the difficulty of maintaining linearity of
amplification in both the vacuum tubes and the circuit components connected
between them. Going back to the diagram showing amplification in a simple
triode tube, we can see how the sine wave signal applied to the input is amplified without change. However, in faulty circuits or in poorly designed ones,
the input wave is distorted within the tube or its associated electronic components. Earlier, the balancing voltage called bias voltage was mentioned.
The
addition of various harmonic energies will change the form of an amplified sine wave
output signal. The resulting effect on recorded music is intolerable
AUDIO AMPLIFIER
34
^,
LOW -FREQUENCY
COMPONENT
LOW -FREQUENCY
SIGNAI
INPUT
Intermodulation distortion is an effect of one
signal on another. A
high -frequency signal
may be modulated, or
"beat," by a low -fre-
OUTPUT
HIGH -FREQUENCY
COMPONENT
HIGH -FREQUENCY
SIGNAL
quency signal
MODULATION
ON HIGH FREQUENCY
EFFECT OF LOW FREQUENCY
NO INTERMODULATION DISTORTION;
HIGH -FREQUENCY SIGNAL UNAFFECTED
Should this bias voltage be the wrong value, too high or too low, the variations
of the signal on the grid would not cause corresponding changes in electron
flow. Should the wrong vacuum tube have been chosen by the designer of an
amplifier, then this situation could have occurred in a different way. This is
one of the reasons there are so many different types of vacuum tubes avail-
able; each has been designed for a specific job. Harmonic distortion can also
take place in the circuits between the various tubes in an amplifier. Another
frequent cause lies in the fact that for increased output power the signal on
the input grid of a tube is sometimes swung too wide, exceeding the linear
amplifying power of the tube. New types of high -power output tubes have been
designed to meet the need for more audio output power in modern high-fidelity
amplifiers.
INTERMODULATION DISTORTION: There would be very little trouble
with intermodulation distortion if audio amplifiers were called upon to amplify
only one note at a time. However, most of the signals impressed onto the input
of an amplifier are of a complex nature: many notes at one time. As these
complex frequencies and their harmonics are amplified in a system with even
the slightest non -linearity, beats occur between them. In the ordinary performance of music, these sum -and-difference frequencies regularly occur, giving a pleasing character to the sound. In an amplifier having intermodulation
distortion, these so-called beats produce non-musical sounds at the loudspeaker.
Intermodulation distortion is one of the most difficult for the human ear to
hear: it can occur in natural sound, in a record or tape, in an amplifier, or in
a loudspeaker system.
Motion pictures were the first to employ a limited frequency range in
audio equipment, but within that range there was as little harmonic and inter modulation distortion as was possible to obtain. As the frequency response
of audio amplifiers was widened, the small amount of intermodulation distortion became more noticeable. The improvement of high -power output tubes
and transformers has tended to limit the rise of intermodulation distortion as
the power of amplifiers rises. Both manufacturers and consumers have found
that intermodulation distortion rating of amplifiers is justifiable as a standard
of quality, providing the method of testing is standard.
35
20
f
CPS
20,000
20
CPS
CPS
11
20,000
CPS
y1f
Frequency distortion is
caused by amplifiers with
frequency limitations at
the ends of their response
curves
FREQUENCY DISTORTION: While seldom referred to as frequency distortion, it is this factor that governs the frequency response of an amplifier. The
frequency response range includes all signals between the very lowest note
and the highest. In the perfect amplifier all these notes (frequencies) would
be amplified the same amount. In most audio amplifiers without any tone
compensation, both the low and high notes drop off in amplification. In the
standard vacuum tube amplifier, the middle-range frequencies are most easily
amplified. Frequency distortion occurs because of the effects of natural
capacity existing between elements of a vacuum tube or elsewhere in the
circuit components. Going back to the basic amplifier circuit, these points can
be made clear.
In most resistor-capacitor coupling circuits, the plate of the preceding tube
is connected to the grid of the next tube through a capacitor. At low frequencies, the capacitive reactance of this capacitor becomes large enough to
limit the signal effect on the grid. The middle frequencies are not affected
by this capacitor and are fully amplified. At high frequencies, the combination
of inter -electrode capacitance, cathode by-pass capacity, and the capacity of
the plate circuit of the tube tend to load down the output of the tube with
their combined low impedance at high frequencies. Inter -electrode and plate
capacity exist due to the physical construction of the tube elements and the
fact that a capacitor is simply two metal conductors separated by a non-conductor. Tubes are not planned in this fashion; the capacity exists due to the
proximity of the metallic elements in the tubes.
High-fidelity audio amplifiers can be designed to compensate for these
failures. Careful choice of design features and components and special vacuum
tubes go a long way to eliminate frequency distortion. The set owner is
provided with bass and treble controls so that he can compensate for these
limitations as he desires.
TRANSIENT DISTORTION: Transient distortion must be limited if an
amplifier is to have good audio response. The complex sounds of speech, music,
and noise change at a very rapid rate. This transient nature of sound raises a
problem in amplification. Most amplifiers can easily pass the steady frequencies
of sound, but in the case of transient portions of those sounds a series of
+10
The
L
frequency distortion
effects
caused
limitations
of
by
the
circuit
components are indicated
by the curve
LOW RANGE
?6
MIDDLE RANGE
HIGH RANGE
minute oscillations may be set up. These oscillations occurring in transformers
or other circuit elements are called transient distortion. Its elimination is
mainly a problem of correct amplifier design.
PHASE DISTORTION: Phase distortion, through discernible, is often obscured by other forms of distortion. It is caused by flaws in the design of an
amplifier. It is a major cause of loss of articulation in recorded speech, and
fuzzy or mushy sounds in percussive passages of music. If an amplifier has
little or no phase distortion, all signals from the very low to the very high
in frequency will take the same length of time to pass through the amplifier.
Definition is lost when phase distortion, caused by an unequal transit time
between the high and middle -range frequencies, exists. If the high-fidelity
amplifier is designed to operate with an extended high -frequency range, it is
likely that phase distortion will be at a minimum.
CONTROL AND PREAMPLIFIERS: Control or preamplifiers serve several purposes in modern high fidelity systems. Since the sources of high fidelity
include not only radio and phonograph records, but perhaps a tape recorder
and the audio from a television set as well, the preamplifier must have some
means of switching in these various signals. A control or preamplifier is, in
essence, an amplifying device which takes these input signals and increases
their power sufficiently to drive the power amplifier. However, in this process
of amplification, there must be a facility for changing the volume level of the
sound and its frequency characteristics through either a record equalization
switch or by the treble and bass controls. In some types of preamplifiers a
loudness control is included which provides for differential volume changes according to frequency, to better suit the response of the human hearing system.
The progress of sound signals through a conventional control or preamplifier is
illustrated here. Though there are several input circuits, each input signal
is sent through the same outlet into the power amplifier. The particular path
Block
diagram of
INPUT JACKS
TUNER
a
preamplifier. The colored arrows indicate the direction of signal flow
through the circuit
INPUT LEVEL
CONTROLS
(1 THROUGH 6)
INPUT SELECTOR
SWITCH
TAPE
CRYSTAL
PICKUP
VOLUME
CONTROL
LOW-LEVEL
CONTROL
LEVEL CONTROL
MAGNETIC
PICKUP
HIGH-LEVEL
MAGNETIC
J
OUTPUT SIGNAL
TO POWER
AMPLIFIER
OUTPUT SIGNAL
FOR TAPE RECORDER
PICKUP
CRYSTAL
MICROPHONE
37
:
_'r,:...r r.r.- 5t;r.a
The English -made
preamplifier
features extremely
Leak
neat parts arrangement
UI
Equipment
1
u;ili,iu turni.hu<l
h' I;'iti.n
llu.u'i
Inc.
chosen depends upon the position of the selector switch. A self-contained
preamplifier which is on a chassis separate from the power amplifier is shown.
This is not always the case; some preamplifiers are incorporated on the same
chassis as the power amplifier. Some are separate, but still employ the voltage
supply from the power amplifier, while others may be included in an FM
tuner. However they are constructed, preamplifiers do the same job in each
case. The lower -priced units are all -in-one and the higher -priced are on
separate chassis. This explanation of the various inputs, and of the switching
and control circuits, is greatly simplified.
MAGNETIC PHONOGRAPH INPUT: The signal from a magnetic phonograph cartridge is very weak in amplitude. In order to amplify this small
signal voltage sufficiently, an additional vacuum tube amplifying stage must be
employed. When the selector switch of the preamplifier is in the position
marked Magnetic Phono, the signal from the cartridge is sent into an amplifying
An amplifier of compact
design featuring both preamplifier and power ampli-
fier on one chassis
Equipment for evaluation furnished
by Sherwood Electronics, Inc.
R-10
SIGNAL OUTPUT
C -SA
MAGNETIC
INPUT
'^
Left: a phonograph pickup
preamplifier circuit diagram.
V1
The colored components in-
dicate the high -frequency
peaking circuit sometimes
employed. The dashed lines
enclose the low -frequency
TO
FILAMENT
FILAMENT
SUPPLY
SUPPLY
(6.3 V)
(6.3 V)
compensation circuit for
TO
i
magnetic cartridges
-
B+ 80 TO
100 V.D.0
INPUT SIGNAL
INPUT LEVELCONTROL
VOLUME
CONTROL
OUTPUT SIGNAL
INPUT JACK
Right: block diagram of an
input circuit for crystal or
ceramic pickup cartridges
TO
OTHER
STAGES
SELECTOR SWITCH
IN CRYSTAL POSITION
CRYSTAL OR CERAMIC PICKUP
stage (represented by a block on the diagram), and then from there into the
rest of the preamplifier. The relative amplitude of the various signals is
shown by the height of the sine waves. As a signal progresses, you can see
the relative voltage increase in each vacuum-tube stage. However, in some of
the vacuum -tube stages there is no amplification, since these circuits are employed as equalizers (tone controls). In some of these circuits a signal loss
is encountered which must be made up in later stages. In the amplifier stage
for the magnetic cartridge, certain specific circuit arrangements of the component resistors and capacitors are made to compensate for the peculiarities
of magnetic devices. In most cases, however, this circuit arrangement is standardized for the average magnetic cartridges.
In the recording and processing of modern phonograph records, each record
manufacturer adopted one of several recording systems. These various systems are classified as industry recording curves. The curves relate to the
amount of change in bass and treble frequencies that occur during the actual
cutting of a disc master. Some of these changes are due to the inherent nature
of the magnetic cutter and others are planned changes, designed to increase
the frequency range recorded. Only recently has standardization in the recording industry to the RIAA curve come to pass, so high-fidelity equipment
manufacturers must still equip their devices with other recent popular record
equalization curves. The equalization control is usually a rotary switch with all
39
Left: an amplifier input selector switch showing various input positions. Right:
an internal view of the
selector switch
Equipment for evaluation furnished
by H. H. Scott, Inc.
input channels clearly marked. As the switch is rotated through various positions, the channels are selected through connections made within the inner
sections of the switch. On these selector switches may be four or five phonograph positions, depending on the manufacturer's readiness to provide record
equalization for whatever records might be in your collection. On the block
diagram you will see how the output of the magnetic cartridge amplifier goes
into this switching network. Since this particular preamplifier is equipped
to use a crystal or ceramic phonograph cartridge, with its higher signal voltage,
a choice switch is used to by-pass the unnecessary magnetic-cartridge amplifier
tube.
THE CRYSTAL OR CERAMIC INPUT: The development of newer and
better-quality crystal phonograph cartridges along with their counterparts, the
ceramic cartridges, has made them the choice of many high-fidelity fans. No
question of the relative merits of the two types exists, but actually just a
question of choice. One of the features of the crystal or ceramic units is their
increased signal voltage over the low signal of the magnetic units. With a
higher output, the crystal -unit signal can be amplified with fewer stages of
amplification, and hence with lower cost in the construction of the amplifier.
While the output voltage of some magnetic units is as low as 2 millivolts (.002
volt), some of the crystal cartridges designed for inexpensive portable phonographs can produce up to 5 volts. The crystal and ceramic cartridges used in
high-fidelity record playbacks produce from .5 volt to 1.0 volt of signal energy.
Some manufacturers provide their crystal or ceramic cartridge with a small
network of resistors and capacitors connected across the terminals of the
unit so that it may be plugged into a magnetic cartridge input without overloading the tubes with too much signal voltage.
The
operation of
a
simple tone control
system indicated in block diagram form
HIGHS BOOSTED OR DROPPED OFF
HIGH -FREQUENCY NOTES
LOW -FREQUENCY NOTES
COMBINED SIGNAL
OUTPUT TO NEXT STAGE
LOWS BOOSTED OR DROPPED OFF
versatility of an amplifier may depend upon
its selective switching
and the number of input
The
and output jacks provided
Equipment for evaluation furnished by Sherwood Electronics, Inc.
Even though the first stage of amplification has been bypassed, the signal
from the crystal or ceramic cartridge must still go into the equalization stage,
as can be seen on the block diagram. The choice of playback characteristics
is made by the selector switch. It is necessary to follow the rest of the inputs
up to this point, where the record signal voltage comes out of the output of
the equalizer circuit.
FM-AM TUNER INPUT: The output signal voltage of the average radio
tuner used in a high-fidelity system is high, like that of a crystal cartridge.
However, the music or sound that is conveyed over the airwaves either is
live, and requires no record equalization, or if it comes from a record being
played at a studio, equalization is performed on the spot, before the signal
goes to the transmitter. This signal does not require the first stage of amplification the magnetic cartridge did, nor does it call for the use of the equalizer
circuit so, as seen on the block diagram, this radio audio signal is switched into
the preamplifier at the same point as the output of the record equalizer. This
point is the input of the bass and treble controls, which can be used to modify
all the input signals to suit individual tastes. These controls and their functions
will be discussed later.
The controls used to alter the character of an output signal in a
high-fidelity amplifier
Equipment for evaluation furnished by Bell Sound Systems, Inc.
41
-quality
preamplifier by Tannoy
This special high
of England features calibrated tone and loudness
filters
Equipment for evaluation furnished by Tannoy Prndu.'tc,
Ltd.
TAPE RECORDER INPUT: This signal is of the same nature as the radio
tuner signal in the amplitude of its voltage output. Most high-fidelity tape
recorders provide their own special preamplifiers, which boost the very weak
signals from the magnetic tape heads. A head preamplifier is located either
in the tape device or in a separate unit. Some of the more progressive amplifier manufacturers have incorporated a tape head preamplifier in the circuit
arrangement, just as the preamplifier provides for a magnetic cartridge. The
dotted line shows the position of this amplifier if it were to be employed.
Notice that it also bypasses the record equalization circuit.
MICROPHONE INPUT: If a crystal microphone is to be used, it may be
plugged into the crystal phonograph input if its output voltage is sufficient.
However, in the case of most crystal microphones and especially in the case
of dynamic microphones, additional amplification must be used. A crystal or
ceramic microphone can be directly connected to the input circuit, since these
are high-impedance devices. Magnetic microphones can be purchased with
either high or low impedance. In the case of a low-impedance unit, an input
transformer is needed to effect a good match to an amplifier. This transformer
may be located on the preamplifier chassis or, more usually, on a microphone cord attachment. The microphone circuit, however arranged, will bypass the
record equalization circuit and go directly into the tone control circuit.
TELEVISION OR AUXILIARY INPUTS: The television and/or auxiliary
inputs are connected into the circuit just before the tone control amplifiers
42
and, of course, bypass the preamplifiers and the record equalization circuits.
THE TONE CONTROL: The tone controls are provided so that a listener
can make adjustments in the final sound coming from his high-fidelity system.
The necessity for these changes is warranted by several facts. Individual
hearing preference is the most obvious reason, coupled with the acoustical
changes caused by the furnishings in different listening rooms. In the descriptive tone control block diagram, one block is used to show the treble circuit,
and another to show the bass -frequency circuit. Regardless of the type of
signal fed to the tone-control circuit, it may be desirable to change the degree
of amplification at any one frequency. If, for instance, the high-fidelity system
is to be used in a room that is highly absorptive at high frequencies, then these
high notes would tend to be lost. An amplification boost in the treble circuit
would solve the problem. The converse is true of a room where bass frequencies are lost or need to be attenuated. In an individual case, a person may
have a hearing loss at any of these frequencies, either high or low, and the
additional amplification of these frequencies would tend to restore normal
hearing.
The signal has thus far been carried through the preamp stage and the
phonograph equalizer selector network, and is now ready to be sent through the
portion marked Treble and Bass Controls on the block diagram.
The tone controls used in many cases are non-resonant in character, employing either capacitive or feedback effects to do the job. In an actual amplifier circuit, the controls may employ the two halves of a dual triode vacuum
tube. The increase or decrease of treble is almost always accomplished first,
as the signal is passed through the circuit with the bass unaffected. Then, in
the second part of the circuit, the bass is increased or decreased according to
the listener's desire; the newly changed treble response is unaffected. It is
Silicon rectifiers are often used in modern amplifier power supplies, replacing vacuum tubes
HIGH -VOLTAGE
FILTER CONDENSERS
WW -VOLTAGE
DRY
O1SK RECTIFIER
OW-VOLTAGE
ER
CONDENSER
43
L
wise to remember that the job of the tone controls is to decrease amplitude
as well as to increase amplitude in a particular range of frequencies. Control
is effected by the rotation of the treble and bass knobs. When these knobs
are left at the middle or "O" position, the tone -control circuit is inoperative,
and the signal goes through the amplifier unaffected. There are many methods
of gaining bass and treble control; an understanding of them would demand
an extensive study of electronics. All these types of controls accomplish the
same thing in that the user has an instantly variable set of controls to increase
or decrease the amount of bass and treble amplification in his system.
LOUDNESS CONTROLS (TONE COMPENSATED VOLUME CONTROLS): The modern preamplifier often has, in addition to its tone controls,
a control called the loudness control. This control is a combined volume and
tone control. The human hearing system has the faculty of losing sensitivity
to both bass and treble frequencies as the intensity level of the sound is reduced.
Earlier in the text it was mentioned that loudness, the response of the human hearing system to different sounds, varied with frequency and intensity.
Since it is not always possible for a high-fidelity system to be operated at a
volume level similar to that of a live performance, some sort of frequency
adjustment should be made under conditions of reduced volume. One of the
functions of the tone controls is to boost both bass and treble when the volume
level is reduced. However, unless an operator keeps some sort of log of his
settings it will be very difficult for him to reset the controls each time the
volume is changed. In the design of loudness controls, automatic tone compensation was kept in mind. By the construction of this special control, with
its own circuit of resistors and capacitors, volume of intensity of a sound is
changed, and as it is, so is the bass and treble relationship. As the volume is
decreased, the bass and treble range is increased and vice-versa, to conform
with the sensitivity of the human ear.
In amplifiers using loudness controls, a switch may be provided to cut out
the conventional tone-control circuits, but should there be none, the tone controls should be set at "O." In the diagram, the loudness control can be
switched into or out of the circuit.
English -made Leak amplifier.
Left: top view, showing line voltage
adjustment taps. Below: side view
The
sa
L,NE
TAP
LINE VOLTAGE PLUG
CLTAGE
POWER
TRANSFORMER
OUTPUT
TRANSFORME
FILTER
CONDENSER
RECTIFIER TUBE
AUDIO OUTPUT
TRANSFORMER
Equipment for
44
i
ln:rtirrrr furnish
Iry
British ln.lu.stries.
INC.
l
_
_
The final vacuum tube stage in a preamplifier is constructed according to
whether or not the preamplifier and the power amplifier are on the same
chassis. If they are, the signal is fed directly into the first tube of the power
amplifier, but in the case of separate units that may be some distance from
each other, a special vacuum -tube circuit called a cathode -follower circuit is
employed. The cathode -follower is a specially operated vacuum tube, which
is not used as an amplifier, but rather as a low -impedance signal output source.
A very long length of cable can be connected to this source without signal
loss at the end of the cable; in this case at the input of the power amplifier.
In this type of circuit, no line -matching transformers are necessary, as would
be the case if the signal were taken off the plate circuit of an amplifying tube.
The signal has been traced through the control or preamplifier to show
what needs this device must fill. Signal is ready now for power amplification.
POWER AMPLIFIERS: The primary function of a power amplifier is to
receive the program signal voltage from a preamplifier circuit and to send it
through various amplifying stages until it has sufficient power to drive a
loudspeaker system. The power amplifier, sometimes called the basic amplifier,
may be constructed on the same chassis as the preamplifier, or it may be on a
separate chassis. The operating voltage supply, commonly known as the power
supply, is usually on the same chassis as the power amplifier.
POWER SUPPLY: As we know it today, all electronic equipment employs
external voltage or power sources. Whether these supplies are in the form of
batteries, radiation -powered units, or generators, their energy is necessary for
the operation of the circuit. Most electronic devices depend upon direct current (d.c.), and this current, if not supplied from battery units, must be converted from alternating current (a.c.). Vacuum-tube rectification is most
commonly employed to obtain the higher d.c. voltages, and metallic disc rectifiers to obtain lower d.c. voltages. Since the performance of an amplifier depends directly upon the quality of its power supply, an explanation of the
operation of this unit follows. The block diagram shows the parts of the
power supply, as: power transformer, vacuum-tube rectifier, filter chokes, filter
condensers, metallic disc rectifier (for the preamplifier filament supply), and
high -wattage load resistor.
A power transformer is an alternating -current device made of two or
more wire coils with laminated steel cores. The purpose of the transformer is
to conduct 110-volt a.c. house current through its primary coil or winding, and,
through the process of induction, produce in other coils or windings (called
secondary windings), higher or lower voltages as needed. The transformer
simply steps up or steps down the a.c. voltage in its primary winding. To step
up 110 volts in the primary to 440 volts in the secondary winding, four times
as much wire is wound in the secondary coil. For the filament voltages, 11
volts may be needed; Mo as much wire is wound on another secondary coil.
Transformers are seldom made by experimenters because of the complexity of
their designs. Other considerations besides voltage are those of power -handling
capabilities and efficiency. While a power transformer operates within a very
narrow frequency range (usually 60 cycles), it must be designed to produce
sufficient power for all the vacuum tubes and other circuit elements in the
system. Heavy construction, with high-grade iron laminations, good insulation
between windings, and the proper gauge of copper wire, are the marks of
good transformer.
Several secondary a.c. voltages have been produced in the transformer;
they must now be converted to direct current where necessary. A vacuum tube rectifier is a device for the conversion of a.c. to d.c. It functions like
the vacuum tube explained earlier in the text, with the exception of the control
grid, which it lacks. The polarity of alternating current is constantly reversing,
according to its frequency. In this case, that frequency is 60 cycles. The need
for unchanging positive voltages on the plate elements of the various vacuum
45
Left: modern silicon rectifiers and mount. Right:
an old-style vacuum -tube
rectifier
Equipment for evaluation furnished by Audio 1)evic... In..
I
tubes in any electronic device can be met by employing a vacuum -tube rectifier.
Operating as a switch during the alternation of the current, the rectifier allows
current to flow naturally from cathode to plate whenever the plate is positive.
During the alternation of the voltage from the secondary winding of the
transformer, the plate is positive half the time and negative the other half.
During the negative cycle the electrons from the always-negative cathode are
repelled, and hence the tube current flow is cut off. As the current again be ,comes positive, it is allowed to flow in the circuit. In this fashion only that
half of the a.c. sine wave representing a pulsating direct current is used. By
putting another set of elements, plate and cathode, in the same glass bulb,
these elements can be made to work during the other half -cycle. Even then,
after the rectifier has done its work, the resulting pulsating direct current is
not smooth enough to be used in the circuits. If this pulsating voltage were
used on the plates of the various vacuum tubes in an amplifier, the resulting
sound from the loudspeaker would be characterized by the familiar 60-cycle
hùm heard in worn-out kitchen radios. When this happens we have all heard
someone say, "I guess a condenser went bad." The filter circuits in a power
supply are used to make absolutely pure direct current.
Filter choke coils and filter condensers are employed together to alternately
store and discharge their current supply into the rectifier circuit. As the
process of filtering goes on, the dips between the voltage peaks are filled in
and a pure direct current results. Alternating current can be used to power
the filament or heater elements in most of the vacuum tubes in amplifier circuits, but in the primary or early stages of the preamplifier it is wise to use
direct current. In these sensitive stages the alternating current used on the
heaters of filaments of the tube might cause 60-cycle hum interference. To
provide this low-voltage d.c., metallic disc -type rectifiers are used. These
operate much as vacuum tube rectifiers, but provide the extra current needed
in these low -voltage supplies.
46
POVIER FILTER CONDENSERS
FILTER CHOKE
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UNCOVERED POWER
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OUTPUT TRANSFORMER
F. 100/201 power amplifier with
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A modern amplifier and
its
schematic diagram
Diagram and equipment for ec::lint
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W6. NO. C99D-C21
H. Scott.
The power supply of an amplifier is quite important to the quality of reproduced sound in that it is often called upon to provide large amounts of
current over the amount used when the amplifier is operating at normal listening levels. This situation occurs during extremely loud passages or during
deep bass passages. If the power supply is not designed with heavy-duty parts,
this extra demand may be partially met with a marked drain on the other
tubes in a circuit. If supply voltages are lowered due to this extra drain,
noticeable distortion will occur. The power supply must be built to withstand these surges without draining power from the other vacuum tube circuits.
POWER AMPLIFIER OPERATION: The fundamental job of a power amplifier is to take the signal voltage that comes from preamplifier stages and
amplify it to a sufficient level to operate a loudspeaker. The perfect power
amplifier should not distort a signal one bit. The only difference in the signal
should be a difference in amplitude. Since the modern power amplifier has
many components and vacuum tube stages, there are many places where
distortion can enter the circuit.
49
RES
The block diagram shows the progress of a high-fidelity signal through
a preamplifier and its various controls, along with the power supply, which
powers the complete device. The block diagram continues with the final part
of the complete amplifier: the power amplifier.
Improvements in the design of the modern amplifier have increased its
frequency range. Extending the bass end of the range has made the disturbing
fact of greater intermodulation distortion more prevalent; extending the high
range has made this distortion easier to detect. Fortunately for the high-fidelity
enthusiast, this extension of frequency range has brought new circuit developments. Until about 1945, very little was done in the development of the final
audio stages of amplifiers. Designers of radios up to the late 1930's gave little
concern to anything but loudness as the function of the audio amplifier. Every
set from the kitchen radio to the console for the living room had the simplest
basic amplifier circuit. The antiquated systems of transformer coupling of
vacuum tube stages gave way to less expensive resistor and capacitor coupling.
Transformer interstage coupling in audio circuits was saved for professional
systems.
The invention of what is called push-pull output was the major breakthrough into high-fidelity sound. This output system will be discussed later.
The next major innovation in high fidelity was the design of the Williamson
amplifier by D. T. N. Williamson of England in 1946. The success of this fine
amplifier design lay in several areas; for one, it was an improvement electronically over other push-pull power amplifiers in that it employed an output
transformer designed for the circuit; also, it used "beam power" tetrode tubes
connected in the circuit much as triode tubes would be, but requiring less
driving signal voltage than triodes. In accomplishing the desirable circuit
action of the triode tube, the Williamson design employed negative feedback
voltage, producing increased power efficiently with less over-all distortion.
Another factor in the acceptance of this circuit was its relative simplicity of
construction and operation, the major expense being in the output transformer,
still the most important part of a power amplifier. Soon other circuits of great
merit, like the ultra -linear, were designed, requiring even greater design care
in the output transformer, with special windings and high-grade iron transformer laminations.
In the block diagram for the power amplifier are included these components:
the input stage, the phase inversion stage, the push-pull output stage, and
finally the output transformer. A brief explanation of these stages as the
signal progresses through the amplifier follows.
INPUT VACUUM TUBE STAGE: Since the job of a power amplifier is
to add power to a signal without otherwise changing it, the input stage-at the
other end of the cable or connecting circuit from the preamplifier-is a simple
amplifying circuit. The same action takes place in this stage as was explained
previously concerning vacuum tube amplifiers. A small signal is impressed
on the control grid of a tube, and is further amplified thereby. In some basic
amplifiers a level control is placed in the circuit at this point for over-all
maximum volume control of the amplifier. It would be unwise to connect and
use the full power of a 30 watt amplifier on a speaker system designed for 15
or 20 watts. Severe damage to a speaker can result during maximum power
overload, because a loudspeaker is designed for specific current and voltage
ratings. When placed in conditions of excessive voltage, like an electric light
bulb, it will burn out. Loudspeaker and amplifier manufacturers are now installing protective fusing devices in their components. The block diagram shows
simply an increase in signal amplitude with circuit connections to the next
stage, the phase inverter.
PHASE INVERTERS: Since the push-pull output tube arrangement with
two tubes is being used, it is necessary to get the signal to the control grids
of each tube in the proper sequence. The signal going to the output trans 50
Iv
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TUBE
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SECTION
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SIGNAL INPUT JACK
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TUBE
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GAIN CONTROL
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SUPPLY
A block diagram of a
push-pull power amplifier showing the relative amplitudes of
amplified in diffe-ent parts of the circuit
a
signal
as it is
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INPUT
SIGNAL
INVERTER
STAGE
PUSH-PULL
OUTPUT TUBES
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AND INVERTER
1st AMPLIFIER
TRANSFORMER
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OUTPUT TUBES
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TRANSFORMER
INPUT
1
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TO SPEAKER
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CATHODE -FOLLOWER SINGLE -TUBE PHASE INVERTER
DUAL -TRIODE
INVERSION STAGE
INPUT
SIGNAL )
PUSH-PULL
OUTPUT TUBES
TRANSFORMER PHASE INVERTER
DUAL -TRIODE
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OUTPUT TUBES
INVERSION ST!GE
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DUAL -TRIODE PHASE INVERTER
TO SPEAKER
SELF -BALANCING
DUAL -TRIODE PHASE INVERTER
Differen- types of phase -inverter stages
51
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former from one output tube must be 180° out of phase with the signal from
the second tube if a null condition, that is, cancellation of the signal, is to be
avoided. There are several methods for accomplishing this phase inversion.
One method is interstage transformer coupling. The plate circuit of the first
amplification stage is connected to the primary of an interstage transformer;
the secondary is electrically isolated but inductively coupled to the control
grids of the output tubes in the push-pull circuit. Much like the action of a
power transformer, the interstage transformer has a constantly changing but
always opposite phase relationship at the connecting ends of the winding. In
modern high-fidelity circuits the transformers required for this type of operation have to be of extremely high quality to compete with the resistance capacitance type of inverter. Where feedback is used, other problems are
inherent in transformer designs. However, there are several extremely high quality amplifiers using transformer coupling and inversion which seems to
dispute this concept.
Resistor and capacitor electronic inverters accomplish the same job, but
with lower cost and greater stability. In some of these electronic inverters
no amplification is gained, and in others the full gain of the tube circuit is
provided. A small block diagram of a simple resistive -capacitive inverter
52
shows how the out-of -phase signal is gained. This circuit is one of the early
phase inverters; it has been abandoned as unstable by all but the manufacturers of inexpensive amplifiers. Two blocks, representing the two inverter tubes,
have been set up. The top tube is a straightforward amplifier stage with the
signal applied to its grid. The signal goes through the vacuum tube and is
passed on to the following stage (in this case one of the push-pull amplifiers).
Now, as has been explained, the output signal in the plate circuit of any amplifying tube maintains a 180° phase reversal with the input signal. In normal
amplifying, this has no relationship to the quality of amplification. However, in
this case it fits our needs very well. It is plain that we need some sort of signal
on the control grid of the bottom tube if it is to power the second push-pull
amplifier. Yet, this signal must maintain this 180° phase reversal with the original signal. Here is such a signal at the grid of the first push-pull tube. Another
requirement is that the two inverter tubes supply their two signals at the same
amplitude. Both these needs may be gained if a variable resistor, or control, is
inserted in the grid circuit of the push-pull stage to tap off the required signal,
as shown in the block diagram. Once instituted in the circuit the two signals, 180°
out of phase, will be supplied to the output stage grids. There are problems in
this type of arrangement, since the balance of such a circuit is very difficult to
maintain. Any change in adjustment or resistor/capacitor value will throw
the whole circuit off, causing unbalance and eventual distortion of the sound.
The more common types of inverters use the cathode -follower type of circuit. An example of this circuit in block form is included here so that it may
be more easily understood. It operates according to the basic laws of electrical
currents flowing through resistive elements and their associated voltages
(signals) across identical resistors. It is a fact that if a certain amount of
current flows through a vacuum tube, this current is also present in each part
of the plate and cathode circuits of this tube. The correlation is easy to see.
If the normal plate load resistor is put in the circuit, split into identical
halves but at different points in the circuit, as shown in the block diagram, the
same current flows through each half. Now the voltage drop (signal) across
the plate resistor is identical to that across the extra half put in the cathode
circuit. The relationship of the phase of the input signal to the output signal
still maintains in this circuit, with the plate signal going straight into the
control grid of the push-pull output tube. The second control grid is supplied
by the 180° out-of -phase signal from the other half of the split resistor in the
cathode circuit. There are several variations of this circuit, but all accomplish
the same job: to supply the control grids of push-pull output tubes. The
INPUT SIGNAL
The operation of a single -
tube cathode -follower
phase inverter
CONDENSER (C)
OIDS ANY VOLTAG
DROP
ACROSS (RC)
+
"
DROPPING VOLTAGE
ACROSS (RP) PRODUCES
SIGNAL ON GRID OF
PUSH-PULL TUBE (A)
RP=RS; SAME AMOUNT OF
CURRENT FLOWS
THROUGH EACH
DROPPING VOLTAGE
ACROSS (RS) PRODUCES
SIGNAL ON GRID OF
PUSH-PULL TUBE (B)
53
purchaser of an amplifier has little chance to choose his type of phase inverter,
but at least this explanation will provide a measure of understanding of what
takes place in the amplifier.
PUSH-PULL OUTPUT STAGE: In the course of explaining the progress
of a signal through the phase inverter stage, the driving signal requirements
have been explained for these two vacuum tubes, which work together in
alternation to provide power amplification with a minimum of distortion.
The functioning of a push-pull circuit is as simple to understand as the
operation of a vacuum-tube rectifier. In the block diagram each tube is
represented by a block, these two blocks connected with the previous stage
by resistor and capacitor elements. The output of this final stage in the power
amplifier is connected directly to the output transformer.
For clarity, let's review the operation of a vacuum tube. Current flows
from the negative cathode element to the positively charged (d.c. voltage
from the power supply) plate element. In the path of this flow of electrons
is a metallic wire grid which can be charged, more or less negatively in
respect to the cathode, by the input signal we wish to amplify. When the
signal forces the control grid to become more negative, then fewer electrons
flow. There is a point or degree of negative charge where all electrons cease
to flow, and the vacuum tube is at cutoff. When the signal in its positive
swing allows the control grid to become less negative, then electrons will
flow, to a point where the grid swings out of the negative portion of its operating
mode, and saturation, or clipping, occurs. In this manner, small signal voltages
are amplified.
In a push-pull circuit the control grids are energized 180° out of phase.
The currents flowing through the two tubes are equal, reaching their maximum
and minimum peaks alternately, 180° out of phase. As one tube's current is
increasing from a zero value, the other's is decreasing from a maximum value.
These currents, flowing in the output transformer's tapped primary, produce
a signal in each half; the two are added together to induce the complete
signal in the secondary (voice coil) winding. There are many other factors
involved which do not bear discussion in a non -technical text, but one that
does is the output transformer, its design and function.
OUTPUT TRANSFORMERS: Earlier in this section we discussed conventional transformers used in power supplies. Power transformers are designed
to do the job of either voltage step-up or step-down, or both. Designers of
such transformers need consider only one operating frequency. In most cases,
AS PLATE CURRENT
FLOW DECREASES
IN TUBE (A),
AN INCREASE
OCCURS IN
TUBE (B)
INPUT
SIGNALS
FROM
PHASE
INVERTER
54
OUTPUT
TRANSFORMER
The operation
of a sim-
ple push-pull output circuit. The black and colored arrows represent
alternate tube operation,
a result of the 180 -degree phase difference between input signals
IDNCEN-RIC MULTICOIL WITH SHI
NDING
G STRAP
Some major components
of a Tannoy amplifier,
illustrating the practical
approach to high -quality
transformer design practised by British manufacturers
DUAL MULTIPLE
WINDINGS
LAMINATED IROn CORE
sALATIO-OUTPUT
TRANSFORMER
4,eOUhyb CbRE
'
PUSH-PULL OUTPUT
TO LOUDSPEAKER
INVERTER
VOLTAGE AMPLIFIER
NEGATIVE
FEEDBACK SIGNAL
R1
R2
(R1) AND (R2) FORM A
VOLTAGE -DIVIDER CIRCUIT GOVERNING
AMOUNT OF FEEDBACK
Rl
MULTI -STAGE FEEDBACK
OUT -OF -PHASE FEEDBACK SIGNAL
Two examples of applications of negative feedback in audio amplifiers
Equipment for evaluation furnished by Tannoy, Ltd.
power transformers are designed to operate on alternating frequency, 60 -cycle
power line supplies. In aircraft electronics, the frequency of operation may be
from 400 up to 4000 cycles, though usually a transformer will be designed for one
frequency range. A high-fidelity audio output transformer, on the other
hand, must function over a range of perhaps 10 to 40,000 cycles. Special
problems exist for designers of these devices. Mistakes in transformer design
almost always show up in some form of distortion, which is often blamed on
the preceding electronic circuits. Frequency distortion (limitation of frequency range) may be due to low winding inductance, reactance caused by
leakage, or resonance. Intermodulation and harmonic distortion may be reflected into the output stage, caused by overloading the primary winding during low -frequency passages. Lightweight construction of a transformer causes
low primary inductance in the bass frequency range; as this occurs, distortion
is caused by a reduction of load impedance in the output plate circuit. Another
feature of poorly designed output transformers, causing these same kinds of
distortion, exists when a situation of saturation of the core by the flux density
permits non-linear operation. These and other faults can be found in inexpensive output transformers though good design has tended to reduce many
of them to unimportance.
Modern transformer design involves the use of balanced coils rather than
the older single coils, and special methods of winding the coils on laminated
iron cores of better design and construction. Once a transformer is finished
it is placed in a metal container and sealed against moisture absorption. If
your output transformer can handle its rated power load, providing a wide
frequency response with good transient characteristics, your amplifier will
perform its high-fidelity function.
NEGATIVE FEEDBACK: The use of negative feedback is universal in the
design and manufacture of audio amplifiers. Its very universality has precluded it from becoming a sales argument at the time of purchase of highfidelity equipment. Since one of the functions of negative feedback in an
amplifier circuit is to partially correct for a poor output transformer, we have
left this explanation until now. With its corrective capabilities, negative feedback can aid in bettering frequency response, reducing harmonic, intermodulation, and phase distortion, correct for a poor output transformer, and in
general stabilize a whole amplifier circuit.
Negative feedback is a most complicated subject and this explanation will
be limited to a surface treatment only. The accompanying block diagram
55
TO LOUDSPEAKER
R2
FEEDBACK CONTAINED WITHIN THE OUTPUT STAGE
illustrates, in reference to the block diagram of the complete amplifier, how
negative feedback is applied in a circuit.
In operation, negative feedback occurs when a portion of the signal voltage
output of an amplifier is taken off the secondary of the output transformer
and fed back through a resistive -capacitive circuit into the control grid of
an earlier vacuum-tube stage. In audio amplifiers, the feedback loop usually
encompasses three stages. For purposes of simplicity let's assume that the
signal being amplified is a note from a 1000 -cycle tuning fork. As this note
is amplified by each vacuum tube stage, it gets larger in amplitude and elements of distortion may creep into its otherwise smooth sine wave. Observe
in the diagram the smooth input signal representing the 100 -cycle note, and
the 180° phase reversal between the sine wave on the control grid and the
sine wave on the plate circuit. Depending on the number of vacuum-tube stage reversals, the phase relationship from input to output may be in phase
or out of phase (180°) . Negative feedback gets its name from the fact that
if two sine wave signals are put together in phase, the result of their signals is
additive. If these two signals are put together 180° out of phase, their signals
are subtractive, so to speak. The former would be positive feedback and the
latter negative feedback. We are concerned with negative feedback. For the
input of our feedback signal we pick some stage that is 180° out of phase
with the ungrounded lead of the secondary winding of the power output
transformer. Through a resistive -capacitive circuit, a portion of the output
voltage or current (depending on the type of negative feedback desired) is
fed back to this control grid, or input element. The signal, being out of phase,
tends to modify the prime input signal in such a way that the resultant signal,
though lower in amplitude, is smoother and more free from distortion.
56
SPECIAL AMPLIFIERS:
PRLNTED CIRCUITS: Electronic devices are constructed of special parts
and interconnecting wires and cables. Conventional wiring and construction
are expensive because of the necessity for hand work and visual inspection. As
the wiring is not fixed in place, insulation must be provided; more space is
thus consumed. A maze of wires is not easily traced in production inspection,
and very little can be done with conventional wiring in automation (optimum
use of machinery).
Printed circuits get their descriptive name from their flat printed -page
appearance. It is a popular misconception that they are all actually printed
by a printer's press. Printed circuits (electronic, electrical) are manufactured
by various processes of printing (conductive paints), etching, embossing, pressure laminating, and inlaying metal conductive patterns on a sheet of
board -like insulating material. The purpose of any wire in an electronic
circuit is to carry current from one part of the circuit to another. Printed
circuitry does the same thing more compactly. In many cases the non-conductive board material can be used to mount other parts, making for much
smaller size with less complexity.
Since metallic conductors can be applied to both sides of an insulating
laminate, other circuit components, such as small capacitors and inductances,
can be formed in place right on the board. Switch contacts, volume control
surfaces, and mounting contact holes are readily formed in place. With transistors and other semi -conductive devices, electronic components can be made
much smaller and manufactured almost entirely by automation.
Printed circuit boards, which
serve as mounts for audio
components, make for easy
soldering and compact chassis
layout
The hand -sized, battery -powered
Fisher transistorized preamplifier
with its schematic diagram
Diagram and equipment for evaluation
Courtesy Fisher Radio
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Advantages in high-fidelity equipment: The major value of printed circuits to the buyer of high-fidelity equipment lies mainly in reduction of size.
The cost factor as yet is mainly in favor of the manufacturer. Little quality
difference exists between a wired and a printed circuit, certainly not a difference
that could be noticed by the human ear. We will see more and more use of
printed -circuit elements in electronic equipment as automation comes into
widespread use.
TRANSISTOR AMPLIFIERS: The advantages of transistors as used in
audio work are obvious. They are small, give long-term trouble -free operation, require no heater or high -voltage supplies, operate for long periods of
time from small batteries, and are current-operated devices rather than voltage operated devices like vacuum tubes. Though not always quiet, because of
design failures, the transistor is not microphonic, i.e., subject to hum fields
or thermal noise, as is the vacuum tube. Transistors are low-impedance devices, unlike the high-impedance vacuum tubes, and have many direct applications in the audio field.
We have shown how electrons in vacuum tubes are freed through heat
from a metallic cathode within a vacuum. Our problems have been concerned with the control of those electrons on their way to the positively
charged metallic plate. The transistor might be termed a solid-state electronic
device. There is no heater or cathode element as such, nor a plate within a
vacuum. To the eye, the transistor is a solid metal object not much larger
than an aspirin tablet.
The operation of transistors and other semi-conductors depends upon a
flow of electric charge carriers within the solid of which the transistor is
made. Rather than electron flow such as we have in the vacuum tube, we are
concerned with the generation and control of these electric charge carriers
within the solid.
The solid most used in transistors is germanium in a polycrystalline form,
prepared in a very involved manufacturing process. The operation of transistors and other semi -conductors depends upon certain impurities in the germanium crystals. Ways have consequently had to be found to control these
impurities, which provide the electric charge carriers and aid in the control of
the flow of these carriers. These imperfections are of several kinds, such as
radiation energy, imperfect atomic structure, and chemical impurities.
The early "point contact" transistor brought in a complete new set of
terms with its invention. Where grid, cathode, and plate were common terms
in vacuum tube work, there were now the emitter, base, and collector for the
transistor. The point contact transistor consists of two closely spaced contact
points on a small wafer of germanium much like the "cat's whisker" of an
old crystal set detector. One of the contacts is known as the emitter and the
other as the collector: the wafer of germanium with its holder is called the
base. The signal input is usually between the emitter and the base; the output is between the collector and the base.
In later transistors, called junction types, two sections of germanium crystal
of the same polarity of conductiveness are separated by a center section of
another material of opposite polarity. Differing from the point contact transistor, where the action takes place at the points, the action in the junction
transistor takes place in the union points between the various materials. The
transistors employing a grown junction, called N-P-N, have an emitter with
negative bias and a positive collector with respect to the base. With this
flexibility, numerous circuits can be devised for the transistor. Newer, more
complex transistors are suitable for many other vacuum tube replacements.
ADVANTAGES OF TRANSISTORS IN HIGH-FIDELITY EQUIPMENT:
The transistor has made possible lighter and more easily portable equipment.
There are no unique advantages in having transistorized circuits outside of the
compactness and simplicity of the external power supplies. The transistor has
59
PRINTED SILVER LEAD
BASE
EMITTER
ALUMINUM FILM
TRANSISTOR
VACUUM TUBE
I
H
CERAMIC
CERAMIC
GROUNDED GRID
GROUNDED BASE
N
10
1
I
p
O
O
p
p
O
O
O
O
O
IA)
jO¡Ióóó°
NOy) pOOP oH
óoi.b
GROUNDED CATHODE
GROUNDED EMITTER
Above: the ultimate in compactness. A printed transistor
developed by the U. S. Army.
Left: junction -type transistors
(A) without potential applied, (B) with forward po-
u)
w:
(7.
r
0
larity, (C) with backward
polarity applied. Far left:
(C)
II)
IA)
IC)
MITRE
--N.TYPE GROWN
transistor applications and
their vacuum -tube counterparts. Below: (A) point contact, (B) grown junction, (C)
alloyed junction transistors
CRYSTAL
MIX CHANGED TO P-TYPE
P.TYPE
P.TYPE
MIX AGAIN.CHANGED
EMITTER
.
°
P0p°o
%
GROUNDED PLATE
GROUNDED COLLECTOR
0-A-
:
EMITTER
^'
,'J/
TO N -TYPE
'
J
++
.:1YY'Y(
N -TYPE
-
COLLECTOR
COLLECTOR
EMITTER
BASE
.-
GERANIUM
BA
BASE
not given better or, at present, less expensive equipment. Preamplifiers in
high-fidelity systems were the first to employ battery-driven transistors, which
help to avoid hum signals and thermal noise. As yet the power transistor has
not made its way into high-fidelity equipment, but when it does there will
be a radical design change in all high-fidelity audio equipment; we will be
able to expect smaller combined units.
MAGNETIC AMPLIFIERS: Permanence and reliability of electronic equipment has always been a problem where parts which wear out, like vacuum
tubes, are employed. In recent years there has been much experimentation
with types of amplifiers that can be put into service and left without care.
Transistorized units have come as close to this ideal as anything in practical
operation at this time. Other new developments in old fields of endeavor are
producing promising results. In 1901 C. F. Burgess and B. Frankenfield
started the basic work that produced what is called the magnetic amplifier.
Their device was primarily used for industrial power distribution and control,
but their investigations have produced the modern magnetic amplifier, so
important to navigation, industrial controls and power devices, modern computers, rocket and aircraft guidance, and many other fields. The magnetic
amplifier is not well known to many high-fidelity amateurs, but one day it
will perhaps supplant the vacuum tube and the transistor amplifiers.
The magnetic amplifier, which uses only coils (reactors), unusual transformers, and disc rectifiers, is not subject to the characteristic changes or
failures of vacuum tubes. Shock or vibration does not affect it, and it is in
operation the moment it is turned on, without warmup. The magnetic amplifier has no moving parts, and so can be potted, that is, enclosed in a metal
container and surrounded with resin, which on hardening protects the device
from moisture and corrosion. Complete isolation between input circuits and
output circuits can be accomplished. The versatility of control of the magnetic
60
Equipment for evaluation furnished by Regency
A single dip -soldering operation makes connections on the printed circuit wiring board of this
compact preamplifier. Preamplifier has self-contained, battery -operated power supply
amplifier makes it the perfect amplifier within its operating frequencies, with
capabilities of many times the power gain of a single vacuum-tube stage. The
input and output circuits can be prepared to cover a wide range of impedances
without extra cost problems. Successful work has been done to provide designs for audio frequency amplifiers operating up to 20,000 cycles. It is
probable that combinations of magnetic and transistor amplifiers will appear
before the magnetic amplifying device by itself.
CAPACITOR (DIELECTRIC) AMPLIFIERS: Rather new developments in
an old idea have brought the dielectric amplifier into the range of possible use
for amplification within high-fidelity audio ranges. Its qualifications are
much the same as those of the magnetic amplifier, with the exception that it
is capacitance -operated.
One of the basic electrical phenomena connected with certain types of
electrolytic capacitors is that their capacity varies with the d.c. voltage applied to them in a circuit. This variation can be used for wide-range amplification. The use of capacitor (dielectric) amplifiers is limited to special
instrumentation and computor circuits at this time, but further work is being
conducted toward their use in audio work.
STEREOPHONIC (MULTIPLE CHANNEL) AMPLIFIERS: Stereophonic
playback, employing two loudspeakers, has brought the need for two power
A simplified magnetic amplifier employing a magnistor. A large flow of current can be controlled by a
relatively weak signal
ES=CARRIER
CONTROL COIL
SIGNAL COIL
EC
IC
INPUT
CONTROL
IS
_1
A dual -channel stereophonic amplifier with operational controls. Left:
top view. Below: front
view
Cuurlcsy Madison -Fielding Corp.
STEREQPt-IONiC
amplifiers in high-fidelity systems. Regardless of the number of channels
used in the playback of stereophonic sound, each must use its own amplifier
and speaker combination.
The power amplifiers used for each channel are no different than any
other power amplifiers used in conventional high-fidelity systems. Whether
the stereophonic sound is recorded on magnetic tape or on a single -groove
disc, each channel will need power amplification. Since the principle of
"stereo" lies in the fact that completely separate channels are used for recording and reproduction, the power amplifiers should be completely individual
even though they are sometimes constructed on the same chassis by the manufacturer. It is not necessary that the power units be together; they may be
separate conventional power amplifiers even of different makes, working in
the same or in different locations.
AMPLIFIER INSTALLATION: The prime requirement in the installation
of any electronic equipment is adequate ventilation, with considerations of
decor, access, and ease of mounting following.
There are two reasons for stressing the importance of ventilation. The
first is the safety factor, and the second the proper functioning of the equipment. All vacuum tubes operate at elevated temperatures. Some, as in portable radios, emit only a small amount of heat and others, like the latest type
of power amplifier output tubes, operate at a temperature high enough to
62
a
ignite paper. In planning the installation of a power amplifier and power
supply, you must give consideration to venting the confining area so that the
heat can escape.
1. Try to mount your amplifier in a horizontal position with the tubes vertical so that the heat can be carried away by natural convection. If possible,
raise the amplifier at least an inch from the mounting board. (Some amplifiers have perforations in both the chassis top and the bottom cover plate
to aid natural ventilation around the component parts.)
2. If the amplifier must be mounted so that the vacuum tubes are horizontal,
the amplifier should be turned so that the tubes are not shielded at the top
by transformers or other component parts.
3. Try to place the equipment in an area with at least 4 inches of clearance
on each side and 6 to 8 inches of clearance at the top. If possible, drill many
1 -inch holes in the back or floor of the mounting area to facilitate ventilation. Wire mesh decorative panels in concealing doors will also aid in cooling
the area.
4. When it is necessary to install two power amplifiers and power supplies
in one area for stereophonic sound, extra care must be taken to provide more
Above: a dual -channel
stereophonic power amplifier. Right: a stereophonic preamplifier for
use with the above power amplifier. Ganged controls are featured
Left: front view of dual channel stereophonic amplifier. Below: schematic
diagram of same
Equipment for evaluation furnished by Bell Sound Sytiteo
64
_AO
Z':
Two separate high -quality amplifiers
are here used together for dual -channel
stereophonic sound reproduction This
is a recommended practise
Equipment for evaluation furnished by H Ií Scott, Inc.
than minimum air passage. Where the area is small, forced -air ventilation
by electric fan is advisable.
5. Avoid close contact between vacuum tubes and flammable materials such
as wood surfaces and grill cloth and paper.
6. Avoid storing plastic disc records close to the heat generated by power
amplifier equipment.
7. Avoid storing magnetic tape in areas that may be heated above 75°, by
either the household heating system or the heat from the amplifier equipment.
8. Avoid storing magnetic tape in areas near the large power transformers
used in both power supplies and power amplifiers.
9. The placement of a separate control or preamplifier is less limited due to
the fact that less heat is generated by these units unless they are self powered. Even those that are self -powered do not generate as much heat
as the power units.
If all the basic safety requirements for amplifier installation are met, the
final requirements of access for maintenance and repair can be considered
along with those of decor. Decor is a matter of aesthetic choice, and should
be appropriate to the mechanical design. Access is simply a matter of mounting the unit so that it can be easily unbolted or unscrewed and removed.
Exterior decor should complement the existing fittings of the room. Interior decor of the installation is a matter of good judgment; no flammable
painted surfaces, water-mixed casein or rubber base paint. A better surface
is 1/4 inch of asbestos board or similar fire-proof material. Heeding this advice
will provide a safe, long-lasting system.
THE DO-IT-YOURSELF AMPLIFIER KIT: Apart from the fun derived
from assembling a modern kit amplifier, you can have a better system per
dollar spent. We do not imply that kits are better than factory-made units.
There are "lemons in both cases. However, if you are interested enough
65
Good soldering practise is one of the
most important requirements for suc-
i.
cessful kit building
e1
4+f[;
Equipment furnished by Heath Co.
in doing the mechanical work, you need not consider the fact that you may
never have had a pair of pliers or a soldering iron in your hands before.
There are many kits available, some good, and some bad. The only advice
we have is to buy your kit from a reputable manufacturer. The companies
that sell kits at marginal prices must use cheaper and less reliable parts. You
will find "off-brand" parts-resistors, condensers, capacitors, and transformers
-of unstable and unreliable quality in these units.
You will have several important considerations before buying and starting
any amplifier kit:
1. Are you willing to follow the directions? The manufacturers of these kits
have gone to a great deal of trouble to provide just exactly the right parts,
Below: Check off kit parts on a check
list before assembly. Right: good tools
and a neat place to work are necessary.
Below, right: finished kits can have
professional appearance
Equipment furnished by Heath Co.
may be constructed from kits.
Above: kit -constructed audio analyzer, left, and
vacuum-tube volt-ohmmeter, right. Right: Coauthor Jordan at work
Test equipment
Ltuipinent furnished
by Beath Co.
the correct lengths of wire, and good design and layout. Even if you have
had considerable experience in electronics, you will find that following
directions will not only save you time but eliminate electronic troubles in
the final unit.
2. You will have to be able to solder correctly, since poor soldering is an
eventual source of trouble. Lead solder has only one function: that of
making a good electrical connection. It is not intended as a support for parts.
The only solder to be used is rosin core solder. ACID CORE SOLDER MUST
NEVER BE USED. ALMOST ALL SOLDERING PASTES ARE BAD FOR
USE IN DELICATE ELECTRONIC EQUIPMENT. The corrosive effects
of acid on metal are well known. Acid core solder, when heated with a soldering iron, splatters small globules of acid on all surrounding parts. In the
course of time corrosion takes place, with serious harm to the functioning
of the circuit.
3. You will have to observe safety rules in connection with the high voltages
used to operate amplifier circuits.
4. Again: Follow the directions provided by the manufacturer for placement
of parts, layout of wiring, and correct soldering of the final connections.
Instructions for balancing the modern high-fidelity amplifier are given by
the manufacturer and should be done exactly as stated. Sometimes a voltmeter is required; it can be borrowed or built from another kit.
MAINTENANCE OF AMPLIFIERS: On the whole, serious maintenance
problems in high-fidelity audio amplifiers are the responsibility of trained technicians. However, there are steps that can be taken to prevent minor problems
from occurring.
1. Follow all requirements laid down for the ventilation of vacuum -tube
equipment to avoid overheating of tubes and component parts, which would
tend to shorten the operating life of the amplifier.
2. Keep all plugs and connectors free from loose connections.
3. Do not run plastic-covered wire or cables near or against hot vacuum
tubes.
4. Check vacuum tubes at least every 6 months to a year, replacing any
tubes showing short-circuited elements or low emission.
5. Be sure that all speaker terminal screws or connections are maintained
tightly against the wires.
67
In making long runs from other units to amplifiers be sure to use the
correct cable.
7. In making long runs between amplifiers and loudspeakers be sure to
use wire of large enough diameter, and try to avoid running it long parallel
distances against grounded waterpipes or metal air ducts.
8. Do not use conventional light plugs and sockets for connectors on loudspeaker lines, since there is always a chance that someone may plug or connect these elements to a 110-volt house receptacle.
CORRECTIVE MAINTENANCE: Should an amplifier fail to operate, there
are certain things the owner can do to locate the trouble and, if it is not serious,
correct it himself. The steps which follow are those to use when trouble occurs.
If the trouble is audible, such as hum or "static," it may be caused simply by
a bad tube or loose connection. However, it may be internal trouble for which
a trained serviceman will be necessary.
One of the cardinal rules of set maintenance is not to try to force a failing
or faulty piece of electronic equipment into prolonged operation after trouble
has set in. It is common for a simple trouble like a shorted tube or part to cause
the burn -out of major transformers, producing an expensive repair situation.
If in doubt call in an experienced repairman or technician.
Many of the failures of vacuum tubes can be averted by periodic checks
and replacement of tubes that have become weak or shorted. It is wise to
keep track of the time in service of all your vacuum tubes and the type and
number of repairs necessary over the life of the equipment.
6.
Proper testing, as on
the
Fisher
100
amplifier shown, calls for high -quality test equipment
Amplifier furnished for e,.d ruj
68
Fisher Radio Corp.
1.
FAULT
Hum
(goes away as
volume control
CORRECTION
CAUSE
Loose cable or lead Check leads and conto input of amplifier nectors for poor connection
is turned off)
2.
Hum
(same)
visible tube
filaments or
dial lights
3. No
4.
Static
(intermittent
noise)
5. Microphonic noise
(ringing sound)
6. No sound
Fuzzy sound
(increased distortion)
on all selector
positions
8. Fuzzy sound on
phonograph position
only
9. Low volume (on all
selector positions)
10. Smoke, no sound
7.
Bad tube or loose lead
in preamp or early
stages of amplifier
Faulty fuse or short
somewhere in circuit
or tubes
Check tubes and leads
Check fuse, replace
once with same size
fuse. If still bad, check
tubes and replace fuse
Loose connection in Check plugs, connectors, and cables for
connecting cable
loose connections
Test by tapping and
Microphonic tube
replace
Blown fuse
Check fuse. Check for
Burned out tube
filament light, substitute
Check cables and
Disconnected cable
Open speaker voice speaker by substitution
coil
Bad tube or circuit Check tubes or call
technician
troubles
Faulty phono car- Check stylus, cartridge
tridge or stylus. Bad or tube
preamp tube
Bad amplifier tube or Check tubes
bad rectifier tube
Shorted part or tube Call technician
69
4
LOUDSPEAKERS
HISTORY: Man's efforts to be heard across long distances were described
earlier in this text. The development of sound amplification started with the
speaking trumpet, which was in use as far back as the pre-Christian era of
ancient Greece.
The primary work in the modern concept of the loudspeaker, a device
for changing electrical signal energy to acoustical signal energy, might be
credited to Alexander Graham Bell and Thomas A. Watson, inventors of the
telephone. However, the accidental invention of a voice reproducer came
before the arrival of the disc phonograph in 1876. Bell and Watson found
that two sets of wire coils and spring steel reeds could be hooked together by
wires in a circuit with a battery; when one spring was set in motion mechanically, the other spring was made to vibrate by impluses sent along the wire from
one coil to the other. It was in this fashion that the first conversion of electrical
signal energy to acoustical energy took place. As distinguished from the
signal of the already useful telegraph, the Bell and Watson discovery showed
that all the subtle variations of speech might be converted into electrical signal
energy at one end of a pair of wires, transmitted any distance, and then reconverted into acoustical energy that could be heard by the human ear.
The whole history of the telephone is closely linked with the necessity of
accomplishing this electricity -sound conversion in the most practical and
efficient manner. The earpiece (receiver) of the telephone was really the
first loudspeaker, however softly it transmitted sound. The first loudspeaker
might best be described by analogy to Bell's idea of a coil of wire and a piece
of spring steel. In the case of the loudspeaker, the coil is wound around a
magnetic iron core; the disc -shaped piece of spring steel (diaphragm) is placed
over the pole pieces of the coil. As electrical signals are fed into the coil,
the over-all magnetism of the iron core is changed in direct relationship to the
signals; the suspended metal diaphragm is attracted to and repelled by the
core accordingly, causing the air to pulsate and produce audible sound. What
has been described is the principle by which most loudspeakers operate. There
are some electrical and mechanical differences between units, but all accomplish fundamentally the same job. Design and production methods vary because of differences in approach to the problems of converting electrical signals
to acoustical energy as efficiently and with as much fidelity as possible.
The radio loudspeaker was developed shortly before World War I by Peter
70
STEEL REED
Alexander Graham Bell
and Thomas Watson used
this device in discovering
that an alternating signal could be transmitted
over wire
VIBRATED
BY FINGER
50 FT. OF
IDENTICAL STEEL
REED VIBRATED
BY ELECTRICAL SIGNAL
STORAGE BATTERY
Jensen and Edwin Pridham, who discovered and patented the moving coil
dynamic loudspeaker in January of 1913.
The personal history of Peter Jensen reads like a novel. It begins in Denmark, where Mr. Jensen was born in 1886. His first important job was with
Valdemar Poulsen, the Danish Edison, at Poulsen's laboratory. It was Jensen's
job, as an assistant engineer, to operate Poulsen's experimental radio station
at Lyngby, Denmark. Jensen initiated and carried out the first successful
experiment in transmitting music and voice by wireless (radio telephony) in
1907. In many broadcasts during the next two years he had what might be
called the first "disc jockey" show, since his programs were made up of talk,
news, and phonograph record music. Included in the text is a photograph of
a letter, received by Peter Jensen in 1909, thanking him for the "wireless
music."
In America by 1910, Jensen went to work in San Francisco and soon got
together with Edwin Pridham and Richard O'Connor to form the Commercial
Wireless and Development Company, which later became the Magnavox
Company. Mr. Pridham remained an executive of The Magnavox Company
until he retired in 1954.
Jensen and Pridham worked together until 1925, acquiring some thirty
U.S. patents during their association. Among these patented devices were
unique and successful loudspeakers, telephone transmitters and receivers, and
radio microphones. They concealed the first public address system at the
Panama -Pacific Exposition high on the Tower of Jewels in 1915 and tried
out their yet-unpatented invention for a few days. It was reported in the local
papers that sailors on the old battleship Oregon, at anchor out in San Francisco
Below: the operation of an iron diaphragm sound reproducer as used
in telephone communications. Right: a moving -armature loudspeaker of
the type used in the 1920's
HORSESHOE MAGNET
HORN
DIAPHRAGM
SIGNAL APPLIED
TO EARPHONE
TERMINAL
THIN IRON
DIAPHRAGM
MOVING
ARMATURE
CONNECTING LEADS
71
Bay, were dancing to mysterious music in the air. Later that year the P. A.
system was used at a public Christmas Eve gathering in front of the San
Francisco city hall.
Jensen and Pridham perfected and produced speech equipment used in
World War I destroyers, and devised and patented the first lip microphone for
aircraft use at the same time. In 1916 they applied for a patent, granted in
1920, on the first electrical sound-magnifying phonograph. They used a microphone and speaker arrangement with a volume control located on the front of
the phonograph. In 1919, they provided the public address system for President
Wilson's speech at the League of Nations in San Diego. A story, filed by
Philip Kinsley of the Chicago Tribune on September 20, 1919, tells all about
this remarkable feat.
In 1915 Jensen and Pridham provided the world with its first stereophonic
demonstration. In a two-story roadhouse and restaurant called the Hoo Hoo
House they installed, at the request of the management, an astonishing and
intricate system. A live orchestra played upstairs, with microphones attached
to each of the five instruments; the microphones were connected to individual
amplifiers and run to speakers on the lower floor which were set up in the
same positions as the live players upstairs. The stereophonic effect was so
astounding that people forgot to dance. Unfortunately, nothing further was
done with this unique and remarkable system.
Jensen and Pridham helped to change the course of cultural as well as
scientific activity with their development of the public address system and
the forerunner of the modern microphone. The first prize fight at which
a Magnavox announcing system was used was the Dempsey-Carpentier
fight in Jersey City in 1921. There ended the era of the "leather -lunged" fight
announcer. Mr. Jensen founded and directed the Jensen Radio Manufacturing
Company until 1940, when he resigned to form Jensen Industries, makers of
phonograph needles under the trade name Jensen. His has been a remarkable
career in a fabulous business.
Of all the types of loudspeakers invented and tested in the past, the moving
coil dynamic system has provided the most efficient and highest quality unit.
Right: Edwin Pridham and Peter Jensen in their
laboratories with one of the first dynamic horn type loudspeakers, circa 1915. Below: The first
dynamic loudspeaker. Designed and produced
by Jensen and Pridham in 1911
tesy of Peter Jensen, Jensen Industries, Forest Park, III.
72
Right: Wilson was the first president to use electronic voice amplification. This speech occurred
on September 20, 1919. The microphones are at
upper left. The horns were used to direct the
sound to carbon microphone units
HORN
DIAPHRAGM
MOVABLE COIL
Courtesy of Peter Jensen,
Jensen Industries, Forest ['ark, III.
TERMINALS FOR 6 -VOLT
STORAGE BATTERY
STEPDOWN COIL
Left: internal cutaway of an early
dynamic horn driver loudspeaker
designed by Peter Jensen and
Edwin Pridham
TERMINALS FOR
HIGH -VOLTAGE AMPLIFIER
Courtesy of Peter Jensen, Jensen Industries, Forest Park. III.
Right: original Mag-
navox dynamic loudspeakers of 1919. The
amplifier illustrated
used
the
De
Forest
Audion three-element
tube
One of the first permanent-magnet dynamic loudspeakers ever put into production. This was manufactured by the Jensen
Manufacturing Company
Courtesy Jensen Mfg. Co.
Other types, such as the flat metallic diaphragm, the moving armature magnetic,
and condenser loudspeakers, have fallen by the wayside and are no longer
used. The transition between the older types of moving-armature cone loudspeakers and the moving-coil cone loudspeakers took place about 1930, with
the advent of the electrodynamic unit. Though many refinements have been
made in recent years, there has been little change in the external appearance
of the loudspeaker. The permanent magnet has taken the place of the large
and heavy field coil that produced the necessary magnetic fields: the former
have gotten smaller as new, stronger magnets have been developed. Speaker
cones have changed somewhat in appearance, with special configurations and
new types of pulp materials. The combining of several speakers of different
ranges into one multi -axial unit has added bulk to some speakers.
THE LOUDSPEAKER: A loudspeaker is a device designed to move air in
response to electrical signals supplied to it by an audio amplifier. Working
back and forth very much like a piston in an air compressor, the speaker
moves the air around it, causing compressions and rarefactions, which result
in the sensation of sound in the human hearing system.
Early Atwater -Kent speaker with thin wood
veneer cone. Left: driver structure. Below: soft
chamois compliance ring
Laboratories of Robert Oakes Jordan, Highland Park, III.
MAGNET STRUCTURE.
COR LEADS
THIN WOODEN
SPEAKER CONE
The modern loudspeaker, a magnetic motor device, consists essentially of
a metal frame, a paper pulp cone, a voice coil ring form and wire coil, a cone
suspension system, and a magnet frame with magnet and cylindrical pole piece.
The basket-like frame and the magnetic system are welded together concentrically to provide a rigid holding and centering device for the moving cone
and coil assembly.
In the manufacturing process the cone and coil assembly is slipped into
place concentrically between the annular (ring -shaped) magnetic top plate
the centered pole piece. The voice coil and its form move freely in this position,
without touching either the pole piece inside or the magnetic top plate outside.
After the cone has been accurately centered in this position, the suspension
system, often called the "spider," and the cone rim are cemented in place
permanently. The leads from the coil system are brought out to soldering
terminals through very flexible wires.
There are many different types of designs in loudspeaker cone, frame,
and magnet structures. In any working system such as a loudspeaker there
are many mechanical and electrical drawbacks which can be circumvented only
O
Copper
Impedance
Matching
Cone and
Voice Coil Assembly
toi, .et1...Et.E
Basket Assembly
...e.
Above: the components of a modern high quality loudspeaker. Right: the assembled
loudspeaker
Equipment for evaluation furnished by Norelco
Pole Piece
and End Plate
Ring
Magnet Support
Magnet
Assembly
Structure
SOFT SUSPENSION RIM
110
///
MOVING DIAPHRAGM
/
//////////
VOICE COIL FORM
uu
ey
Equipment for cvab ati.r rn,.ted by
James B. Lansing Sound, Inc.
Above: the James B. Lansing
D-130 loudspeaker showing the
4" voice coil and the aluminum
dome which aids in the dissemination of higher frequencies
7
POWERFUL
PERMANENT
POLE
VOICE COIL OPENING
DUST COVER
MAGNET
PIECE
I
FLEXIBLE INNER
CONE -SUSPENSION
DIAPHRAGM
í
VO
CE
COIL W NDING
//////
Equipment for eva_uatfun rurnfahed by
Jeun Mfg.
Co.
METAL MAGNET FRAME
METAL BASKET FRAME
Above: the construction of an ordinary permanent -magnet loudspeaker.
Left: a dual cone on a single base is used to provide wide -range sound
reproduction
Equipment for evaluation furnished by British Industries, Inc.
SOFT SUSPENSION RIM
MADE FROM ISOCYANATE
FOAM PLASTIC
Extremely soft rim suspension materials
are sometimes used
to raise the compliance of a loudspeaker
cone
by special considerations in design. Each
what makes a good loudspeaker, and this
a basically simple device. These variations
the result of the jobs they are designed to
manufacturer has his own idea of
accounts for the wide variation in
in types of speaker units are partly
do.
THE FULL RANGE ALL-PURPOSE LOUDSPEAKER: To understand the
difficulty a single loudspeaker has in reproducing a full frequency range of
sound, it is necessary to see what happens to the speaker and its cone as the
range is widened. Since the function of a loudspeaker is to move air, the cone
should be active at all frequencies in the audible range. If a theoretically
perfect speaker cone were possible, it would be weightless, not subject to
inertia (the resistance of an object in motion or at rest to any change of mode),
and non-resonant; i.e., it would not move more easily or to a farther point
at any one particular frequency.
If just one note at a time were played through a speaker, many so-called allpurpose loudspeakers would function very nicely. However, loudspeakers
are expected to reproduce faithfully each of the many different sounds coming
simultaneously from an orchestra. If the action of a loudspeaker cone were to
be photographed with a slow-motion camera, it would be apparent that the
rim of the cone was unable to keep pace with the rapid movement of the
portion of the cone nearest the voice coil. This is due to the inertia of the
cone. No cone material is entirely without inertia.
Large speaker cones reproduce low-frequency sounds best; small cones
reproduce high-frequency sounds most efficiently. Designers of single-unit
speakers usually try to compensate for the inertia -caused disadvantages of
single speakers by utilizing the central portion of a cone for high frequencies
and the peripheral portion for low frequencies. Thus, many single -unit speakers
have hardened central areas or aluminum domes over their voice coils which
radiate sound in the high-frequency range.
If a single -unit speaker is to be used, it should ideally be large, stiff, and
light in weight, with an extremely soft suspension system. The resonant frequency of the cone, e.g., the frequency at which the cone tends to vibrate
sympathetically, should be below the operational limit of the amplifier with
which it is used.
B.
Lansing
loudspeaker. The
shallow frame and thin
styling make this speaker suitable for use in
cramped installations
The
James
D-123
Equipment for evaluation furnished by James B. Lansing. Sound. Inc.
77
Left: an 18
Electro -Voice
Equipment for evaluation furnished by Erectro-Voice, Inc.
P15 -LL low -frequency driver
"woofer." Right: Jensen
Other cone design features can include annular embossed rim corrugations,
which lower the mass of a diaphragm at high frequencies through greater
compliance, and variations in the actual shape and depth of the cone itself.
Two popular types of cones for full -range work include the plate type (shallow
diaphragm with large -diameter, dome-covered voice coil) and the exponential
(curvilinear) type with a large-diameter, dome -covered voice coil. Even with
all the design features that can help a single speaker to operate over a wide
range, there is still the possibility of intermodulation distortion caused by the
two modes of diaphragm operation at either end of the frequency spectrum
However, in better speakers intermodulation distortion is at a very low point
and not too noticeable.
LOW -RANGE LOUDSPEAKERS (WOOFERS): The limitations of full range, all-purpose single speakers have given rise to a popular loudspeaker
system in which the audio frequency range is divided among two or three,
or sometimes four, individual loudspeakers. Each of these individual loudspeakers is designed to operate over a narrow range of frequencies. With correct design, the narrow ranges of these units overlap in such a way that the
complete audible range is smoothly covered. Some manufacturers advocate a
separate speaker for each range to be reproduced; others advocate coaxial
systems, where two or three speakers are constructed on the same frame.
There are merits and drawbacks to each system. In considering a system
using individual loudspeakers the low-frequency unit, often called the woofer,
is the starting point.
The frequency range in which any loudspeaker operates best is determined
by the peculiarities of its design and manufacture. In the woofer's driver
unit the important thing is the lower range of frequencies in the audible scale.
Here are the major points to look for in a woofer: good low frequency performance, insured by a moderately stiff, large -diameter cone backed by a
large permanent magnet of high efficiency; a highly compliant suspension
system allowing relatively unrestricted movement at high driving power; a
deep cone diaphragm, desirable if the woofer is to be used in a direct-radiator
enclosure (reflex or infinite baffle type); heavy frame and magnet construction
to prevent distortion due to warping of the structure in mounting; low diaphragm resonant frequency (below the lowest response point of amplifier
operation); and the proper crossover frequency in relation to other loudspeakers in the same system.
The crossover frequency between the loudspeakers in any system is the
78
point on the frequency response curve at which the operation of one speaker
drops off and another begins to operate. The crossover point is not a point
at which a speaker which has been going full blast suddenly stops operating.
It is the point at which the action of the speaker begins to taper off. As an
example, the woofer in a system may have a crossover point of 600 cycles per
second. The woofer will reproduce all signals between about 35 and 600 cps
in a realistic, linear fashion; above 600 cps the sound from the woofer will fade
slowly. The speaker handling the next range will have realistic, linear reproduction characteristics from 600 cps on up. However, it will begin operating
at about 400 cps, gliding into operation gradually. The response of the two
speakers is controlled by a filtering system known as a frequency -dividing
network. The same type of tapering action takes place between all successively
operating speakers in the system, insuring smooth coverage of the entire frequency range.
In a three- or four -speaker system the woofer is designed to cover a frequency range of 35 to 600 cycles (or lower). In a two-speaker system, however, the low-range speaker may have a crossover frequency near 1200 cycles,
starting from 35 to 50 cycles at the low end. How the speaker operates depends
upon the manufacturer's concept and design. How well the speaker operates
within its designed limits depends upon the quality of construction and the
type of enclosure employed.
MIDDLE -RANGE LOUDSPEAKERS (MID-RANGE DRIVERS): An old
theory suggests that a separate loudspeaker for each of the many thousands of
note combinations played by an orchestra would make possible a perfect reproducing system for sound. It should be added that each of these speakers
would have to be especially designed to operate at its own frequency. This
type of high-fidelity system might be the best obtainable, but of course it
would be large and expensive. As a practical compromise, manufacturers of
speaker systems have provided high-fidelity enthusiasts with multiple speaker
arrangements.
Cone -type speakers are often used as mid -range units. Such speakers are
governed by the same design factors that apply to any other narrow -range
units. Tweeter units, however, have attained many partisans because of their
great efficiency and high fidelity in the middle and high -frequency ranges.
The average tweeter unit employs a relatively large diameter voice coil
for the small area of its diaphragm. Of the general types of tweeter systems
available, the annular (ring -shaped) style, the V-shaped limited -area style, and
the dome-diaphragm style are most used. For situations where a tweeter must
cover a fairly low range of frequencies, as in the case of a mid -range driver,
0
ó
I
(
EFFECT OF
curves
R
I,
HIGH -FREQUENCY BOOST
NATURAL HI
TURAL LOW -END
ESPONSE OF
indicate the
point of crossover between the operation of
two speakers. The effects of tone -control operation are shown
The
I
I
I
I
I
EFFECT OF BASS BOOST
RESPONSE OF
AND SPEAKER
AND
SP
CROSSOVER
EFFECT OF
EFFECT OF
HIGH -FREQUENCY
BASS DROP
DROP
I
Ó
i
LOW -FREQUENCY
IGH-FREQUENCY
SPEAKER BEGINS
OPERATING
SPEAKER CEASES
TO OPERATE
I
(BELOW HEARING LEVEL)
LOW RANGE
MIDDLE RANGE
HIGH RANGE
RELATIVE FREQUENCY RESPONSE
79
Speaker furnished for evaluation
by Jensen Mfg. Co.
High -frequency unit furnished for evaluation by James B. Lansing Sound, Inc.
Left: unusual cone arrangements are sometimes used to increase the frequency range of a loudspeaker. Right: special dispersion elements are sometimes added to high -frequency horns
the V-shaped unit is preferable. Since the diaphragm is clamped in the structure by both its inner and outer edges, and the voice coil structure is attached
to the apex of the V, it performs without the characteristic break-up which
often occurs in a dome or cone -diaphragm speaker. The horn structure used
to transform and couple the sound to air in a room is designed in such a way
that the most efficient and widest possible radiation of the higher frequencies
takes place. This feature eliminates the problem caused by the fact that the
higher audible sounds go, the more directional they become. Dispersion over
a wide angle becomes necessary if wide -range sound is to be heard everywhere
in the listening room. With the horn-type driver the problem of a suitable
enclosure is solved by the tweeter structure and its horn casing. Often a mid -
VOICE COIL
SOUND CHAMBER
MOUTH
DIAPHRAGM
MAGNETIC
Different types of midrange driver units
STRUCTURE
FIELD COIL
TOP PLATE
MAGNET CASE
MAGNET CASE
VOICE COIL
PHASE
CORRECTION
PLUG
POLE
PIECE
SOUND
CHAMBER
"WOOL"
FILLING
DIAPHRAGM
80
VOICE COIL
C
The effect of a horn and o
,urtesy of James
B.
Lansing Sound, Inc.
diffusion element on the sound from a high -frequency driver
range driver unit and its multicellular flared horn are separate from the lowrange speaker and its complex enclosure.
HIGH -RANGE LOUDSPEAKERS (HIGH -FREQUENCY TWEETERS):
The hard -cone, small-diameter diaphragm or the hard -shelled dome diaphragm,
along with an annular ring driver unit, performs with great fidelity and efficiency in the extremely high audio frequency ranges. The hard cone and the
dome types of driver units are, in reality, simply small loudspeakers that
operate in piston fashion at the frequency ranges where the current is weakest.
These diaphragm driver units (tweeter), have large -diameter voice coils in
relationship to their small -diameter cones or domes.
Most horn units have small apexes or throat openings in comparison with
the size of the plastic or metallic dome. This results in some difficulty as the
Different types of tweeter units. Below, right: a multicellularhorn unit. Below and above, right: cone -type units
Equipmen for evaluation furnished
by Jensen Mfg. Co.
81
Courtesy of Kingdom Products. Inc.
Above: a two -unit coaxial loudspeaker. Right:
the Tannoy dual -concentric loudspeaker. The
high -frequency horn runs through the center
of the voice coil of the low -frequency cone
Equipment for evaluatidn furnished by Tannoy Co., Ltd.
dome and voice coil structure move in and out. The difficulty lies in the fact
that points on the surface of the dome diaphragm are not at the same distance
from the sides of the apex or throat of the horn. If this situation were allowed to
exist, cancellation at certain frequencies would occur, destroying the fidelity
of the higher sounds. A solid phasing core is consequently used in the throat
area in such a way that only the outer areas of the diaphragm radiate effectively
into the throat area. As in other horn -type drivers, the flared-horn throat and
flange couples the moving part of the unit acoustically to the listening room.
The dispersion of high frequencies can be accomplished in a number of
ways: the flared horn is most in use, acoustic lenses are the most efficient method, and in very high frequency tweeter units a dispersion plug at the mouth
of the unit accomplishes the job. Regardless of the system used, a simple test
involves only a selection of full -range music and your own ears. If you can
hear the higher frequencies at some position off -center from the axis of the
tweeter system, it is doing its job.
COMBINED (COAXIAL) LOUDSPEAKER SYSTEMS: It was only natural that designers began to conjecture that if more than one loudspeaker was
desirable in high-fidelity systems a way might be found of putting them all
together on one frame. This way of thinking led to the origin of the coaxial
loudspeaker. Someone took a cone -diaphragm speaker large enough to receive
a cone speaker of smaller diameter suspended from the front rim of the former.
82
The addition of a capacitor in the speaker leads feeding the small unit divided
the high and low frequencies, and the first coaxial speaker was constructed.
Many such combined units are manufactured and sold today.
It is easy to see some of the drawbacks to these early designs. One of the
chief problems inherent in this type of system involves the fact that the lowfrequency speaker is coupled by air to the small tweeter, causing the larger
to drive the smaller. In effect, this type of interference causes a form of intermodulation distortion in the higher ranges. Added to this is the fact that the
usual small cone speaker used as a tweeter is not as efficient as the larger unit;
hence, the additional driving force adds even more distortion to the highs. If
the small tweeter were increased in size for the sake of efficiency, it could cause
additional trouble by blocking the output of energy from the large unit.
In recent years, speaker manufacturers have offset some of these basic flaws
in coaxial speakers by combining the units into truly coaxial systems. Instead
of placing the high-frequency driver at the front of the cone, they have employed the interior of the voice coil as a portion of the high frequency horn,
with its own driving unit at the back of the magnetic structure of the bass
speaker. Where a dispersion unit, such as a flared horn or a multicellular horn,
is used, it is of solid construction and free from influence by the large driver.
With such designs it has been possible to lower the crossover frequency of the
large unit by virtue of the efficient self -powered mid -range driver with its own
magnetic structure. Then, to provide an extra -high range, a small tweeter
driver is sometimes placed on a bracket at the front of the large cone, without
a chance of interference through obstruction. This third unit is usually as far
off center as possible, so as not to obstruct the mid -range frequencies coming
through the center of the woofer. The final considerations in loudspeaker selection will be space requirements, budget restrictions, and finally the effectiveness of the different systems as they sound to the listener. Space considerations
will involve both the choice of enclosure and the room space that can be devoted
to the speaker system.
Eight -inch coaxial loudspeaker. Capacitor is used as dividing network
HIGH -FREQUENCY DRIVER
DIVIDING NETWORK
HIGH -FREQUENCY
HORN
1}
LOW -FREQUENCY
RADIATOR
Eeufpment for evaluation furni,hed
by
University inülyeaker
Cu.
83
A single -unit coaxial loudspeaker system will fit into almost any enclosure
designed for a single speaker. It can easily be adapted to in -the -wall mounting
or to closet-door mounting, or to other situations where extensive alterations
are not feasible. The over-all cost for achieving a desired degree of quality is
lower with a single -unit coaxial system than with a multiple -speaker system.
SPECIAL LOUDSPEAKER TYPES AND APPLICATIONS: In the course
of the development of loudspeakers, improvements have been made in conventional types, but at the same time new concepts have been produced and marketed. Some of these newer devices are simply refinements of earlier concepts;
others are completely new. These progressive steps have been inspired by the
need for more efficient and better means of reproducing audio material. A
need continues for speakers that require less room area. In all, loudspeaker
systems need meet but one basic requirement; they must reproduce sound with
true fidelity. The search for mechanical and electrical improvements stems
from the inherent weaknesses in any physical system. In overcoming these
drawbacks the inventor -designer must use every means at his command.
Knowledge of prior accomplishments in the field, coupled with technical and
manufacturing skills, has brought new units and applications into being.
THE ELECTROSTATIC LOUDSPEAKER: In the early days of radio, when
the electrostatic speaker was first proposed, it was then thought of as a condenser
or capacity speaker. It was conceived of as a cone -type diaphragm with capacitive elements as the driving force. The inception of the moving -coil dynamic
loudspeaker, however, forced such speakers out of use before further development could take place.
The modern electrostatic speaker was first brought into use as a high -frequency tweeter; it found its first widespread use in conventional factory -assembled phonographs. About all it accomplished was to make the listener more
aware of the surface scratch in standard disc records. Later, with the addition
of certain refinements, the range of operation of electrostatic units was broadened so that more advantageous crossover frequencies could be obtained between them and the conventional cone -diaphragm speakers in the same system.
LOW -FREQUENCY DRIVER
MID -RANGE
DRIVER
drivers covering various frequency ranges in a triaxiat speaker.
The mid -range driver horn passes
through the voice coil form of the
low -frequency driver. The high -frequency driver is mounted off -center
within the low -frequency cone
The
HIGH -FREQUENCY DRIVER
Equipment for evaluation furnished by Jensen Lire.
84
MOTION OF PLATE
Left: A simplified block diagram of
an early condenser loudspeaker
CONNECTING ARMATURE
FIXED PLATE
SPEAKER CONE
MOVING
PLATE
Below: The basic building block unit
used in the Janszen electrostatic
loudspeaker
POWER
MODULATING
DEVICE
INSULATED
MOUNTING
AUDIO
PLATE
AMPLIFIER
POWER SUPPLY
STATIONARY ELECTRODE
CONDUCTORS
A
BIAS SUPPLY
A SINGLE ELEMENT
O
o
O
O
B
TO POWER
1
CUTAWAY VIEW
AMPLIFIER
DIAPHRAGM
T
STEP-UP TRANSFORMER
N
O
INSULATING
STATIONARY ELECTRODE
CONDUCTORS
O D®
SHEATHS
O
O O
0000001000
DIAPHRAGM
50
O
O
O
PLASTIC -COATED ELECTRODE
CONDUCTORS;
DIAPHRAGM BEHIND
Above: A schematic diagram of the
Janszen electrostatic loudspeaker.
The actual placement of the elements is indicated below
-
100
a
40
H
PROTECTIVE SCREEN
wo
-30
z0
-20
12
a
W
h
-101=
1
-L
I
I
200 500 1000 2000 5000 10000 20000
FREQUENCY IN CYCLES PER SECOND
AXIAL ACOUSTIC PRESSURE RESPONSE
C
OF A SINGLE ELEMENT
FOR CONSTANT -VOLTAGE INPUT
Electrostatic speakers employ straightforward electrical concepts in operation. Objects with opposite electrostatic charges attract each other in accordance with certain physical laws. First, the force of attraction is directly affected
by the amount of static charge that exists between the two objects. Small charge,
small attraction; large charge, strong attraction. Secondly, the distance beween
the two objects governs how much attraction there will be for any given static
charge. Imagine two sheets of rigid clear plastic with different static charges,
one positive and the other negative. If these are mounted 1/4 in. apart in some
sort of rack, a certain force of attraction will exist between the two, dependent
upon the degree of the charge and the distance between them. This force is
divided equally throughout the entire area of the two sheets of oppositely
charged plastic. This is, essentially, the condition that exists in an electrostatic
loudspeaker before any audio signal is fed into it. Suppose one of the rigid
plastic sheets is fixed in the rack; the other is placed in a flexible frame so that
it is free to move in or out. The flexible sheet will naturally move toward the
rigid sheet according to the amount of static charge existing between them. If
the static charge were to be varied in response to an electrical audio signal, the
thin sheet would move back and forth, simulating mechanically the variations
85
JANSZEN MODEL
1-30 SCHEMATIC
RI
5 WATT
C
6 mh
GREEN -
3 mh
0R
1
1000 n,
8.n.
5m
YELLOW -Oa
16,n.
270
CANARY
K
a
T
2
5 WATT
o
IM
Right: Schematic diagram of an electro-
IM
static loudspeaker
IM
22M 22M
22M 122M
circuit
Courtesy of Janszºn, Inc.
IM
22
M
CL AR OSTAT
SERIES 43
AGC
117
V
5000
,n,
l
-
0
8
60 4s
2 WATT
AC
WIREWOUND
22 M
.05µf
600 V
22
22 M
1.4
05µf
T600V
»
o->
A, NORMAL
MB
STARTING
M
-1100V
'4
A through F: In sequence, the process of ionization in an electrostatic high -frequency loudspeaker in diagrammatic form
STATE
IC) IONIZATION AND EMISSION OF LIGHT
.
Tri l
rnrJ
AT PEAK
_111 LLI_LÍJ111111_
R.
Q
s
o
'
¡>
/
MD)
.
7.717-11>
.
0
t5
'
POSITIVE IONS CONTROLLED BY
ELECTRICAL
MODULATED
.
' .
.
.
O
i
STRESS
MINIMUM
©
II,Q
R. F.
o
o
(F) DECREASED STRESS;
DECREASED NUMBER OF IONS;
LESS
AGITATION
Right, above and below: Alternate pressure
and rarefaction of air is a result of the
changing electrical stress field in an elec-
trostatic loudspeaker
1
.
OO
O
(E) INCREASED ELECTRICAL STRESS;
INCREASED IONS;
LINES OF STRESS
MODULATED
F
.Oo
INCREASED AGITATION
..
R.
F
O
o
HORN
IONOVACC CELL
i
Left: Ionization occurs in the narrow
throat of the lonovac
loudspeaker horn
IONIZED AIR
SOUND
WAVES
R. F
MAIN CHASSIS
---HORN
-1
ASSEMBLY
CELLI
AUDIO
IN
Right: Schematic circuit diagram of the
lonovac loudspeaker
Dt
SIMPLIFIED SCHEMATIC
or loNovAc
Referring to earlier discussions of conventional cone
diaphragms, you can see that the moving plastic sheet, if properly constructed,
could produce sound.
This is, of course, an oversimplified description of the electrostatic loudspeaker, but in essence it is true. In the actual case the statically charged sheets
are metallic, and as they move a rather serious drawback occurs. The spacing
of the two plates governs the force of attraction with any given charge on them.
In motion, however, this force does not simply increase or decrease directly
with distance; rather, as one sheet is moved away, the force decreases dis in static charge.
88
proportionately rapidly as the distance increases. The same thing occurs as
the plates move toward each other; the force of attraction increases more and
more rapidly as they move together. The increment of attraction varies inversely with the square of the distance; another case in which the law of inverse
squares applies. The effectiveness of the electrostatic speaker thus decreases
as the separating distance increases. To counteract this effect somewhat, designers have employed one fixed -charge plate in the center, and on either face of
this sheet used two movable changing -charge plates. Thus, one plate is repelled
as the opposite plate is attracted. In this push-pull movement the effectiveness
of the unit is not impaired.
In constructing these push-pull electrostatic units in large sizes it is possible to bring down the low end of the response curve to a point where a
crossover frequency of 600 cycles is possible. When a woofer and a divider
network of high quality are used, sound reproduction can be accomplished
smoothly over a very wide range of frequencies without the stridency and harshness found in the early tweeter -type electrostatic units. Coupled with the fact
that the whole charged speaker plate area moves in and out at one time, there
is no chance of break-up or of serious intermodulation distortion, as may occur
in some conventional speaker units.
THE IONIZATION HIGH-FREQUENCY LOUDSPEAKER: It is strange
how many devices have grown out of what was once considered simply an electrical phenomenon. The ionization speaker is just such a device. It operates
on the principle that two oppositely charged objects will cause the air between
them to ionize when conditions of electrical charge, relative position, configuration, and humidity are right. When ionization occurs in air or other gases
(neon, fluorescent lights, etc.), it is visible through the light that is formed.
Ionization-or properly, corona discharge-was first noted and misunderstood by the mariners of ancient times. As the early ships plied their ways over
the oceans they would sometimes build up a static charge, either positive qr
negative. Such a static charge is the same sort you can produce by scuffle
your feet on a thick carpet. When the charge is sufficient and an opposite??
charged object is near, such as a doorknob or another person, a spark can
occur. On the sea such a static charge will build up enough so that it will dis Right: Operational sketch and circuit diagram
of a new type of ionization high -frequency
speaker. Below: A high -frequency speaker work on the ionization principle
Equipment for evaluation
furnished by Electro -Voice
CONTROL ELEMENT
EMITTER AXIS
-x
COLLECTOR
IONIC
SPEAKER
10 KV
89
4-
charge itself into the air from any high, sharp point such as a yardarm or
masthead. The charge is large enough, and the rate of discharge slow enough,
so that instead of a quick spark a long-lasting glow can be seen. The ancients
called this phenomenon Saint Elmo's fire. It was believed to be the corpus
sancti, the body of Saint Elmo, patron saint of the sailor, coming to watch over
the ship and its contents. At times, when the sea was calm enough, a sound
could be heard coming from the weird light. It is this part of ionization that is
important to loudspeakers.
In a confined area, air at rest does not produce sound. In this new concept
in loudspeakers, the confined area is the throat of an ionization chamber. The
molecular distribution of the air at rest is undisturbed. As the static charge in
the ionization chamber is built up, electrolysis takes place in the air and ionization occurs, represented by the emission of light. If the static charge is maintained by an electronic radio -frequency oscillator and a high voltage supply at
a fixed level, no audible sound can be heard because of the disturbed air. The
electrical stress field caused by the existing static charge remains constant when
no audio signal is applied to the accompanying electronic circuit. When an audio
signal is applied, 20-megacycle radio -frequency oscillations are modulated
(varied) according to the signal voltage. There is then a variation in the electrical stress field in the chamber. This results in a compression of air. As the
audio signal approaches its negative peak in the other portion of its cycle, a
period of minimum ionization occurs, represented by a minimum electrical
stress field. At this time a rarefaction occurs. These alternate compressions
and rarefactions of air result in sound.
At this point in its development the ionization loudspeaker is limited in
usefulness to the relatively high audio frequencies, within a range of 3,000
to 50,000 cycles. The ionization speaker system affords the engineer a unit that
has virtually no critical mass, and hence is not subject to the ills of cone diaphragm and voice coil speakers.
Improvements in the two -element ionization speaker have been made
through the use of an added control element. With this system, an ion stream
Charles Parsons' Auxetophone. The first practical compressed -air "loudspeaker"
OLD-STYLE SPRING -WOUND PHONOGRAPH
ACOUSTICALLY
OPERATED AIR VALVE
t
SLOW BELT-DRIVE SYSTEM
RECIPROCATING DEVICE
i
,
%
BELLOWS
is produced between a highly positively charged collector ring and a negatively
charged emitter. The corona discharge (ion stream) is maintained at a fixed
level to prevent arcing. The control element, with its relatively positive charge
varied by the signal voltage, is placed near the emitter and acts much as the
grid in a conventional vacuum tube. The need for a complicated oscillator circuit
is thus eliminated, making for a simpler and less expensive system.
THE COMPRESSED -AIR LOUDSPEAKER: The independent attempts of
Thomas Edison, Chichester Bell, Charles Parsons, and others to produce a practical means of sound amplification through the use of compressed air were mentioned previously. Edison called his patented machine the Aerophone and
Parsons, who produced the most practical device of this type, called his the
Auxetophone.
In reality the compressed -air loudspeaker and the compressed -air amplifier
are one and the same device. The Auxetophone operated on the principle that
added outside energy was necessary to provide for the amplification of sound.
By the use of a source of compressed air in conjunction with a sound -operated
valve (in this case the valve -operating sound was a voice or vibrations from a
mechanical phonograph pickup) amplification could be obtained.
Basic principles have changed very little, but refinements in electronic driving circuits and the modes of operation of modulating valves have brought
compressed -air speakers and amplifiers into use in many specialized applications. The fact that tremendous energy gains can be accomplished with a
Right: A compressed -air
loudspeaker. Below: An
exploded view of the
amplifying section of the
compressed-air speaker
Courtesy Cook Itesearch, Inc.
1
minimum of equipment gives this system widespread use in air-raid warning
devices, replacing conventional rotary sirens. Through modern developments
by Stanford University and Cook Research the audio range of compressed-air
units has been increased to a point where large amounts of power can be distributed over wide response limits. Work is under way on a unit that will cover
the entire audio range. The use of these compressed-air devices will most
certainly be limited to industrial and military applications.
HIGH -POWER INDUSTRIAL AND MILITARY CONVENTIONAL LOUDSPEAKERS: World War II saw the first wide -scale use of extremely high-power
units. These devices came into use in areas where there were heavy ambient
noise levels from machinery or battle. No problems existed in building an amplifier that would produce continuous power of 500 to 1,000 watts. However,
certain problems arose in finding a conventional voice -coil -diaphragm speaker
that could handle this power without breakdown or undesirable distortion. The
problem was solved by employing as many high -power driver units as were
needed in a single bank. These driver units looked very much like the conventional wide diameter voice coil, dome-type high-frequency driver units used
today. In order to cover the limited range of audible communication frequencies,
the dome-shaped diaphragms were constructed of heavy resin -impregnated materials. Large magnetic elements were used to give them the power reserve necessary under all conditions.
Where these units were used on submarines, the diaphragms were constructed in such a way that limited movement produced sufficient power, yet
great pressure during submersion would not destroy the mechanisms.
Peacetime applications for high-powered systems exist in railroad marshalling yards, jet airstrips, and large industrial plants.
THE SERVO-COIL LOUDSPEAKER: A servo coil loudspeaker is an ordinary permanent magnet loudspeaker with an extra voice coil winding. The
added voice coil serves to damp unwanted vibrations of the diaphragm. The
desirability of a well -damped cone arises from the fact that if a momentary
signal is fed to a loudspeaker the cone will continue to oscillate after the
signal has ceased. These oscillations are analogous to the operation of a pendulum, which will continue to swing until friction finally brings it to a halt.
The same situation applies in the case of loudspeakers. Unwanted loudspeaker
vibrations are usually damped mechanically; lightweight cones and soft suspension systems help keep them to a minimum, as does the high flux density
(magnetic field strength) brought about through larger, more powerful
magnets.
Unwanted sympathetic cone vibrations cause distortion. When a loudspeaker diaphragm vibrates after a signal has ceased, the movement of the
voice coil over the magnetic pole pieces generates current. The small voltages
generated affect the general linearity of the output circuit, including the ampliFront and rear views of a 16 -unit 500 -watt military loudspeaker with wide dispersion
92
l
PERMANENT MAGNET
Cutaway view of a high power, all-weather military horn loudspeaker
fier, the transformer, and the speaker, which may not be able to recover
soon enough to reproduce succeeding signals with fidelity.
The damping occurring in any ordinary loudspeaker is a combination of
acoustical, frictional, and electromagnetic factors. At this point we are more
concerned with electro -mechanical damping as a function of the effective
plate resistance of the amplifier circuit. In the over-all view damping, involving
feedback and special electronic circuits, is far too complicated to explain in a
few phrases. In loudspeaker design, desirable damping can be increased by
increasing the size of the magnetic structure and its flux density. Beyond this
measure and the application of feedback from the secondary of the output transformer to increase damping, the servo loudspeaker system has been developed
as an answer to truly instantaneous electromagnetic damping. In previous unsuccessful attempts to accomplish the same feat the inventors have called the
system by several names; "motional feedback" and "velocity feedback" are
among them. With two coils wound in close proximity to each other on the
same voice coil form, the second coil produces its feedback voltage by the
motion of the coil over the magnetic pole piece. This voltage is a pure motional
voltage at most frequencies and may be used to increase damping and reduce
any distortion arising from non-linearity of the diaphragm and cone suspension
system.
The only difficulty that may arise in this system may come into play at very
high audio frequencies, where mutual inductance may occur between the feedback servo coil and the adjacent voice coil, causing a signal component which
is dependent upon the induction between these two coils rather than upon
their motion. Counter-inductance is added to correct this at a point outside
the magnetic field.
Since a certain amount of damping is highly desirable, manufacturers
93
MOTION -
DRIVE COIL
_
ELECTRONIC
CROSSOVER
SENSING
TRANSISTOR
SERVO AMP.
FOR TWEETER
CAST HOUSING
COIL
r
TWEETER
I
SERVO FEEDBACK
DRIVE COIL
INPUT
FROM
TUNER
OR
PREAMP
TRANSISTOR
SERVO AMP.
ELECTRONIC
CROSSOVER
SERVO
FEEDBACK COI!
MOTIONSENSING
COIL
FOR MID -RANGE
MID -RANGE
VOICE COIL
I
SERVO FEEDBACK
ELECTRONIC
CROSSOVER
-
POLE
MOTION,
DRIVE COIL
SENSING
TRANSISTOR
SERVO AMP.
FOR WOOFER
PIECE
MAGNET
COIL
WOOFER
L
SERVO FEEDBACK
15 IN. WOOFER UNIT
MID -RANGE MAGNET
9
MID -RANGE CONE
SERVO
FEEDBACK
COILS
Above: A basic block diagram of the Integrand
-way servo -speaker system. Right: The types
of speaker units used in a servo system
HIGH
FREQUENCY
UNIT
3
VOICE COILS
HIGH FREQUENCY
MAGNET
DAMPING RING
MID -RANGE AND TWEETER
have employed one or all of the existing systems of obtaining critical damping.
However, none of these systems ever provided the complete answer to successful damping until the development of the servo -speaker -amplifier system produced by Integrand. Taking a cue from industrial engineers and their servo
systems, designers added the second coil; with accompanying circuits a continuous "auditing," or sampling, of the output signal condition in the voice coil
winding could be taken. This sample, in the form of feedback voltage, can be
sent back into the amplifier as instantaneous current factors, calling for corrective damping as needed.
THE OPERATION OF THE SERVO SPEAKER -AMPLIFIER SYSTEM:
The production of operational audio -power transistors is the major factor in
making servo systems possible. These new amplifiers make it possible to provide any speaker with a direct -coupled, output transformerless driving source.
The Integrand Servo Speaker Amplifier System, made in Westbury, New
York, is the only available example of this new working concept. It uses a
three -speaker system, with an individual woofer, mid -range speaker, and tweeter. Each speaker has its own transistorized power amplifier and crossover/
divider network. The loudspeakers are of special design, containing extra heavy magnetic systems and dual coils on each voice coil form.
SENSING COIL
VELOCITY NET
VOICE
COIL
,.
X OVER
A
diagram of
a
used in
SERVO NET
_L POWER SUPPLY
compact transistor amplifier, as
a servo -speaker system
A special multiple installation
speaker
utilizing the construction elements of a
home. Builder Sherwin Janows used
two -inch -thick solid
wood baffle boards
The servo coils act as sensing units, feeding back corrective "information"
to each individual servo-amplifier network. The information data instantaneously and automatically corrects for the effects of listening room acoustics,
cabinet or enclosure resonances, and distortion arising from inherent amplifier
characteristics. Nonlinearities, which occur in most loudspeaker suspension
and magnetic structures, are virtually eliminated through the feedback "cueing" from the sensing coils. These coils are wound over each driving coil in
such a way as to be in the exact center of the magnetic field; the information
voltage will thus be as nearly linear as is possible. Operating on a principle
of velocity feedback voltage, a change to current feedback is necessary. This
is accomplished through a conversion network, which produces a pressure
control of the sound output rather than the initial velocity changes. The resultant output is remarkably free of the usual audio system distortions.
THE INSTALLATION OF LOUDSPEAKERS: No matter what the choice of
enclosure may be, the installation of any loudspeaker must be guided by certain acoustical and mechanical principles. The acoustical rules mainly concern
type of grille or covering, placement, and choice of unit. The mechanical
rules mainly involve the mounting structure and the method of mounting the
speaker.
Let us first go into detail on coverings for the speaker. A loosely woven
fabric of almost any type can be used to cover a cone -diaphragm speaker used
to radiate the lower frequencies. If the speaker is of the coaxial variety soft,
A
Klipschorn
enclosure
properly located. The
corners of the room are
used as extensions of the
horn
Equipment for evaluation
furnished by Klipsch Aasociates
Above: Mounting a speaker on a baffle. board.
Screws should be tightened uniformly to avoid
warping speaker frame. Left: Technicians Richard
Hilton and Theodore Osborne reassemble an
enclosure after testing it
loosely woven fabric can obstruct and absorb the radiation from the high frequency tweeter. If any grille cloth is to be used at all in such a case it
should be a tightly stretched, open-weave, hard cloth (nylon or commercial
plastic grille cloth) . In factory -finished enclosures special grille materials
are used which fit the particular speaker system; additional coverings should
not be used.
The covering that goes over the port below the speaker in a bass-reflex
cabinet is as important as the material covering the speaker itself. The wrong
cloth can change the characteristics of the enclosure. Again, open -weave,
hard cloth, tightly stretched over both openings, is the best bet. Experiments
can easily be made by the interested to determine just what effects various
cloths can produce.
The placement of a loudspeaker system is governed first by the size and
shape of the listening room. With the exception of stereophonic speaker placement, corner placement usually provides best operation. The corner should
be chosen with regard to frontal obstructions, and, in the case of horn -type
enclosures, adjacent wall areas should also be free of major obstructions. It
is not always possible to have absolutely free wall space in the average living
room, but a simple solution to the problem is to move the furniture just a
few more inches away from the wall. In installing the base unit, try to get
it as near the floor as possible. High -frequency units should be elevated and
as free of obstructions as possible. If a corner is not available, try to place
the speaker system in the center of a short wall of a rectangular room, as close
to floor level as possible. The bass response of a system is diminished as less
desirable locations are chosen. Corner placement is best, the. middle of a short
wall at floor level is next best, and positions at eye level, at the center of a
room, or behind diffusing or obstructing objects are least desirable.
Your choice of a speaker unit or system should be established by your
budget, existing amplifying equipment available, room for placement, and
common sense, which tells you that one can put too many large speaker systems in a room that is too small for them.
The mechanical rules of good practice are in the main determined by your
enclosure. If you are mounting the speaker system in a factory- or kit -assembled
enclosure, you are almost sure to use the mounting board supplied by the
manufacturer. Manufacturers, for the most part, realize the importance of
having a securely braced mounting board at least 3/4 inch thick. In some better
grade enclosures these mounting boards go up to 21/2 inches in thickness.
In assembling a homemade enclosure from the plans furnished by a manufacturer, you will be wise to follow his instructions to the letter, without
substituting parts. Kits are laid out with all sections ready for assembly, and
substitution is impractical.
96
The actual mounting of a loudspeaker to a mounting or baffle board requires
good mechanical practice. Since a loudspeaker is a device that depends upon
concentric motion of a voice coil over a magnetic pole piece without any
warping stresses on the paper cone diaphragm, it is absolutely essential that
the mounting surface be flat and that the speaker be set firmly upon it. When
placing a speaker over factory -tightened mounting bolts, take care not to
puncture the diaphragm. After the speaker is positioned it is wise to tighten
the units all around the speaker frame, tightening each only slightly until all
are secure. If each nut were tightened fully in turn, warping of the speaker
frame might occur. After the speaker has been mounted, but before the grille
cloth is stapled in place, see that the speaker cone is free of wood splinters or
obstructions.
THE MAINTENANCE OF LOUDSPEAKERS: There is very little that can
be done in the way of preventive maintenance for the conventional loudspeaker
except to use it with care. In this day of the high-powered amplifier, it is wise
to install fuse protection in the leads feeding the speaker. Without this protection it is possible to burn out the voice coil windings should the speaker
be played at extremely loud levels or during large transient peaks of power.
The amperage rating of the fuse should be less than the maximum current the
speaker is designed to carry. Some amplifiers provide fuse holders for both
the a.c. power line and the speaker line. In high -power public address speakers, protective devices similar to circuit breakers are employed to safeguard
the speaker voice coil.
As for the actual use of your equipment, it is not wise to move a loudspeaker and enclosure onto a porch or patio, where moisture and dampness
might attack and dissolve the paper pulp speaker diaphragm. Most highfidelity speaker units are designed for conditions of average indoor temperature
and humidity. If necessary, special outdoor weatherproof speaker units with
metal re-entrant horn-type enclosures are available.
When installing your loudspeaker or checking it, a wise maintenance procedure is to take care that no small particles of iron or steel, such as might
be found on a workbench or a basement floor, get into it. Iron filings and
steel wool are fatal to a high-fidelity loudspeaker with the usual high-powered
magnetic system.
It is often necessary to solder wire leads to voice coil terminals; during
this process drops of solder may fall into the space between the speaker basket
Left: Pictorial and schematic diagrams of a fuse
in the speaker line. Below: A simple ohmmeter
resistance -checks voice coil
Meter furnished by Heath Co.;
speaker by Wharfedale
AUDIO AMPLIFIER
PRI
SEC
OUTPUT TRANSFORMER
frame and the diaphragm. Being hot and still fluid, these drops will adhere
to the soft fibrous cone material. While not noticeable at the time, they can
come loose during operation and cause rattles in the speaker. Care should be
exercised to avoid such particles, which could also cause wedging of the cone.
CORRECTIVE MAINTENANCE FOR LOUDSPEAKERS: High-fidelity
loudspeakers have become such complicated mechanisms that even skilled
technicians have trouble repairing them. Factory facilities are the best bet
for any major speaker repair, especially of non-cone -diaphragm -type tweeter
units.
The problems that can occur in speakers include: open voice coils due to
burn -out or broken wires; broken flexible connection leads between speaker
frame and voice coil terminals on the cone diaphragm (these two leads seldom
burn out, but may be broken by constant flexing) and diaphragm failures,
including warping, tearing, or loose rim and/or suspension systems.
Should the voice coil of a speaker burn out with no broken wire apparent,
it is a problem for factory repair. A test with a simple volt -ohmmeter, an
instrument easy for anyone seriously interested in high fidelity to use, will
show where the open is. Broken or open flexible leads can be replaced by
the semi -skilled with a small soldering iron and very flexible woven wire. Stiff
or solid wire will tend to drag down the movement of the diaphragm.
Cone problems can sometimes be corrected by a layman. However, warped
or off -center cones are factory jobs. Sometimes, a cone rim will come loose
in spots from the frame due to dryness. Glue can be used if the whole rim
is not loose and the diaphragm off center. With the modern covered diaphragm
and suspension system, centering shims are out of the question except for
factory repair shops. Slight tears or holes in a cone can be repaired: clean both
sides of both edges with fine garnet sandpaper, being careful not to allow grit
and dust to get into the opening between the voice coil and the magnetic pole
piece. Then a thin coat of rubber cement or Goodyear Ply -O -Bond can be
applied to each side. Thin strips of onionskin paper can be used to cover both
sides of the break in the cone. If the hole was caused by a mounting bolt, and
is not a large rent (less than 1/4 inch), it is sometimes best to clean out the hole
so that the edges will not vibrate together, and then leave it open. Or, it
can be covered in the same manner as explained above, with a small circle
of onionskin paper applied only to the face of the cone, formed to the shape
of the cone or suspension rim. If you are not sure about the problem, your
best bet is a factory repair job. Be sure you have proper instructions for
;
Right: An early speaker enclosure by Jensen.
Below: An early multiple -speaker enclosure provided with an Atwater Kent radio
Courtesy Sonic Arts, Inc., Chicago
98
llli
,.
COMPRESSION
V!I
1r1411
1r1
1.6'1lr1
11
wci,
RAREFACTION
1
r'lr V1
1
r
1
A compression wave from the front of a loudspeaker diaphragm occurs at the same time as a
rarefaction cycle from the back of the diaphragm. Unless the back waves are interfered with or
altered by means of some type of baffle, they will mingle with and negate the front waves. The
physical dimensions of the loudspeaker are such that high -frequency sounds are diminished less
than low -frequency sounds. The result of operating a speaker without a baffle is a dropping -off
of bass response
packing and shipping to avoid further damage. In following this procedure
you will find that your repairs will usually be made not only quickly but
inexpensively.
THE LOUDSPEAKER ENCLOSURE: Audible sound is dependent upon the
transfer from one air particle to another of the energy of a vibrating body,
which finally enters the human ear. These air particles must be set in motion
by some means. In the case of high fidelity, the loudspeaker and enclosure
do the job.
To push any object from one spot to another, force or motion must be applied. Application of force implies some sort of contact or mechanical coupling.
For the distribution of sound energy in air, there must be a contact or coupling
between the air particles and the vibrating object. As has been explained
earlier, it is the cone diaphragm of the direct -radiator loudspeaker and the
flared column of air in the horn speaker that provide the mechanical coupling
necessary to move air for sound.
A violin string can be stretched between two points in open air; a sound
can be produced by plucking this string. However, the characteristic pound of
99
the violin would not be there without the resonating wooden body of the
violin. An open string with a small contact area could not be heard as easily
as could the string and violin body combination. No direct comparison between
the body of a violin and a loudspeaker enclosure is intended, for the violin
body acts as a resonator, adding color and character to the music played on its
strings. We desire a loudspeaker enclosure, on the other hand, to act upon
or change the energy given off by the speaker as little as possible. In essence,
we depend upon the speaker and the enclosure to work together in unison
to accurately convert the signals from an amplifier.
Since the reproduction of high-fidelity sound is, in normal cases, secondary
to other elements of life in our homes, it is ordinarily adapted to fit these
conditions rather than adapting everything else for hi-fi. The cost, size, and
location of an enclosure should be fairly well defined before a high-fidelity
system has been purchased. Even with unlimited funds, size and location
govern the final choice. In most cases all three factors (plus the sharp -eared
neighbors) effect control.
As an individual unit, a loudspeaker will function in any position and
without any enclosure, so long as energy is applied to its voice coil terminals.
It could be set face up on any flat surface. It could be hung from a string in
the center of a living room or placed alone on the floor. The resulting sound
would be intelligible, however undersirable it was to listen to, or however
limited its range. The usual first impulse would be to get it out of the way,
of course, for its purely mechanical appearance would add nothing to a room.
Now, if the thing is to be hidden, let's see how we can most effectively do the
job, enhancing the sound and appearance at the same time. This combination
of needs is what has led to the high-fidelity enclosure-and -loudspeaker combination.
Sound is produced by the back of a loudspeaker diaphragm or cone as well
as the front. Due to the in -and -out motion of the diaphragm, the two sounds
are 180° out of phase with each other. When air is compressed in front of
the diaphragm, it is simultaneously rarefied in back of it. When a compression
of a substance meets an equal but opposite rarefaction, the result is a cancellation. In practice, because the size of a loudspeaker results in some physical
interference, this cancellation is most prevalent in the low -frequency range.
The only way to utilize the full low-frequency response of a loudspeaker is
thus to eliminate the sound from one side of the speaker, or to alter it in such
a way that it will no longer cancel the sound from the other side. Usual
practice is to alter or eliminate the sound from the back of a speaker rather
than the front.
BAFFLES WHICH ELIMINATE BACK WAVES FROM THE
LISTENING AREA
perfect speaker baffling would neTheoretically,
BAFFLES:
INFINITE
cessitate placing the speaker in a wall between two identical rooms. The
mounting wall would have to be insulated against speaker vibration. All the
sound from the front of the speaker would go into one room; all the sound
from the back of the speaker would go into the other room. Since the volume
of air enclosed by each room would be identical, the cushion effect on the
speaker diaphragm would be the same front and back. No mingling of sound
would occur; baffling would be perfect. This is not often a practicable solution,
but it is one that can for all instances and purposes be duplicated either by
wall -mounting a speaker between two fairly isolated rooms or by mounting
the loudspeaker on a solid closet door, backed by sound -absorptive clothes.
Effectiveness can be improved through the use of interior weatherstripping
around the closet door.
While the usual infinite baffle is not exactly what its name connotes, if
properly constructed it can very closely approach the perfect baffle. In the
.4r214*4
100
Courtesy Sonic Arts, Inc.
set-up
Two infinite -baffle enclosures of the air -suspension type are seen in use in a stereophonic
for a business office
early days an infinite baffle was simply a tightly sealed box with only one
opening, filled by the loudspeaker. It was of heavy construction to avoid
vibration, and usually contained sound -absorptive padding to reduce the effect
of higher sounds on the speaker diaphragm.
To be effective, the size of an infinite enclosure has to be large to reduce
the effect a small volume of air has in raising the resonant point of a speaker.
Each particular speaker has a low resonant point resulting in a practical limit
in its ability to reproduce bass notes. If a loudspeaker with a resonant point
in free air of 50 cycles were placed in a large infinite baffle enclosure, that
point of resonance could be maintained. However, if the size of the enclosure
were reduced to about half the volume, due to this smaller and stiffer air
cushion, bass would be lost through an increase in the apparent diaphragm
resonance to perhaps 60 cycles. Very little can be done with most typical
speaker suspension systems to reduce diaphragm resonance to a point where
an infinite baffle of extremely small dimensions will not destroy the lowfrequency portion of the sound.
A new and practical step has recently been taken by Acoustic Research,
Inc., of Cambridge, Massachusetts, in their AR-1 and AR -2 speaker systems.
It was thought that a loudspeaker could employ the air cushion sealed in an
infinite enclosure as a spring-suspension element in the speaker system. AR
produced a speaker with only 10% of the suspension stiffness of the
usual speaker, providing a subsonic resonant point of around 10 cps. The AR
systems use a totally enclosed, acoustically sealed cabinet of less than 2 cubic
feet to gain the additional 90% of needed elastic stiffness. The small volume
of air enclosed in the cabinet acts like a spring, "stretching" as the diaphragm
moves out and compressing as it moves in. The resulting system resonant point
is 45 cycles for a 12 -inch speaker and a cabinet volume of 1.7 cubic feet. Performance is comparable to any conventional 12-inch speaker in an infinite
baffle, yet much less space is taken. The use of a critical amount of glass fiber damping material eliminates the possibility of standing waves in the
higher frequency ranges. (Too much internal filling will affect the smoothness
Back -illumination of the grille cloth shows the unusual configuration of the opening in
the front
panel of the R -J enclosure. The 8 -inch Wharfedale speaker has a foam plastic suspension rim
Left: The
R -J enclosure from the bcck as an engineer prepares to install the speaker. Below:
The AR -2 air -suspension enclosure. Note the
padded interior
SEALING AND
SOUNDPROOFING GASKET
SPACE BETWEEN -:
MOUNTING BOARD'
AND `FRONT PANEt<
KER
MOUNTIhtö
BDARD
SPEAKER PORT
Equipment for evaluation from British Industries. Inc.
102
UNIQUELY FLEXIBLE RIM
CANTED TWEETER UNITS
The components of the
-baffle
AR -2
infinite
system. The
tweeter units are angled
toward each other to
provide greater horizontal dispersion
speaker
Equipment for evaluation furnished by Acoustic
of the bass response; too little can cause a marked drop-off of bass response
under 100 cycles.) The sole disadvantage of the AR system is that a good deal
of amplifier power is required to drive it; efficiency is rather low. The infinite
baffle in this smaller modified form will become more and more common where
lightweight equipment is desirable.
THE FLAT BOARD BAFFLE: A large, flat, acoustically treated board 100
feet square with a loudspeaker placed exactly in the center might be used as
a baffle. By the time the sound waves traveled from the back of the speaker
around the edge of this enormous flat baffle, they would be so weak that no
effect would be noticed; for all practical purposes we would hear only the
front waves. As the size of such a flat board was reduced to practical dimensions,
the effects of the two waves together would grow, destroying the effectiveness
of the baffle. Most of us will remember the open -back radio cabinet, prevalent
from the late 1930's up to the present day, which was in effect simply a flat
baffle with its edges turned back to form a box. Better than previous speaker
enclosures, it provided the listener with a false or "booming" bass, with
every instrument producing low-frequency notes sounding pretty much the
same. In such baffles attention was seldom given to size relationships between
The AR-2 loudspeaker system as supplied by the manufacturer
103
PORT OPENINGSPACE
o
FLAT BOARD
BAFFLE
-r
Equipment for evaluation furnished by Allied Radio Corp.
Above: The port and the shelf are acoustically
important parts of this bass reflex enclosure.
Left: A baffle and its folded -side equivalent
FOLDED -SIDE BAFFLE
(OPEN BACK)
the front and depth dimensions and to the strength and thickness of the walls
of the box. The thin walls of such baffles vibrated violently, producing their
own characteristic sounds, which, combined with the sounds from the front
of the speaker, produced completely unnatural "noise."
BAFFLES WHICH MODIFY AND UTILIZE THE BACK WAVES FROM
A LOUDSPEAKER: Early in the history of loudspeaker enclosures someone
thought it might be a good idea to use the sound energy coming from the back
of a loudspeaker rather than trying to get rid of it. Several enclosures have
been designed which do just this; the first successful design was the bass-reflex
enclosure. A bass -reflex baffle is an enclosed cabinet with two openings at the
front, one for a loudspeaker and another called the reflex port. With an en The
operating characteristics of
a
bass -reflex loudspeaker enclosure
BACK WAVE FROM
SPEAKER OUT OF
PHASE WITH
FRONT WAVE
FRONT VIEW
FRONT COMPRESSION
WAVE FROM SPEAKER
IN PHASE WITH
COMPRESSION
WAVE FROM PORT
OPENING
104
CUTAWAY SIDE VIEW
closure of correct size and with the right openings in relationship to a particular speaker, it is possible to reverse the phase of the sound energy coming
from the back of the speaker. This reversal occurs near the resonant point
of the system. Back waves are fed into a room in phase with, and in the same
direction as, the frontal energy from the speaker. When correctly designed
and constructed, the bass -reflex remains today one of the best enclosures at
a reasonable price. The outstanding feature of such units is that by changing
the size of the port opening (called "tuning the port") an enclosure can be
made to operate with almost any loudspeaker. Tuning the port to the resonant
point of the speaker limits the diaphragm excursion at this point. Tuning is
accomplished by experimentally changing the size of the opening temporarily,
using something solid as a cover. As a small d.c. voltage from a 1.5 -volt flashlight cell is applied to the speaker wires either a "click" or a ringing thump
will be heard, since the speaker diaphragm is relatively undamped (uninhibited) at its low -frequency resonant point. This is an undesirable condition,
caused by a too-free motion of the diaphragm at this one particular frequency.
By tuning the port, the movement of the diaphragm can be limited at this
point, causing the thump or bong to become a "click" as the battery voltage
is applied and removed.
Once port tuning has been accomplished the speaker will function at its
optimum, and a permanent tuning cover can be put in place from the inside.
If the speaker is changed or if the location of the enclosure is altered, retuning
is wise. The general effect of a properly constructed and tuned bass reflex
cabinet is that of smoother and lower bass response with little transient distortion, the mid- and high -frequency ranges being reproduced in quality consistent with the loudspeaker employed and the size of the enclosure.
The desire for small bass-reflex enclosures giving smooth response over
a wide range has brought about several design innovations. These include
variations in internal construction, in which the port is covered at the top
by a shelf extending into the enclosure's interior. This increases the effective
isolation of the speaker and the port while still maintaining proper baffling
at reduced enclosure sizes. Certain other internal dividers and spacers are
said to give additional smoothness of transition between the low- and highfrequency ranges. The additional cost of such complicated structures over that
Right: The design of the
Acoustical Labyrinth (TM
Stromberg -Carlson) enclosure forces the back
waves from a speaker to
travel a measured distance before emerging.
Below: The straight-line
equivalent: Back waves
emerge with 180 -degree
phase reversal
SIDE VIEW
FRONT VIEW
LOUDSPEAKER
ABSORBENT MATERIAL
f
HALF THE SPEAKER'S RESONANT -FREQUENCY WAVELENGTH
105
I
L
warranted by the slight increase
in quality.
THE ACOUSTICAL LABYRINTH BAFFLE: One of the oldest and most
successful baffles is the acoustical labyrinth, used for so many years by the
manufacturers of Stromberg-Carlson radios at a time when all other set manufacturers were still employing less -than-adequate open-backed cabinets.
Every loudspeaker diaphragm has one dominant point of resonance in the
range of frequencies it will reproduce. At this point, low in the scale of
frequencies, the cone moves in and out more easily and does not come to
rest quite so quickly as at other frequencies. The labyrinth enclosure was
designed to lower the resonant point by introducing behind the loudspeaker a
tuned (to a specific size and length) column of air which maintains a definite
loading value at the resonant frequency point. The length of the column of
of the conventional bass -reflex unit is not
air is exactly one-fourth of the wavelength of the frequency of the resonant
point of the speaker being used. This column forms an anti -resonant chamber
that restrains the motion of the speaker at the resonant point, smoothing out
and lowering the otherwise peaking bass end.
Labyrinth enclosures are most conveniently constructed in box form, with
the tuned air column formed by dividers placed in such a way that a bent path
is formed. Once constructed, the inner surface of the path is lined with soundabsorptive material to kill any high -frequency reflections or standing waves
that might have an effect on the diaphragm of the direct -radiating loudspeaker.
The port at the end of the labyrinth forms bass reinforcement at a frequency
where the air column represents 1/2 wavelength.
HORN AND FOLDED HORN ENCLOSURES: The flared -horn shape was
recognized as a superior means of coupling speaker diaphragm motion to air
as early as 1919. At that time Dr. A. G. Webster patented the exponential -horn
shape. Webster suggested in his patent that anyone building such a horn
would do well to make certain that it formed a "reasonably rigid boundary
for an air column."
FOLDED HORNS: It wasn't until electro -dynamic loudspeakers came into
use in motion-picture theaters that horn designs became a permanent part
of the sound -reproduction art. However, the honor of devising and putting to
work the first folded corner horn of practical size went to Paul W. Klipsch in
1940. This system is unique in that it has the air column required for reproducing very low -range sounds, yet does not use up half the area of a room.
The Klipsch-horn does this by using the walls of a room as an extension of the
folded horn, which is placed in a corner of the room.
A horn loudspeaker for the lower ranges of sound reproduction would of
necessity be very large and long. If the horn went straight out into a room
without being folded, the device would resemble a large square -sided funnel,
over 3 feet square at the front opening, tapering back for 6 feet to a 12 -inch
VOLUME CURRENT=1
PRESSURE=100
DIAMETER=1
6
The
VOLUME CURRENT=100
PRESSURE=1
DIAMETER=100
106
multiplying action of a
loudspeaker horn. Usual
practice is to fold horns intended
for low -frequency
reproduction for the sake of
compactness
A. A straight, unfolded horn
of exponential shape. B and
C.
Two
compact folded
horns. D. A folded -horn
loudspeaker enclosure intended for reproduction of
low -frequency sounds
D
loudspeaker at the rear in a corner of the room. It would be anything but
inconspicuous, and at that it would only reproduce sounds well down to 60
cycles.
In order to conserve space, the folded horn was developed and has come
into popular use. If constructed correctly, it will reproduce the intended
sounds without adding to or subtracting from them. In "folding" a horn, it
is very important for the designer and manufacturer to regard the ratio of
sharpness of the bends to their lateral dimensions, which may approach a
half wavelength at the crossover point between low and mid ranges. Serious
attenuation may occur at this point, noted by a change in character of midrange "color" and power output. In many less -expensive folded horns, thinner
construction materials and poor design produce undesirable changes in reproduced sound due to internal structural vibrations and standing waves.
A horn or folded -horn loudspeaker is not necessarily intended for corner
placement. Many horn units are intended for side -wall placement and function
with exceptional fidelity. Most motion-pitcure theaters employ some form of
flat wall positioning of a horn speaker behind the motion -picture screen. In
any application of horn or folded -horn loudspeakers one must consider the
fact that the horn itself is simply a means of guiding sound into the listening
area; it must not radiate or vibrate itself. If it does, the horn will be no more
efficient than a plain direct-radiator loudspeaker diaphragm. Often stressed as
a sales point for horn speaker systems is their better than 50% efficiency, as
compared to 15% or lower for direct -radiator speaker systems. However, it
should be remembered that efficiency has no direct bearing on quality. Naturally, an inefficient loudspeaker will not function well with an amplifier which
107
nt br evaluation furnished
by ..i n.
S. i...i
m..
MID -RANGE HORN
Equipment for evaluation furnished
by Klipseh Amoeiatea
iv
Above, left: The James B. Lansing
Hartsfield speaker system features
heavy-duty construction and well integrated driver units. The full
audio spectrum is reproduced ac-
curately with such equipment.
Above and left: Cutaway views of
the Klipsch folded -horn enclosure.
The walls of a room act as extensions of the corner enclosure
108
Rigid cast -metal multicellular horn features wide dispersion. Horn is attached to
a high -frequency driver
is seriously overtaxed by its requirements. In A -B switching tests with one
+m`
amplifier and two speakers, the more efficient speaker will seem to have
the edge, due to its greater loudness. Loudness, of course, is not the criterion.
HIGH -FREQUENCY HORNS: Contrary to popular belief, horn enclosures
are not directional at low frequencies. In the higher ranges, however, a horn's
directivity increases. The dispersion of high-frequency sounds over a wide
area constitutes a problem for manufacturers of high-fidelity speaker systems.
In some cases manufacturers depend simply upon increased flare in the exponential horn shape, but too often the problems of acoustical phasing detract
from gains in dispersion.
THE MULTICELLULAR HORN: The multicellular horn comprises several
individual horns directed outward into the listening room at slightly different
angles; all are powered by one high -frequency driver unit. Cast in rigid metal,
these horns may have as many as 15 sections, and may cover an angle of 130°
horizontally and 90° vertically. Though used in some high-fidelity speaker
systems, most frequent application is in motion -picture sound installations.
THE ACOUSTICAL LENS: While it is not exactly correct to refer to an
exponentially shaped metal horn as an enclosure, in some ways it accomplishes
the same thing. An exponential horn is a coupling device between the
diaphragm of a driver unit and a listening area. If the horn is correctly designed
it will beam sound over quite well-defined areas. The higher the frequency of
the sound to be beamed, the more restricted the area of dispersion in front of
the speaker, until at the high end of the audible spectrum-needed for good
high-fidelity reproduction of sound-the area is almost too limited to be
practical. While acoustical lenses have been the subject of much investigation
by Bell Telephone Laboratories over the past two decades, the James B. Lansing
Sound Corporation put the "Koustical" lens to work in the high-fidelity field.
109
LARGE MOUTH ALLOWS WAVE
TO EMERGE SMOOTHLY
WITHOUT REFLECTANCE
THIN WALLS "RING"
AT CERTAIN FREQUENCIES
SOUND WAVE EXPANDS
GRADUALLY ALONG
LINEAR AXIS
FOLDS INTRODUCE RESONANCES
AND STANDING WAVES
SMALL THROAT
FOR
OPTIMUM COUPLING
HORN PATH NOT
TO DRIVER DIAPHRAGM
TRULY EXPONENTIAL
RIGID NON -ABSORPTIVE WALLS
THE EXPONENTIAL HORN
INEXPENSIVE RE-ENTRANT HORN DESIGN,
FOR HIGH-FIDELITY REPRODUCTION
UNSATISFACTORY
SMALL SOLID OBJECT
SPHERICAL WAVEFRONT EMERGES
PLANE WAVEFRONT
MOLECULAR "SKIN"
FURNISHES MEDIUM
DENSER THAN AIR
SOUND FLOWS
AROUND OBJECT
WITHOUT REFLECTION
OR ATTENUATION
SOUND WAVE PASSING SMALL OBJECT
PERFORATED LENS ELEMENTS
SOUND IN
MORE OPEN
CENTRAL AREA
TRAVELS FASTEST
CROSS-SECTION OF ACOUSTICAL LENS
15
o 15
MULTICELLULAR
5000 CPS
15
HORN;
o 15
ACOUSTIC LENS;
5000 CPS
Above: The operation of the James B. Lansing
acoustical lens high -frequency driver and dispersion unit. Left: the horizontal dispersion
characteristics of a standard horn and an
acoustical lens
It was once thought that the use of lenses applied only to light. We now
know that this is not true; that there can be, in effect, lenses for any type of
wave motion, providing a substance exists which can be "transparent" to this
wave motion (radio, radar, infrared, x-rays, etc.), and yet effect a change in
speed of transference. As far as a lens system for sound was concerned, all
that was needed was a medium denser than the air through which sound
usually travels. This material had to be "transparent" to sound, yet still provide a medium denser than air.
Air must travel at great speeds in order to enable an aircraft to have
lift. Researchers in aerodynamics have found that a layer of air molecules
covers the entire surface of an airfoil at high speed. These molecules cannot
be disturbed by any force. This layer of air is called the "boundary layer,"
and in the acoustical lens it is put to work.
Any perforated material, regardless of how substantial or durable it may
be, is transparent to sound to a degree governed by the size, number, and
placement of the holes. In the acoustical lens system designed by the James
B. Lansing Corporation, many perforated metal screens are arranged together
much as a lens system might be put together for a telescope. The so-called
transparency of the perforated screens and the all-over boundary layer of air
molecules provide a denser -than -normal air path for the sound waves as they
tlo
leave the driver unit and pass through the exponential horn. In the "Koustical
Lens" 175DLH assembly there are 14 separate lens elements arranged to form
a double -concave lens which refracts sound energy evenly over a solid 90°
angle. In the course of this refraction in the acoustical lens system, less sound
is absorbed than the percentage of light absorbed in a high -quality glass lens
system.
With the acoustical lens, the listener's position need not be centered on the
loudspeaker system for him to enjoy the full range of sound, otherwise often
restricted in the higher end due to the directional characteristics of most
tweeters.
MONAURAL HIGH-FIDELITY ENCLOSURE PLACEMENT: With the exception of stereophonic systems, corner placement of any enclosure is considered best by most experts. The corner should be chosen with regard to
frontal obstructions, and, in the case of some horn -type enclosures, adjacent
wall areas should be free of obstructions for several feet on either side. Since
it is not always possible to have absolutely free wall space in a living room, a
simple solution to the problem is to move the furniture a few inches away
from the walls. When positioning the bass unit, try to get it as near the floor
as possible; the high-frequency tweeter should be elevated more and as free
of obstructions as possible. If a corner is not available, try to place the enclosure in the center of a short wall of a rectangular room, as close to the
floor as possible. The bass response of any enclosure will be diminished as
Equipment for evnIuntinn furnished by
Jnmen B. I.auvinti Sound. Inc.
Above: The components of
an acoustical lens. There are
32 interior elements. Right:
The James B. Lansing Koustic Lens with high -frequency
driver model 175 DCH. Below: Cutaway view of lens
structure
Ic-
mun
111
AMPLIFIER
MICROPHONES
-c
-
SPEAKER
-DF\
RECORDER
TWOCHANNEL
TWO CHANNEL
TAPE
PREAMPLIFIER
AMPLIFIER
SPEAKER
PLAYBACK
The
two channel stereophonic tape recording and playback process. The separation of the two
microphones approximates the separation of the ears
other and less desirable locations are chosen. A corner is best, the middle of
a short wall at floor level next best. Positions at eye level, at the center of
a floor area, and behind diffusing, absorbing, and obstructing objects are least
desirable.
STEREOPHONIC SOUND: The ultimate goal of high fidelity is to reproduce
in the listening room exactly the same sounds that a listener sitting in the
best seat of the best concert hall might hear. Since people have differences
of opinion regarding the best seat of the best concert hall, or may even prefer
music which is never performed in a concert hall, it is unlikely that we will
ever have conformity in high-fidelity recordings. Let us assume for the moment that everyone agrees on the best seat of the best concert hall. What are
the sounds the listener will hear? Amazingly enough, only a small percentage
of the sound reaching his ears comes directly from the orchestra; most of it
is sound which has been reflected from the walls of the hall. Yet the listener's
ears and brain are able to analyze this sound, coming to him from every
conceivable direction. He can locate with some accuracy the various instruments in the orchestra, and at the same time distinguish the direct from the
reflected sound. If a microphone is substituted for the listener, and the sound
recorded for later playback on a high-fidelity system, monaural or monophonic
sound results. The microphone, unfortunately, cannot duplicate the feats of
the listener's two ears. All the sounds reaching the microphone are mixed together, never to be separated again; thus, directionality is gone. Also, the
direct and reflected sounds are mixed, resulting in poor definition and listening
fatigue. If, on the other hand, two microphones are substituted for the listener's ears, binaural sound results; this very nearly approaches the ultimate goal
of high fidelity. Sound recorded and played back binaurally does not seem to
emanate from earphones. Instead, the walls of the listening room seem to disappear; the big acoustics of the concert hall surround the listener; the orchestra
is spread out in space. Unfortunately, wearing earphones is uncomfortable
and impractical for prolonged listening.
It has been found that a successful illusion of three-dimensional sound can
be gained by the use of two loudspeakers instead of two earphones. In this
case, the recording microphones are spaced farther apart, usually from 5 to 25
feet; the loudspeakers are spaced from 3 to 10 feet apart. Roughly speaking,
we are able to judge direction by virtue of the fact that an instrument will
be closer to one microphone than the other, and will thus appear closer to
one loudspeaker than the other. We can also distinguish between direct and
reflected sound because one speaker can supply the direct sound while the
other emits reflected sound from another direction, an occurrence similar to
112
-s
4e,
that in an actual concert hall. There is an additional complication in stereo
in that the sound from the speakers is reflected from the walls of the listening
room, reaching the ears from all directions. The time delay of this reflected
room sound is much smaller in most cases than that in a concert hall. Evidently,
the ear favors the larger room acoustics, since nothing could be more ridiculous
than having a 100-piece orchestra perform in the average listening room. Thus,
the function of stereophonic sound is to enlarge the listening room acoustically,
as well as to supply a sense of direction to the instruments in the orchestra.
If directionality only were supplied, the equivalent would be putting the orchestra in the listening room, depriving the music of the acoustics of the concert hall.
The ultimate goal of high fidelity is perhaps in sight, when speakers can be
hung as pictures on each wall of a room, the sound from each being equivalent
to what would emanate from the comparable direction in a concert hall. Experiments along this line are already being carried out by the Philips Co. in
Holland, known in this country as Norelco. The problem of cost will perhaps
be solved by electrostatic speakers of great efficiency, requiring low -power
amplifiers.
STEREOPHONIC SPEAKER -ENCLOSURE PLACEMENT: Placement of
speakers and enclosures for stereophonic sound reproduction has been a subject of considerable controversy. This is to be expected, since individual listen a Dynakit
The home of co-author Robert Oakes Jordan. Two AR -2 speaker units are driven by
Mark Ill amplifier (not shown.) An Ampex 612 stereophonic tape player is used
113
arrangement. Distance bebe determined
by
experiment
This arrangement may be preferable to the
others if the long walls of a room are heavily
listener s position is restricted here, hut corner
placement of speakers produces better lowfrequency sound
This arrangement gives the best stereo effect.
The most -used stereo
tween
speakers
This arrangement
reflective
walls,
should
is good for rooms with highly
but the listener's position is
restricted
Not recommended. This arrangement has all the
disadvantages of corner placement and none of
the advantages
114
draped
Mixing
is
limited because of the toe -out of the
speakers
unusual arrangement. Sound is reflected
from the back wall to the -wo adjacent walls,
An
'hen finally to the listener
Possibly the only arrangement suitable for L shaped rooms. Facing walls must be non -
reflective
For good stereo results, a
listener's ears should
subtend at least a 20 degree angle with the
centers of the loudspeakers. A position too distant from the speakers
will result in mixed
sound
Movement of a listener
away from a central position affects stereo balance. Corner placement
of speakers upsets balance most with movement
ing rooms, and the preferences of the people who listen, vary widely. No
hard-and-fast rules can be laid down to cover the placement of speakers for
"stereo," but we will propose a number of suggestions which have been tried
and tested and have been found satisfactory, some more so than others. Some
experimenting on the part of the listener will lead to good stereo speaker
placement for most rooms.
There are several "packaged" stereophonic systems which are complete
in one cabinet, carefully designed to function well in the home listening room.
The stereophonic effect is produced in your room by a combination of intensity,
time, and frequency differences at the two loudspeakers. It follows that these
conditions must be maintained by correct loudspeaker placement. For example,
if the loudspeakers are arranged so that the general listening position is much
closer to one speaker than the other, the balance of time and intensity will be
upset, resulting in a poor stereo effect. It is preferable to have the wall
opposite the two speakers as non-reflective as possible. By hanging drapes,
tapestries or rough wallpaper on this wall, you can eliminate undesirable reflections. The latter can cause general diffusion of the sound, which would
destroy the important intensity differences that should exist between the two
loudspeakers. An interesting experiment is to set up your stereophonic speaker system outdoors on a quiet summer day. Here all reflections are eliminated
and the resulting effect is rather astounding and pleasant.
The most common successful stereo speaker set-up is one in which the two
speakers face outward from a short wall of the listening room, faces parallel
to the wall, 6 to 12 feet apart. The distance between the speakers should be
arrived at by experiment. For your tests it is wise to use an acknowledged
high -quality stereophonic tape, playing the same tape for each change of
speaker position. The authors suggest a stereophonic demonstration tape
us
CHANNEL ONE
LOUDSPEAKER
ANNEL TWO
UDSPEAK=R
Speakers are usually positioned at each end of a
stereo console machine to
achieve the required stereo
separation. Other components are positioned between
the speakers
Equipment for evaluation furnished by Ampex Audio Inc.
called Sound in the Round (in two volumes), marketed and sold nationally by
Concertapes of Wilmette, Illinois, as your test tape.
If your speakers are too close together there will be little stereophonic
effect because the sound from the two speakers will be mixed together before
it reaches our ears. Of course, some acoustical mixing takes place in any
stereo set-up, but in order that the spatial effect be natural you must be able
to perceive the direct sound coming from one location and the delayed sound
from the other. Should the speakers be too far apart there will seem to be a
"hole," or lack of sound, midway between the two units, making some types
of sounds and music quite unpleasant to listen to for very long. The isolation
caused by speakers placed too far apart makes the listening area more critical,
since any movement to one side or the other will cause greater time and intensity differences between the speakers. Ideally, the listener's ears should
subtend an angle of not less than 20° with both loudspeakers. Corner speaker
placement has some merit, since most loudspeakers radiate more low frequencies in a corner position than any other. However, the position of the
listener will be seriously limited to the exact center between the two units.
The best stereo effect of any of the speaker positions discussed here is one
in which the speakers are toed out by a small angular displacement from the
wall. This placement gives the least mixing before hearing and therefore the
best stereo effect. This placement does not work well where the two side walls
are heavily non -reflective; this causes the higher frequencies to be absorbed,
rather than reflected into the room. The walls should be highly reflective for
this type of speaker situation, with the far wall covered with drapes or some
other type of absorbent material. In such a room this placement can give an
interesting "spread" to the stereo effect, with no feeling of a "hole" in the
center of the sound. In some cases there might be too much mixing for some
listeners, since sound has a tendency to become more evenly dispersed in such
reflective areas. An unusual speaker set-up for stereophonic sound is one in
which the two speakers are enclosed in the wings of an easy chair at ear
116
High -quality infinite baffle enclosure and
driver kit features heavy
construction, two low frequency drivers, heavy
cast multicellular horn
with high -frequency
driver
Equipment
by Heath Co.
level. Much like earphones, this placement is for only one person at a time,
but for the real fan there is much to be said for this stereo easy chair.
It is of great importance that your two speaker and enclosure systems be as
much alike as possible. However, this does not mean you cannot have stereophonic sound if you are not able to have duplicate equipment. The main thing
is to get started, obtaining the best possible sound with what equipment you
can afford, improving your system as you go.
LOUDSPEAKER ENCLOSURES IN KIT FORM: Even before the days of
commercial kits, audio enthusiasts frequently put together their own baffles
or enclosures. There was very little information on either exact size or methods
of construction. In recent years an ever-increasing number of commercial kits
and plans have become available. As with other do-it-yourself kits produced
by reputable manufacturers, no amount of advice from an outside source can
supplant the directions given with such kits. If a reader intends to design and
construct an enclosure from his own plans, he would be wise to stick to simpler enclosures, like the infinite or the reflex baffle; the more difficult and complex horn enclosures are well left to the more experienced.
THE INFINITE OR CLOSED -BACK ENCLOSURE: As in all enclosures, the
speaker and the baffle work together to produce the sound. The resonant point
of any loudspeaker is raised if the size of the enclosure is too small. If it is
sized according to dimensions evolved from the basic resonant point of the
particular speaker to be used, a minimum number of cubic inches of volume
must be provided.
Except for loudspeakers with especially low resonant points, the volume
requirements given here are minimums, and require completely lined inner surfaces. It is recommended that you use a speaker with the lowest possible
resonant point at any given diameter.
Minimum inner-enclosure volume requirements:
8 -inch diameter requires 4,000 cubic inches
10 -inch diameter requires 6,000 cubic inches
12 -inch diameter requires 8,000 cubic inches
15 -inch diameter requires 10,000 cubic inches
Example:
For an infinite baffle using a good quality 12 -inch speaker (e.g., the James
B. Lansing D-123) having a cone resonance of about 35 cycles, the approximate
dimensions might be 31 inches high, 24 inches wide, and 14 inches deep. Actually, any dimensions which produce the minimum interior volume requirements will work just as well.
117
A compact ducted bassreflex enclosure. Speaker leads are fastened to
a terminal strip on the
back of the air -tight
cabinet
Equipment for evaluation furnished by Allied Radio, Inc.
All enclosures should be constructed from 3/4 -inch plywood; 1 -inch glass
fiber interior padding should be used. In constructing the box, it is wise to
glue all joints to prevent unwanted vibrations and air leaks. However, a hole
box to provide a
1/16 inch in diameter should be drilled through one wall of the
slow air leak so that changes in temperature and pressure will not move the
cone from its natural position of rest around the magnetic pole piece.
KIT -TYPE INFINITE BAFFLE ENCLOSURES: Large multiple -speaker
enclosures are available in the infinite -baffle type. The speaker combination
provides full-range coverage in one box. Two woofers cover the range from 25
to 500 cycles, with an exponential -horn -type tweeter covering the rest of the
range to 20,000 cycles through a crossover network. As complicated as these
large kits may appear, each manufacturer provides adequate detailed instructions; a minimum of woodworking and electronic experience is necessary.
THE BASS -REFLEX ENCLOSURE: The bass -reflex baffle is perhaps
more desirable for the novice to build from "scratch" or from a kit, since it can
be adapted to employ any loudspeaker simply by tuning the port. The bass reflex enclosure has specific minimum external dimensions and critical speaker
and port opening sizes. Apart from these two factors, construction involves
little skill. The most important thing to remember is that a port in the enclosure
face does not necessarily insure good bass response; the port must be tuned
even on commercial reflex enclosures.
Minimum external bass -reflex dimensions and critical port size:
Port Size
Depth
Width
Height
Speaker Size
9.5" x 4"
11.5"
18"
24"
8 -inch
12" x 6"
12"
24"
29"
10 -inch
13.5" x 7"
15"
24"
36"
12 -inch
15.5" x 10"
18"
30"
42"
15 -inch
These sizes conform to the most economical lumber sizes. All sections should be at least 5/8 inch thick or better. Glass fiber padding
at least 1 inch thick should be fastened on back, one side, and top
or bottom interior surfaces. All pieces should be securely glued
and nailed together.
KIT -TYPE BASS -REFLEX ENCLOSURES: Several good kits are available
for the construction of either ordinary reflex port or ducted-port reflex enclo118
D
C
A
ACOUSTICAL
PADDING
E
3,á" TO
I"
PLYWOOD
The construction of a bass -reflex enclosure. Proper dimensions
for different speakers are given in
the text. Three sides of the enclosure should be padded
sures. May we repeat some valuable advice again: Follow the directions furnished by the manufacturer of the kit you are constructing.
CROSSOVER NETWORKS: The electrical crossover network has become
an important part of high-fidelity systems since the development of modern
wide -range audio amplifier and loudspeaker combinations. A single cone diaphragm speaker has difficulty reproducing all the possible audio frequencies
available from an amplifier. A good low -frequency unit will tend to "break
up" in the higher ranges, and a speaker that will produce good highs has difficulty operating in the lower end of the frequency response curve. A simple
and logical conclusion leads to the provision of one or more speaker units for
each portion of the response curve to be acoustically represented. This is accomplished through the use of crossover networks.
A
three-way speaker system with crossover networks and balance controls
MID -RANGE DIVIDER
NETWORK
HIGH -FREQUENCY
l
L'
LOW -FREQUENCY
DIVIDER
NETWORK
TWEETER
BALANCE CONTROL
FROM
AMPLIFIER
MID -RANGE
DRIVER HORN
BASS -FREQUENCY
DRIVER
Equipment for evaluation furnished by Jensen Mfg. Co.
119
Equipment for evaluation Itrni_;hed by Jensen Mfg. Co.
Above: The Jensen G-600 Triaxial speaker with crossover network and balance controls
Below: Electro -Voice 4 -way speaker system with dividing network and balance controls
Equipment for evaluation furnished by Electro-Volee, Inc.
120
TWEETER
CROSSOVER
PARALLEL CONSTANT RESISTANCE CROSSOVER
NETWORK
NETWORK
GRADUAL
PARALLEL FILTER
CROSSOVER NETWORK
SIMPLE FILTER
CROSSOVER NETWORK
WITH VARIABLE HIGHS
Different types of loudspeaker crossover networks
Crossover networks serve to divide the various frequency ranges needed for
each speaker in a system. As the top of the frequency division for each speaker
is reached, the particular speaker drops off in effective operation, and the next
speaker takes over in such a way that there is a smooth transition between the
two. As the mid -range unit drops off in effectiveness, the high -range tweeter
system takes over; through this division of operation the complete frequency
response range available from the amplifier is represented acoustically.
A designer has a choice of crossover points, depending on the loudspeaker
units he intends to use. It is fairly hard to find a combined middle- and high range speaker unit that will carry its response low enough to have a point of
optimum crossover with a low -frequency driver. It is also hard to find a lowfrequency driver with a range extended to a point where it will cross over
efficiently with the majority of single mid- and high -range units. To meet this
difficulty, three- and four-speaker systems have been developed; these have a
drawback of added cost. For the most part, speaker designers have been able to
make two -speaker units function well.
Some experts extol the virtues of concentric placement of whatever speakers
121
they intend to use; others prefer separate mounting of the various components
on a single mounting board. This is, however, largely a point of personal
preference, the quality of the entire system remaining the most important factor.
In any speaker system, account must be taken of the difference in efficiency
of the various units. A direct -radiating loudspeaker is apt to be far less efficient
than a mid- and high-range horn tweeter unit. Some manufacturers include
variable resistance controls in one or both speaker lines as they leave the crossover dividing network and enter the voice coil of each speaker; others provide a
means of switching in and out different values of capacitors and inductances in
the network itself. Both devices may accomplish a lowering of the efficiency
of the horn unit and a limiting of the higher frequencies. However, the more
competent manufacturers will have made provisions to account for these differences.
TYPES OF ELECTRICAL CROSSOVER SYSTEMS: The operation and
effect of a crossover network is essentially the same in all cases; the variations
occur only in the point of crossover frequency and the parts necessary to build
each system. As the crossover point between the woofer or low -range speaker
and the mid- and high -range unit is lowered, the inductors and capacitors used
in the network must be larger in size, and hence cost more; not only does a network itself cost more, but the necessity for better speakers boosts the cost of
the complete system. For example, in a constant -resistance crossover system,
using a woofer for the range between 20 and 500 cycles would involve a series
network with a crossover frequency of 400 cycles, using two capacitors and low
inductances. The capacitors would be approximately 28 mfd and 55 mfd, the
inductances 2.8 millihenries. These are relatively expensive; if the crossover
point were raised to 1000 cycles, the parts would be far less costly.
A crossover system can be made by combining a low -frequency driver that
has a decided drop-off at the high end of the scale and a smaller high -frequency
unit that has had the lower notes excluded from its circuit by the high reactance
of a series capacitor. When the output of the low -range speaker drops off, more
and more high -range energy is delivered gradually to the tweeter unit. Very
effective coverage can thus be accomplished with a modest expenditure. Most
low-cost speaker systems are of this gradual -crossover type.
THE FILTER -TYPE CROSSOVER NETWORK: To provide more accurate
power distribution between two speakers, a dividing filter network is employed, with which it is not necessary for the low -range speaker to handle and
absorb high -frequency energy, nor for the high -frequency speaker to handle
the lows. A combination of an air -core inductance in series with the low -frequency speaker lines and a capacitor in series with the high-frequency speaker
lines accomplishes this result. With a crossover point of about 1200 cycles these
components are small in size, relatively inexpensive, and quite effective.
crossover network
employed in the Klipschorn loudspeaker sys-
The
tem uses inductances and
capacitors to divide the
different frequency
ranges
122
ti
HIGH FREQUENCY
POWER
AMPLIFIER
OUTPUT
TRANS
FORMER
TWEETER
LOW-PASS
FILTER &
ELECTRONIC
CONTROLS
LOW FREQUENCY
POWER
AMPLIFIER
OUTPUT
TRANS
FORMER
CROSSOVER
NETWORK
Block
WOOFER
diagram iif an érciroñic crossover network. Two power amplifiers are used
\
ELECTRONIC CROSSOVER SYSTEMS: The previously discussed crossover networks are all designed for operation in the low -impedance circuits that
exist between the output transformer of an amplifier and a loudspeaker. However, there are cases where it is desirable to have a more versatile and variable
system. The electronic crossover system is such a device; it can be found in a
number of versions, but all work in essentially the same way: by splitting the
frequency range of the sound into two or more channels. This can be accomplished by a complete dual amplifier and speaker set-up, by dual filter network
and output stages on the same chassis, or by an electronic crossover, a device
which has its own power and is situated after the preamplifier but ahead of the
power amplifiers.
A main advantage of such electronic systems is that they are connected
into a circuit at a point where they need not handle power, as they would if
they were connected after the power amplifiers. With each channel variable
within its own range of frequencies, it is possible to compensate for differences
in speaker quality and efficiency. Drawbacks to this type of adjustable system
lie in the fact that the person operating the controls may not realize the importance of smooth crossover between the two ranges. The filter networks which
divide the composite full range signal from the preamplifier can be adjusted for
narrow or wide bandpass features. If the lower end of the upper band begins too
sharply there may be a noticeable separation between the highs and the lows.
Proper control settings will effect a smooth transition for any combination of
speakers.
123
5
DISC RECORDS
THE PHONOGRAPH CARTRIDGE: Regardless of the type of phonograph
cartridge used, and regardless of cost, the cartridge has but one function: to
get sound off a disc. The restrictions and requirements imposed upon the
cartridge in use alone call for differences. Only a handful of types of cartridges
exist, but within those few are hundreds of variations, past and present. As
with the old soundbox on the acoustic phonograph, every inventor has his own
ideas on the subject. There is one difference, however: the electrical phonograph cartridge calls for complicated industrial equipment to process even the
simplest design concepts. The results are about the same as in the days when
a kitchen table was good enough to support the experimentations of dreamers
and doers; there are some bad and some good products. The major difficulty
still remains: how does one know the difference?
Of the major types of units, the crystal, the ceramic, the magnetic (all forms),
the electrostatic, and the strain-sensitive, none is perfect, and all have their
limitations. Electronics and electro -mechanical design have done much to overcome these defects, but they still remain to some extent. If the perfect cartridge
were to be designed, these would be the requirements for a standard monaural
pick-up (they will vary for single -groove stereophonic disc cartridges):
1. The widest possible frequency response in the unequalized state, so that
no extensive equalization circuits will have to be used, thus keeping down
the cost of amplifier equipment.
2. Low stylus pressure against the record to prevent wear,
3. High lateral compliance of the stylus as it moves back and forth in the
grooves.
4. Reasonably high signal output to eliminate the more complicated pre amplifying circuits.
5. Low distortion from mechanical and electrical sources.
6. As little stylus "talk" as possible. Stylus or needle talk occurs as the
stylus itself vibrates, causing an audible sound at the record surface.
7. No noticeable hum signal at normal or low volume levels.
8. No signal voltages produced from vertical movement of the stylus in the
groove, such as the pick-up of turntable rumble or other distracting noises.
9. No effect from heat and humidity.
10. Small but rugged design.
11. Low cost.
124
CONDUCTIVE METAL FOIL
ON TOP AND BOTTOM
ELECTRICALLY CONNECTED
construction of a bimorph Rochelle salt
crystal element. When such an element is
stressed, a small current is produced between
the contacts
The
CONTACT 2
CONTACT
ROCHELLE SALT
1
SANDWICHED
CRYSTAL SLABS
METAL FOIL
CRYSTAL PICKUP CARTRIDGES: Since the piezoelectric effect was applied to phonograph cartridges, much improvement in output quality has taken
place, but the principle of ope ration remains the same. The effect got its name
from the fact that certain crystalline configurations, if pressed or twisted, produce a potential difference between two surfaces. A Rochelle salt crystal in a
modern phonograph cartridge is about half the size of a postage stamp and less
than 1/8 inch thick. Without its protective black plastic covering, it appears to
be made from very thin frosted glass. Actually, two slabs cut from a large crystal
block are cemented together with one electrode between the two, and one electrode connected to the outer surfaces of the crystal. Depending upon how these
slabs are cut from the large block, a voltage is produced by either a torque
(twisting) motion or a bending or flexing motion. This voltage is directly proportional to the distance of movement of a stylus in a record groove. The natural frequency characteristics of crystal pickups are such that they may produce 1 volt of signal at 1000 cycles, but will produce as high as 3 volts of
signal for recorded frequencies of only 250 cycles. This is not a desirable trait
for a pickup. However, in manufacture, resistive and capacitive equalization
can be added to smooth out the bass frequencies. Depending on the natural
A lever -type crystal cartridge. By means of a lever assembly, torque is increased 5:1
LOW-MASS
PIEZOELECTRIC
CRYSTAL
NEEDLE
CHUCK
f
i
REAR
SUPPORT
SHOCK
ABSORBER
CONNECTION
BLOCK
TORQUE
LEVER
5:1 TORQUE
LEVER RATIO
CRYSTAL
INSTALLED
IN LEVER
PIN -TIP
TERMINALS
GROUNDING
PIN JACK
PIN JACK
125
STYLUS
PLATED CERAMIC
CRYSTAL UNIT
ANCHORING
RUBBER
DEVICE
REMOVED
DAMPING
REPLACEABLE STYLUS UNIT
CRYSTAL
SUPPORT
SOFT PLASTIC
DAMPING
SCREW
BLOCKS
PLASTIC
CASE
NEEDLE
TERMINAL
TO ONE SIDE
OF CERAMIC
CRYSTAL
NEEDLE
HOLDER
LEVER
STYLUS
CONTACT
POINT
STYLUS
TURNOVER UNIT
78
STYLUS
-
SOFT PLASTIC
CHUCK
STYLUS
SWITCHING
HANDLE
Above: A modern ceramic cartridge in crosssection. Right: Conventional crystal cartridges of
different styles
33
STYLUS
l''''''':.21
` ARMATURE
CRYSTAL SUPPORT
TURNOVER KNOB
point of resonance in the high -frequency end of the pickup's response, an
effective rise in the otherwise drooping high end can be corrected by proper
mounting of a unit within its case.
The earlier crystal units were protected from moisture by black asphalt
coatings, but little could be done against harmful rises in temperature except
to warn the owner. Now, with new crystal materials such as ammonium dihydrogen phosphate and others, heat and temperature are not such important
factors. Total response has been raised in crystal units, while distortion and
weight have been reduced. Crystal and ceramic pickups still provide high
fidelity at moderate prices.
THE CERAMIC PICKUP CARTRIDGE: A piezoelectric effect occurs in
ceramic cartridges as it does in crystals. Because of unusual manufacturing
techniques, ceramic cartridges are in some ways superior to magnetic cartridges.
Composed of barium and calcium titanate, the material for a unit is first mixed
in a watery solution and allowed to dry in a mold, much as a small porcelain
object would be formed. After it becomes hard and dry at normal temperatures, it is exposed to intense baking heat. The ceramic casting is allowed to
cool while under the influence of a very strong electrical field. After this
process, the unit will exhibit piezoelectric effects when bent or twisted. The
actual structure of the cartridge unit is very similar to the bimorph Rochelle
salt crystal in that it usually consists of two thin slabs of ceramic material separated by a metal contact plate. The other contact is taken from the open sur -
Ceramic stereo cartridge
by Electro -Voice. Note
the built-in finger lift
Equipment fur evaluation furnished by Electro-Volce. Inc.
126
faces. Since ceramic cartridges are not affected by humidity and temperature,
no special protective coating need be applied.
Placement and mounting are very important to the correct operation of
ceramic cartridges. They have relatively smooth, distortion -free response and
high output signal level over a range from 50 to 10,000 cycles when unequalized.
As does the modern crystal unit, the ceramic cartridge will provide high-fidelity
operation at minimum cost, and has thus become almost universal in low-priced
phonograph units. Some manufacturers have perfected ceramic cartridges to
such an extent that they can compete on an even basis with conventional magnetic pickups. Equalization networks are available that can be used in conjunction with a ceramic unit when it is to be plugged into an amplifier's magnetic cartridge input.
THE MAGNETIC PICKUP CARTRIDGE: There are three electromagnetic
pickup types: moving armature in a magnetic field, moving coil in a magnetic
field, and moving magnet in a stationary coil. The earliest was the moving
armature in a magnetic field, where magnetic power was supplied by an oldstyle horseshoe magnet of limited power. The whole unit weighed more than
31/z ounces, or over 100 grams. Compared with late -model cartridge -and -arm
combinations, some of which track at just one gram, this old unit could not
do much more than wear out shellac records. In tracking a groove the stylusarmature combination moved in the center of the coil, producing a small signal
voltage, which was amplified into audible sound.
Modern versions of this moving armature, or variable -reluctance, cartridge
have become very popular with high-fidelity enthusiasts for their low tracking
force, high compliance, and good frequency response. The major dawback
to most magnetic units is that their low output signal requires additional
Right: A cartridge operating on the moving magnet principle. The armature, M, rotates
in bearings R, P within the mu -metal core, J.
Below: A ceramic stereophonic cartridge by
Electro -Voice
127
stages of amplification. The earlier high-fidelity magnetic units were subject
to hum pickup and magnetic attraction to iron or steel turntables. Developments both in electronic circuits and in pickup cartridges have just about
negated these problems in modern high-fidelity systems.
THE MOVING -COIL MAGNETIC PICKUP: The second type of electromagnetic pickup cartridge is the moving-coil device. A magnetic field is set up by
a strong magnet, and the coil is forced to move through the lines of force
created by the field. In the process, a small signal voltage is produced in the
coil and fed into an amplifier. There are several methods of introducing such
a coil into the field. In one case, a coil of microscopic wire is wound along the
long axis of a stylus and armature and centered between two magnetic poles.
In another case coils already centered in the field area are actuated by a
stylus -lever assembly. The moving -coil pickup has exceptionally good low frequency response with low distortion. As in all magnetic pickups, the
moving-coil unit is subject to hum pickup from a turntable motor or through
the high -ratio transformer used with this type of low -output -voltage device.
Special mu -metal shielding has all but eliminated this problem in present-day
units.
MOVING MAGNET PICKUP: The third and newest type of magnetic pickup
employs a moving magnet in a stationary coil. With high -permeability shielding
and special hum -bucking coils, there is no problem with annoying hum signals.
Since the effective mass of the stylus magnet is so much lower that that of the
other magnetic types, tracking force is lowered to about 1 gram, permitting
this type of cartridge and arm to be used safely to play master discs, from
which molds are made for LP recordings. In laboratory tests conducted on
this type of magnetic pickup, the authors found a new arm and cartridge
combination called the Dynetic, manufactured by Shure Brothers of Evanston,
Illinois, to be a standard by which most magnetic devices of this nature might
well be judged.
Working on the moving magnet principle, the Dynetic unit has friction -free
suspension through a system of jeweled pivots and thrust bearings. It has
a unique system of stylus and cartridge retraction through a remote pushbutton
on the arm. The tone arm itself is counterbalanced on a damped suspension
The ESL P60-1 cartridge and 310 tone arm
Equipment for evaluation furnished by Electro Sonic Laboratories
128
TERMINALS
MAGNET
m
g
UPPER BEARING
COIL-WIRE
f/< THICKNESS
OF HUMAN HAIR
_é
LU YYCK
BEARING
SHOE
t
IIIrOLE
PIECE
STYLUS
moving -coil magnetic cartridge. Above:
Components identified. Right: A typical cartridge, the ESL C60
The
Equipment for evaluation (Urn iohctl by
Electru Some I.abnra twies
bar, which provides critical damping even at subsonic frequencies without
impeding the lateral movement of the arm. The arm is so constructed that
it has no vertical movement, swinging only in a lateral direction across the
recording. Cartridge and stylus clearance and pressure are provided by a
hinged joint between the arm and cartridge.
SPECIAL PICKUP CARTRIDGES: In science and industry there are many
devices that will produce signal voltages which correspond to a variety of
applied external influences. Most of these devices have been tried as signal producing elements in phonograph cartridges. Of this group there have been
few that could compete commercially with piezoelectric or magnetic units.
The most successful of these is the strain -sensitive pickup.
THE STRAIN -SENSITIVE PICKUP: This pickup cartridge is based upon
the principle that the resistance of some conductors will change as the conductor is twisted or strained. If direct current is passed through such an element as its resistance is changed according to the movements of a stylus, a
signal voltage can be introduced into a vacuum-tube amplifier. Strain -sensitive elements have been used with great success in industrial processes where
the mechanical motion is greater than ordinary stylus motion and where the
additional voltage supply to the pickup element and associated amplifiers is not
a problem. They have, however, not gained much success on the high-fidelity
scene.
THE CAPACITANCE PICKUP: Movement of a stylus against capacitive
elements in this type of pickup can cause a slight variation in the total capacity
of the device. It is possible to obtain a very small signal from this motion and,
through many stages of amplification, to produce an audible signal from a
high-fidelity loudspeaker. It is more practical to connect this varying capacity
to an oscillator, causing it to be frequency modulated, and later to be detected
into a usable signal.
THE STEREOPHONIC SINGLE -GROOVE PICKUP CARTRIDGE: The
dual track magnetic recorder developed after World War II gave the start
to practical stereophonic sound on tape. However, the first stereophonic recording on disc records employed a system devised by Emory Cook, requiring
two separate bands of recording on the disc. In playback, two separate pickup
cartridges, spaced apart on the end of a bifurcated arm, were required.
Problems arose in the fact that no two discs could contain these simultaneously
129
JEWEL THRUST BEARING
COUNTER WEBWIT
MOUNTING SPRING
DYNAMIC DAMPING
LIFT BUTTON
MOUNTING SCREWS
BUTTON SPRING
JEWEL BEARINGS
DYNE TIC CARTRIDGE
IFwlui
OFF -SET NEEDLE
IInlPllt
fir
evaluation IUrIIIYhe
I
by Shure Bros.,
Inc.
Shure Brothers' Studio Dynetic tone arm and cartridge assembly
recorded bands of grooves at exactly same distance apart; consequently, much
adjustment was necessary to keep the stereophonic effect. The amount of
material recorded on a disc was cut in half, thereby raising the cost per
minute of music to twice that of the average monaural LP.
It would not be fair to say that stereophonic disc records originated recently; the idea goes back to the turn of the century, when the disc record
itself was a new idea. Dating back to 1904 there are nearly 400 patents pertaining to stereophonic sound recording and reproduction. Some of these concern multi -channel devices and some single -channel methods. Needless to
say, these inventions were left by the wayside until the present day, when
stereophonic disc recording is a reality.
Several single -groove stereophonic disc recording methods have recently
been under consideration by the industry. One is a combination of lateral
and vertical recording in the same groove; one channel is supplied with the
conventional lateral cut and the other channel with the vertical, or hill -and dale cut, as used in Edison's machines. Obvious problems exist with this type
TONE ARM
Right: An early stereophonic disc record system
employing two pickup cartridges and a dual channel disc. Below: Operational diagram of a
moving magnet cartridge
TO AMPLIFIER
COILS
METAL CORE
STYLUS AND MAGNET
130
BAND 2
CHANNEL 2
BAND 1
CHANNEL
PICKUP
HEAD 2
1
PICKUP
HEAD 1
2D
.
2D
i
.
s
t
0
i
i
.K' ,
\
i'
i
Á
i
i'ri
Á
\ ,'
.
45-45
LATERAL
MAXIMUM
GROOVE
i
\v' \W,
ViÁ
i
VT
STEREO
EXCURSION
20+2d
2D+2Á
RELATIVE OUTPUT
PER
CHANNEL
A
g
RELAT: VE DB PER CHANNEL
0
-3.0
Stylus movement in standard lateral and 45-45 stereophonic pickups
of system, since modern amplifying equipment would bring to the fore all
the unwanted sound produced by most turntables and record changers. It is
entirely possible that the vertically recorded portions of the grooves would
wear out quickly, causing a signal loss in one channel and hence a loss of
stereo effect. Dirt, imperfections, and other plaguing conditions would mar
the general over-all worth of stereophonic sound. The authors do not intend
to say that this system is impractical and cannot be worked out successfully,
but there is not sufficient information available at this time to make any sort
of judgment.
Microphotographs of stereophonic record grooves cut with Westrex 45-45 equipment
CHANNEL 2
MICROPHONE
/
STEREO
DISC
SYSTEM
PROCESS
TWO-CHANNEL
TAPE RECORDER
IDENTICAL
LOUDSPEAKER
SYSTEMS
1
CHANNEL
MICROPHONE
TWO -CHANNEL
TWO -CHANNEL
STEREO DISC
CUTTER
STEREO DISC
PLAYER
1
ORCHESTRA
STANDARD
-DIRECT SOUND
y
=DISTANT
DISC
PROCESSING
SYSTEM
&
REVERBERANT SOUND
RECORDING SYSTEM
IDENTICAL
POWER
AMPLIFIERS
PLAYBACK SYSTEM
Block diagram of the stereophonic disc recording and playback system
Another system, one that has recently gained industry acceptance, is the
system designed and perfected by the Westrex Corporation. Manufacturers
of pickup cartridges have produced both ceramic and magnetic single -stylus
cartridges for stereophonic playback using this system. The Westrex system
employs 45° movement of the stylus in one direction for channel 1 and 45°
movement in the other direction, at right angles to the first channel, for
channel 2. As a single stylus moves through the grooves of a stereophonic
disc, it is impulsed to the right or left at a 45° angle from the vertical, giving
a composite vertical and horizontal movement to the signal -making elements.
This system is a combination of both lateral and vertical recording, but
overcomes the drawback, experienced in all vertical recording, of distortion
due to the recording cutter having to remove different amounts of material
from a disc. In lateral recording, the cutting stylus removes essentially the
same amount of material regardless of the intensity and frequency of the cut.
With the new feedback cutters now used in master cutting, there is no reason
why the 45/45 Westrex disc should have any less quality than a standard LP
recording. The pickup heads themselves can employ two ceramic elements, two
coils, or even two moving magnets to gain the two signals necessary for dual channel stereophonic operation.
TONE ARMS: It is difficult to separate the interaction between a tone arm
and a pickup cartridge. Each has a specific and important, but not mysterious,
job to do. If one were to read and believe advertisements, each different manufacturer would have his tone arm and cartridge accomplishing a different
Left: a magnetic stereo cartridge. Right: two
Equipment. for . ::luation furnished In
ii,i
In.
laic.
.
132
coil assemblies distinguish the stereo cartridge
,--9-1-
i
1
1I1Ill1111
DIAGONAL RECORDING
LEFT CHANNEL
1
L
11 11
I
I
1
11
DIAGONAL RECORDING
RIGHT CHANNEL
J
DIAGONAL RECORDING
CHANNELS IN PHASE
(EQUIVALENT TO VERTICAL)
I
DIAGONAL RECORDING
CHANNELS OUT OF PHASE
(EQUIVALENT TO LATERAL)
---
MIL
\v
í
4
-al
1*-
I
MIL
The Westrex 45-45 disc recording principle. Each of the two channels is cut diagonally. Equal
signals in phase result in vertical stylus movement. Equal signals out of phase result in horizontal
stylus movement
The Westrex
model 3A
StereoDisk cutting head
in operation, producing
a master disc
A closeup view of the
Westrex 45-45 cutting
head for stereophonic
disc masters. The cutting
stylus is actuated by the
angled connecting rods
near the tip of the stylus
lever
purpose. We have defined the job of the pickup cartridge, and now let's look
at the work cut out for the tone arm.
The tone arm (a term dating from the acoustical phonograph) is a moving
device holding the sensing element called the cartridge. It does not have any
other job but to carry this cartridge unit over a disc record as the disc revolves. The facility with which any tone arm accomplishes the job establishes
its difference in quality and price from any other tone arm. To define the
requirements more exactly:
1. A tone arm must move freely over the horizontal surface of a disc
recording.
2. A tone arm must allow a stylus to be easily set down on a record and
removed.
3. A tone arm must be so designed that it comes as close as possible to
accurate tangential tracking of each successive groove as it moves toward
the center of the turntable.
Some tone arm kits are
available. This high quality kit requires only
a small screwdriver and
a minimum of manual
skill
Equipment for ct'aluation furnished by Audax. Inc.
134
Wrapped up in the requirements for the perfect arm are, as always, the
problems that inhibit perfection. Any material object has a resonant frequency
at which it can vibrate if set in motion by some force. This resonant frequency
is the point at which it will vibrate more easily than at any other frequency.
This resonant point is governed largely by size, mass, and configuration. The
functions of a tone arm involve audible frequencies. Unfortunately, most tone
arms are of such a size that their resonant frequency points are somewhere
below 100 cycles. If an arm is set into resonance by motor rumble or vibration
of the rotating mechanism of a turntable, this vibration is easily transmitted
to the cartridge and stylus element and translated into audible sound. Both
vertical and horizontal elements of vibration can cause tone arm resonance to
occur. By damping the lateral and vertical motion of the arm, the mechanical
coupling between the arm and its mounting can be eliminated. A series of
high-grade pivots or viscous -controlled joints will aid in this. However, these
elements of damping also resist the force of the groove walls which move the
stylus from groove to groove, and faulty tracking or disc damage can occur.
If the record is not centered, or is slightly warped, the arm will not stay in
the groove. If arm resonance is unchecked, the stylus may have a tendency
to jump the groove through the force of a high -amplitude signal recorded at
the resonant frequency of the tone arm. The choice of construction materials
and methods of design should be such that the arm resonant frequency, undamped, is below the low -frequency limit expected to be reproduced. Various
techniques have been employed to damp tone arms through added weight,
interior packing of the arm channel with a viscous plastic material, or solid
construction of lightweight wood or plastic.
Designing a tone arm with the facility for getting the stylus into the first
groove and removing it at the end of the disc involves an internal lift lever
where the arm has no vertical motion, or simply an external finger lift on the
cartridge element. Other systems may employ a jointed arm, hinged either
Right: A completed tone
arm, as constructed from
a kit. Below: The head lifting device on the
Shure Studio Dynetic
tone arm
Equipment for evaluation furnished by Andas, Inc.
HEAD -LIFTING
PLUNGER KNOB
PIVOT POINT
Equipment for evaluation furnished by Shure Bros., Inc.
135
Ei
iugnnent for evaluation furnished by British Industries. Inc.
Modern tone arms have adjustments allowing them to be fitted to a variety of turntables
at the head or at the rear pivot. There is no real advantage in any particular
system, providing the pivot joints are secure and non-vibratory. Mechanically
operated arm systems do not usually maintain their hold on the arm once it
is in contact with the first groove of a record; they are simply an added
convenience.
One of the most difficult problems with any pivoted tone arm is that of
tracking the grooves of a disc at the right angle. The disc might be considered
to be a series of many hundreds of concentric circles, decreasing in diameter
toward the center. As a pivoted tone arm moves in toward the center, the
angle between the axis of the arm and the tangent to the groove must change
constantly. Most arms are of fixed length; they are expected to play all sizes
of discs equally well. Since the arm is set in a mounting board at one spot,
Refer to the text to find the proper position for straight, left, and angled, right, tone arms
ANGLE
ANGLE OF TANGENT
TO THE GROOVE
OF HEAD
AXIS OF
TONE ARM
ANGLE OF
TANGENT TO
THE GROOVE
STYLUS
POINT
STYLUS
POINT
DISC
RECORD
136
DISC
RECORD
little can be done to eliminate this complex problem. Incorrect tone arm
placement can cause harmonie distortion, stylus "talk," and abrasive side
thrust on the grooves. It is interesting to note that tone arm placement for
minimum tracking error will not produce a condition of minimum distortion,
since tracking error is most critical at the inner grooves. The placement of
any particular pivoted tone arm is at best a compromise; yet, if manufacturers'
recommendations and installation templates are followed, the results will be
highly satisfactory.
For a straight tone arm without an angled head it is best that the stylus
fall short of the center of the record. Follow this formula:
L represents the length of the tone arm as measured from the pivot point
to the tip of the stylus:
For 12" discs
... d
3.18
L
d represents the distance from the center of the disc to the stylus as it
rests over the center pin of the turnt able.
For a tone arm with an offset or angled head, it is best that the stylus
pass beyond or below the center of the disc. In the following formula, d
represents this distance, and L again represents the length of the tone arm
from pivot to stylus point.
For 12" discs
...d=
4.60
L
High-fidelity tone arms of today incorporate several standard adjustments,
including stylus weight adjustments through a counterbalance system and
adjustments for height relationships to various turntables. One arm produced by Garrard is fully adjustable in length from 12 to 16 inches, with a
template to guide the user in selecting the proper length. It provides for
differences in turntable height and stylus pressure, and has a fully adjustable
head for selection of the proper tracking angle in relationship to the length of
the arm, with instructions and a protractor provided for perfect installation.
The TPA -10 Garrard tone arm is designed to suit the needs and requirements
of installations where adjustment is important.
In recent years, designers have tried to overcome the problem of tracking
error by employing an overhead lathe structure that lies above the turntable,
parallel to the radius of the disc. A pickup, mounted on the horizontal travel -
Modern turntable and
tone arm combinations
are supplied with instructions and templates
to insure correct mounting
Equipment for evaluation furnished by British Industries, Ine.
137
A cardboard template
serves as a guide for
positioning this tone arm
assembly correctly on a
Garrard turntable base
Correctly positioned
holes have been drilled
in the turntable base to
accept the tone arm and
mounting hardware
The completed turntable
and tone arm assembly
rests on a vibrationproof
shock -mounted base
Equipment fur evalualim furnished by British ladusu'ies, Inc.
138
Rek-O-Kut Rondine turntable with Audax tone
arm assembly mounted
in correct position
I.luilutut
turul,hed by I<ek-(I-Iul and :\udax
ing rod, is allowed to move freely over the disc without the constantly
changing arm and tangent relationship introduced by the pivoted arm.
THE HIGH-FIDELITY SINGLE DISC TURNTABLE: The first electric
turntables employed standard -speed motors with some sort of geared speed
reducer. In early days, the acoustical and first electronic phonographs had
very little bass response, hence the rumble of metal gears produced no audible
sounds. With the advent of better amplifier electronics, rumble in the drive
system of any turntable was untenable; better drive systems had to be devised.
Any sort of vibration or speed variation transmitted to a cartridge will result in audible disturbances.
The most widely used turntable power system is the rim drive system.
Any type of induction motor can be used; drive power and speed reduction
are accomplished through a series of rubber -tired wheels, one of which is
driven by the motor armature. The former in turn powers another wheel,
which has been placed in contact with the rim of the turntable. This system
has found wide usage because of several features: speed reduction is easily
Heavy
cast -aluminum
turntable and single point bearing combine
with soft rubber drive
wheels to provide constant speed and low
rumble content
Equipment furnished by ltek-O-Kitt
139
Courtesy R.C.A.
Direct drive turntable in use with a professional recording lathe
accomplished; vibration and rumble are reduced through the use of rubbertired drive wheels; a wide price range of standard motors can be used; while
the turntable is driven at the rim, the center hub can be used during change
cycles to power a mechanical disc changer; and the complete mechanism can
be made small enough for general usage. One disadvantage of the rim drive
system is the possibility that "flats" or indentions will be formed on the rubber wheels during periods of idleness as the wheels press against either the
motor shaft or the turntable rim. In some of the better turntables and
changers all rubber wheels are disengaged when not in use.
Direct drive turntables appear mainly in professional equipment where the
added cost of special slow -speed motors is not important. The system of operation involves the placement of the turntable directly on the shaft of a
slow -speed motor. The cost of a motor designed to revolve at 331/t rpm would
far exceed the cost of the rest of the high-fidelity equipment of a non-professional. In reality, most direct -drive systems employ some sort of gear -reduction
mechanism which has in it a series of nylon or teflon gears to inhibit motor
vibration from creeping into the cartridge and stylus unit.
Belt -drive and shockproof mechanically coupled turntables have found
their way into all price groups except the very inexpensive. They work on
a principle of coupling the power of a motor to the center shaft of a turntable
through a flexible shaft or belt. In this way the full torque power of the
motor can be transmitted to the turntable without vibration. The motor can
be placed on its own shock mounting at a considerable distance from the turntable. In some types of belt -drive turntables the belt is connected to a pulley
below the unit; in one case it is run around the outside rim of the turntable
itself. Speed changes are accomplished by shifting the belt to motor pulleys
of different diameters.
Regardless of the turntable power system used, performance is dependent
140
upon sufficient motor torque power to maintain speed with constancy, and low
hum coefficient. Mechanical speed regulation depends upon the quality and
alignment of bearings, and the balance of the armature and all rotating parts.
Garrard of England has for years insisted that each of its turntable armatures
be dynamically balanced. Any rotating device can be balanced through the
use of added weights, as automobile wheels are balanced. Close static balance
can be effected by adding weights according to slow -moving balance. However, effective balance can only be accomplished by correcting eccentricities
and checking the final balance dynamically at the proposed rotating speed of
the armature. A high -quality turntable should be rumble free, with speed
constancy within lA of 1%. High -quality synchronous motors of several types
account for the major differences between so-called budget turntables and the
true high-fidelity units. To counteract slight speed variations in disc recordings, some manufacturers have incorporated stroboscopic speed-checking devices, with a means of varying the speed of the turntable mechanically.
Right:
The
operating
of a gear -drive
turntable with a vibrationproof shaft coupling.
parts
Below: The
three -speed
turntable features stroboscopic speed adjustment
SHOCK-ABSOREH2
CCUPUNGS
Eruipme-t ter evaluati.r f wrished
by H. H. Scott Corp.
141
Armature-balancing device used by Garrard of England to check armatures dynamically
DISC RECORD CHANGERS: The first record changers were manufactured
while acoustical recordings were still being made. At that time, all recordings
were made so that alternate sides were in sequence. The first changers were
given the job of turning the records over. The legendary device of this type
was the oversized Capehart which could, with "human" dexterity, sort through
piles of discs of various sizes and play at least two hours of music. And, like
human beings, it could be temperamental and imperfect. The advent of the
automatic or drop changer allowed a series of records to be handled; all the
listener had to do was to tend the machine once during the course of play to
turn over the stack of discs. During the period of 78 -rpm shellac recordings,
manufacturers began to put out sequential albums intended for changers.
Most early changing devices were mechanically imperfect and sources of
constant trouble, in addition to being very hard on valuable record collections.
Garrard of England brought out the first really practical disc record changer
of high quality. The familiar bent spindle and balanced tone arm with a
rotating head for ease of stylus replacement were all part of this first unit.
Low rumble noise and ease of operation with a minimum of record chipping
and damage were characteristic of this early English changer.
Present-day changers are available in a very wide range of price and
quality. The less expensive devices are often underpowered, with unconstant,
weak shaded -pole motors. Still, they perform the job of changing discs about
as well as the most expensive changers. The drop changer has become standard, with various manufacturers adding features such as variable speed control and intermixing of all sizes of records. New extra -long-playing recording
techniques have brought about the 162/3 -rpm disc, but few manufacturers
have been able to produce, for the average budget, a changer or player that
will play these discs without the drawbacks of rumble and wow. Each year,
progress in changer design eliminates some of these problems. The modern
changer, compared to the changers of 15 years ago, can provide up to 20 hours
of continuous LP programming.
142
$00111KlIlN/lle
Right:
A
stroboscopic
turntable chart, used in
ccnjunction with a 60 cycle lamp (preferably
fluorescent), provides an
accurate means of checking turntable speeds
I
MO -OR SHAFT
Above: The armature of
a Garrard record change-. Holes and rivets on
the fan blades are added for balance. Right:
Small speed adjustments
are effected by shifting
the positions of screws
on turntable mechanism
RUBBER -TIR -D DFI'/E V'HEELt
RFK-O-RUT
e, a.
Ìrrru,
.. 34
SPEC-ADlUSTMEdT SC; VS
35
33',
SPEED -SW"Ti
-G KNOB
1<,a<-::_
The record -handling side of a three -speed
Garrard record changer
THE HIGH-FIDELITY PHONOGRAPH STYLUS: The subject of the correct stylus for playing phonograph recordings has had widespread discussion
in the past, each expert proposing his own special type of needle. Edison, from
the first, proposed that the hardest substance would be the best; present-day
experts agree with his diamond stylus precept. Others in the past have tried
various kinds of metals, woods, and glass, each inventor claming that his
system was the only one that could get everything out of an acoustical recording
without damage to the record. In later years bamboo, cactus spines, plastic,
fingernail parings, and other and odder materials were tried. As better manufacturing techniques came to the fore the choice of stylus resolved itself into
three materials: metal osmium, synthetic sapphires, and diamond points. No
matter how hard the stylus material or how soft the vinyl LP disc, a stylus still
wears out and must be replaced; the materials wear out in the order listed
above. Eventually the diamond, because its range of damage -free playing time
runs from 500 to 2000 hours, will take over from the osmium utility point,
which has virtually no durability, and the sapphire, which has an average life
of 200 hours.
The underside of the record changer pictured above
Equipment for evaluation furnished by British Industries, Inc.
144
A modern
record
four -speed
changer by
Garrard (162h, 33%, 45,
and 78 RPM)
Equipment for evaluation furnished by British Industries, Ins.
The size of disc grooves has, by and large, determined stylus tip size from
the old acoustic groove size of 5 mils to the modern LP of 1 mil or less (1 mil=
lhnoo inch). The leaf spring and bent -shank stylus shapes are intended to add
a measure of vertical compliance to otherwise stiff cartridges.
THE INSTALLATION OF DISC PLAYING EQUIPMENT: It is not advisable to separate units in installing the group of composites making up a
high-fidelity phonograph. There is a special relationship between a stylus, a
cartridge, and a tone arm, and a further relationship of the arm and the turntable mounting base with the turntable in place. Each manufacturer will provide specific instructions for his particular piece of equipment; because he
would like to have this piece of equipment enjoy the widest sale possible, he
will include instructions for its application with most allied devices. Follow
these instructions to the letter.
Various styles of old phonograph needles. Left, metallic; right, fibrous
145
liiili
üiüiilill
E
uipment for evaluation furnished by Shure Bros.. Inc.
Above: A moving -magnet monaural cartridge.
Left: The stylus and magnet assembly used in
the above cartridge
Basic rules must always apply; they are not always emphasized by manufacturers' instructions. They are:
1. Try to install your disc -playing equipment away from the vibratory influence of your loudspeaker system.
2. Try to install your disc -playing equipment away from the hum -producing
electric fields created by power transformers and a.c. power lines.
3. Pick a spot that is convenient for ease in handling disc equipment. Most
people prefer to have it at table height for accessibility to all but small
children.
4. Make sure the tone arm is positioned correctly in relationship to the
turntable; it must not touch or drag on the cabinet sides during the inner
portion of its swing.
5. It is wise to check both signal wire and power connections against badly
soldered or protected joints.
6. The mounting board and turntable surface should be as nearly level as
possible. A small spirit level can be used to position the unit correctly.
7. After the turntable or changer has been installed, check all spring -loaded
shock mounts and mounting screws to make sure that the turntable or
changer base is actually floating freely and not bolted or fastened securely
to the mounting board at any point.
MAINTENANCE FOR DISC -PLAYING EQUIPMENT: Maintenance for
disc -playing equipment comes under the category of preventive maintenance,
but it is quite important. The following simple steps can be performed by anyone interested in keeping his disc -playing equipment in top shape:
Stylus and cartridge assembly
a. Keep a time chart on the use of the stylus and replace it when necessary.
b. Keep the stylus clean and free from dust and dirt where it enters the
cartridge element.
c. Check to see that the screws holding the cartridge in place are secure.
d. Check all leads for secure fastening to the terminals of the cartridge.
Tone arm assembly
a. Keep all pivots free of dirt and gumming materials that might inhibit the
normal movement of the tone arm.
b. If called for by the manufacturer, lightly oil the main vertical pivct
tone arm.
146
Eiluipment for evalunliuu furnished by Ilek-ll-liul
Above: Lubrication points on a Rek-O-Kut turntable motor. Right: Garrard turntable spindle
has a grease fitting
Equipment for evaluation furnis:ie,l by
British Industries, Inc.
c. Make sure all construction bolts and
hardware are securely fastened as
designed.
d. Check for proper adjustment of tracking weight in the tone arm.
e. Be sure that lead wires are not pulled too tightly through the arm and
mounting hole; the normal swing of the arm must not be hindered.
Turntable equipment:
a. Check for level turntable and mounting base.
b. All rubber -tired drive wheels and the inner rim of the turntable itself
should be cleaned occasionally with isopropyl alcohol (denatured). They
should be free of any oil or grease.
c. Do not oil or grease the turntable mechanism except where the manufacturer has made provision for such maintenance. Too much oiling can be
more serious than too infrequent oiling.
d. Clean the air vents on the drive motor with a small brush to get rid of
lint and dust, which might cause overheating of the motor.
e. Check periodically for obstructions that might become wedged under the
spring -mounted turntable base and contribute to added mechanical vibration.
f. In disc -changing equipment, the turntable is usually rim -driven; a cam
and gear arrangement near the center of the turntable powers the changer
during recycling. Often, in less expensive units, the changer will seem to
lack power during the changing process; this can be accounted for by either
dehydrated rubber drive wheels (which should be replaced) or by a gummed up vertical turntable center bearing. In the latter case, cleaning with isopropyl alcohol and reoiling will ease the situation.
147
s
TUNERS
THE HIGH FIDELITY AM AND/OR FM TUNER: Tuners are more temperamental than any other piece of hi-fi equipment. While there are several
combinations of tuners, AM -FM, TV, and those for short wave, the separate
AM and FM sections only will be covered here.
AM (AMPLITUDE MODULATION) BROADCASTING: Every radio station is assigned a basic carrier frequency of operation within the AM radio
spectrum. Some large stations are given clear channels, with no other station
sharing the same point on the dial. Others, far distant from each other, are
given the same frequency, but allowed to operate only during daylight hours
so that they will not interfere with each other at night, when radio waves
travel farther.
The Fisher model 500 AM -FM tuner -amplifier needs only a speaker to function as a complete unit
e
TNFoFiSNEt
jr
44,grx>'et'e,6
a
e
Equipment for evaluation furnished by Fisher Radio Corp
148
I,It'IIi._IIc.(I
I
Complexity of Fisher model 500 AM -FM tuner -amplifier
0
is
I::I.Ii!, Corp.
indicated in top and bottom views
A station dial number such as 670 means that the frequency of the electromagnetic carrier is 670,000 cps. The frequency of this radiation is so high
that we cannot see, hear, or feel its vibrations. The only way we can know it
is there is to detect it electronically, with a radio set. Such a carrier frequency is similar to perceptible vibrations, but its rate of oscillation is higher.
The frequency range of the broadcast band runs from 550 kc (i.e., 550,000 cps)
to 1650 kc. A carrier signal must be modulated, just as we modulate the basic
frequency from our vocal cords, if it is to produce sound. Modulation of steady
vibrations is the key to any form of communication. With a human's movements of mouth, lips, and tongue, recognizable sounds are formed. As a carrier
is modulated-that is, changed or varied by the program material-the radio set
sorts these variations from the carrier wave, and the radio "plays." The term
amplitude modulation (AM) simply denotes the method of modulation. In this
case, as opposed to FM, or frequency modulation, the carrier is changed in
amplitude according to the characteristics of the program being impressed upon
it. This is the reason any type of electrical interference can affect AM -broadcast sound. Electromagnetic radiation is susceptible to electrical influence, and
in AM radio there is virtually no way of eliminating its effects. FM accomplishes
this quite easily.
FM (FREQUENCY MODULATION) BROADCASTING: In FM broadcasting, a transmitter sends out a carrier signal similar to that of an AM transmitter, except that its basic frequency is much higher in the radio spectrum.
The allotment of wavelengths to stations is governed by the F.C.C., just as in
the case of AM radio; these lie between 88 mc and 108 mc (i.e., 88,000,000 to
108,000,000 cps). Basic carrier waves are transmitted much like those of any
AM radio station, the difference lying in the mode of modulation.
In AM radio, the carrier stays at one specific frequency; in FM, the basic
carrier frequency is shifted back and forth, higher and lower than the allotted
channel. For example: the frequency of a carrier is 100 mc; this would be
100,000,000 cycles per second. In the process of frequency modulation this basic
carrier frequency deviates from its normal position according to the character
of the modulation signal. In the case of a high-fidelity FM station, this deviation will be 75,000 cycles on either side of the allotted carrier slot. In the course
of operation the carrier will operate in a range from 99,925,000 cycles to
100,075,000 cycles.
The detection of carrier variations is accomplished in the discriminator
section of a receiver and just as in the case of an AM receiver the modulating
signal is removed from the carrier; the latter is then rejected and not used.
149
TUNING FORK
radio, the amplitude of a carrier signal
of constant frequency is
altered by the program
material received by a
listener. The high -frequency carrier signal is
filtered out in the 2nd
detector section of the
listener's radio
In AM
a
A. M. RADIO
TRANSMITTER
0
á
f
Pa
A. M. TUNER
SIGNAL
MODULATED CARRIER WAVE
The audible portion of the transmission remains as the high-fidelity, noisefree FM program. Any static or noise impressed upon the FM carrier has-been
clipped off by limiters in the receiver.
HIGH FIDELITY TUNERS IN GENERAL: Whether the tuner you have in
mind is a single unit designed to receive just AM or just FM, whether it combines the two or has incorporated preamplifiers for phonograph or tape recorder, or if it has its own self-contained power amplifier, there are certain
requirements it must meet or you will not get the proper service from it.
Each tuner is designed to meet a specific need dictated by the amount and
type of equipment already in the high-fidelity system.
REQUIREMENTS FOR A HIGH-FIDELITY AM TUNER: Several sections
make up the modern radio tuner; we will undertake to describe simply what
takes place in each section rather than go into all the variations of systems
and components.
STATIC EFFECT
OF A. M. CARRIER
411111111L
150
A
di
Static alters the amplitude of an AM carrier
signal, causing distortion in the program material received by a
listener
In FM radio, the frequency of a carrier signal
of constant amplitude is
altered by the program
material received by a
listener. The variable frequency carrier signal
is filtered out in t he
listener's radio
MODULATING SIGNAL
CARRIER
r
t
The antenna is essential for the reception of radio frequency energy from
the air. In the early days of radio, the antenna was one of the most important
parts of a tuner since the power of radio stations was limited. There were
fewer stations, and distance reception was desirable, but the sensitivity of receivers themselves was limited. Today every major city has several 50,000 watt stations and many of lesser power. Seldom, however, does a radio listener
try for distance reception because network coverage in his own area is probably
satisfactory. Present-day receivers are much more powerful and selective.
These factors have limited the need for old-style outside antennas; now they
are almost always contained within the set.
In most high -quality tuners, there will be one or two stages of amplification
for the basic carrier wave before the essential conversions are made. The
radio frequency (called rf) section affords coupling between the antenna and
the rf-amplifying vacuum tubes. The chief purpose of this section is to provide
the tuner with selectivity of desired signals on the radio band over undesired
signals. It must provide for constant transfer of the right amount of energy
from each signal over the entire frequency band. It must accomplish perfect
tracking of the tuned input (radio frequency amplifier) section, and of other
adjustable circuits in the tuner. Tuning the station -finder dial on a radio
EFFECTS OF STATIC
ON AN
FM CARRIER LATER
REMOVED IN A RECEIVER
Static
affects only
4-7
the
amplitude of a carrier
signal. Since frequency
relationships remain the
same, FM program material is not distorted by
static
AMPLITUDE CLIPPING
LEVEL ACCOMPLISHED
IN A LIMITED SECTION
OF THE RECEIVER
J
J
151
,,
IA
}
iº i
1r;i
1,1,
1?d
Courtesy Fisher Radio Corp.
A precision AM tuner of high quality
is something like turning on a series of water valves in a long pipeline all
at once, so that water flows instantaneously.
After the incoming waves at one particular frequency have been amplified
by means of the tuned-rf section, it is necessary to change the character of
the wave. The next stage in the tuner is really two units, the oscillator and
converter section. If two tuning forks are struck together, the resulting sound
is composed of several notes; one is the difference between the two fundamental notes. For example, if two forks emit 200 cycles and 300 cycles, the
resulting notes would be 100 cycles and 500 cycles. In a radio tuner we have
one frequency coming in from the air; we provide another by means of a
vacuum-tube oscillator circuit. These two frequencies are heterodyned (beat)
together to produce a difference frequency, in this case called the intermediate
frequency. Tuning across the broadcast band changes the incoming radio
frequency, which would change the intermediate frequency relationship if
the oscillator frequency were not changed at the same time. As these two
radio frequencies are put together in a converter, sometimes called the first
detector, they produce a constant intermediate frequency of 455 kilocycles,
which still contains all the modulation information but which being constant,
can now be more easily handled. Separate oscillator and converter sections
can give a tuner a low noise level.
The course of a compound AM radio signal through a receiver
ANTENNA
hA
rn
R. F.
i
STAGE
I. F.
q
qi
MIXER
STAGE
.
ir
.
1st I. F.
STAGE
LOCAL
OSCILLATOR
152
AUDIO
LOWER FREQUENCY
R. F.
SIGNALS
SIGNALS
SIGNALS
.
,
.
.
2nd
I. F.
STAGE
\
SIGNAL TO
POWER
AMPLIFIER
.
2nd
1st AUDIO
DETECTOR
AMP.
CATHODE
FOLLOWER
A.F.C.
88-
88mc
GRD.
GRID
F.
R. F.
M.
MIXER
AMP.
10.7
10.7
108
108
FILTER
me
I. F.
I. F-
me
1st
LIMITER
me
AMP.
10.7
10.7
10.7
me
AMP.
2nd
me
LIMITER
DISCRIM-
INATOR
88108
mc
DE -EMPHASIS
F
M.
A. M.
A. M.
535-
DET._,
8 A.VC
1620 kc
REACT.
1.ANT.
MOD.
10 kc
TRAP
455 kc
AUDIO
AMP.
OSC.
PHONO
PRE -AMP
PH.
BASS
AUDI
AMP.
COMP
TV
POWER
SUPPLY
5351620 kc
A. M.
MIXER
TREB.
TONE
BASS
TONE
455 kc
REL
XTAL
PH.
INPUT
AUDIO
DET.
OUTPUT
Block diagram of a modern AM -FM tuner with preamplifer, tone controls, and selector switch
The intermediate frequency section of a tuner consists primarily of vacuumtube amplifiers and coupling networks between the converter section and the
2nd detector. The purpose of these IF stages is to provide a maximum amount
of signal gain before the modulation information is detected from the radio
frequency carrier. It is possible to have a large amplification factor in the IF
section because there need not be any provision for amplifying any frequency
but the 455 kc intermediate frequency.
After sufficient gain has been realized in the IF section it is time to separate
the radio frequency and the modulation signal frequencies. The 2nd detector
section accomplishes this through diode rectification, passing the radio frequencies to ground and the modulation signal to the first audio amplifier. In
the case of a high fidelity tuner, the signal from the first amplifier tube is fed
from a special circuit arrangement, called a cathode follower, to a remote
power amplifier. The cathode follower is simply a transfer arrangement which
results in no quality loss in the signal in transit.
One circuit feature common to almost all tuners is automatic volume control (A.V.C.). Wide signal strength variations exist from station to station
across the broadcast band. If it were not for A.V.C., these changes would
necessitate manual resetting of the audio volume control. A.V.C., through
special circuits fed from the 2nd detector back to previous rf stages, maintains
constant carrier voltage at the detector by instantly sensing any great change
in carrier signal, and then altering the amplification factor at these first stages.
The usual kitchen radio or console radio seldom can produce more than
5000 cps in audio response from an AM signal. The bandwidth of a high quality receiver's circuits must be widened to provide a wider audio frequency
spectrum. As a wider audible range is made available to the ear, less amplitude
distortion will be tolerated. As fidelity improves, noise levels are more noticeable, hence, more care must be given to the necessary increase in signal-tonoise ratio. A filter must be employed to remove the 10,000-cycle note that
appears because of the beating together of two adjacent radio channel (station)
carriers. This filter must be very narrow so that it removes only the beat at
10,000 cycles, but allows the audio response to go above that level.
153
I
REQUIREMENTS FOR THE HIGH-FIDELITY FM TUNER: Though different in concept and execution from the AM tuner, the FM tuner must accomplish essentially the same job: producing audible signals from radio waves
in the air. The differences between the two can be shown as replacements in
the block diagram of the AM tuner. Radio waves are picked up in much the
same manner as in any radio and sent through an RF amplier; the system of
oscillator and converter and intermediate frequency amplifiers is similar except all must be designed to accommodate a much wider band width because
of the inherently wider range through which the carrier is modulated.
The first actual changes lie in the substitution of a frequency -modulated
detector, called a discriminator, for the 2nd detector of the AM tuner, and in
the addition of one or two carrier amplitude limiting stages which will remove
any static or amplitude interference signals from the FM carrier. An additional
increase is usually associated with these changes of gain between the signal
from the antenna and the discriminator. Oscillator drift, resulting in stations
going out of tune, is a frequent source of annoyance in FM tuners, particularly
during warmup. Most high -quality tuners compensate for drift by employing
automatic frequency control (A.F.C.) circuits. In many, the A.F.C. feature
may be bypassed for fine tuning on weak station signals.
To try to cover the relative merits of various types of detector circuits,
discriminator circuits, and limiting methods would be pointless for the average
consumer. Too much of the technical element has already crept into advertising literature, where it does no one much good. However, how to judge any
tuner for quality operation is not a question the authors can answer very
easily, either. The tuners pictured in this book have been laboratory tested
and were found to be honestly represented within their price ranges.
THE INSTALLATION OF THE HIGH-FIDELITY TUNER: While a tuner
is not a power -producing device, it has a considerable number of heated vacuum
tubes, along with a power rectifier -and-transformer circuit. All these produce
heat which, if not dissipated, will cause a noticeable shortening of the useful
life of the tuner. No special placement precautions need be taken for the
average tuner unless it has its own power amplifier circuit on the same chassis.
It must, of course, be so situated that good natural ventilation is obtained. It
is not wise to stack two units together on one shelf unless some sort of forced
ventilation can be accomplished; otherwise one unit will tend to overheat the
Right:
Sherwood
FM
tuner underchassis. Below: view of cascade
tuner
CATHeO
FOLLOWER;
OUTPUT
SECTION
LIMITER,
DISCRIMINATOR,
AND I. F. SECTION
POWER SUPPLY
eu pnuai
154
rr
o. ;a u:uo tern .n.a by ..n._r..,...d El_ei.ubIc,
I C.
Sherwood
FM
tuner after undergoing the extensive tests torr ed out on oll eeuipment depicted
other, effectively shortening the operating life of both. The tuner should be
placed where it is accessible to an operator.
FM tuners require external antennas. In areas where almost everyone has
a TV antenna, it is possible to purchase a coupler that allows two or more sets
to operate from the same antenna. The conventional all-channel antenna will
cover FM. channels quite nicely. In remote areas away from strong FM signals
special antennas can be installed to receive maximum signal levels. In major
metropolitan areas, FM signal strength is sufficient that signals from the larger
stations may be received by use of a dipole antenna constructed from 300-ohm
television feed -in line.
Left: Top view of the
Sherwood FM tuner. Below: Scott FM tuner under test
[`,._
Etu
p,:i__e.rt
._...
......._..__.
AMINO.
....
rar iwaluauou fur,..c;hcQ by H. H. Je,rtl Corp.
155
300 -OHM
LEAD-IN TAPE
DIPOLE ANTENNA
PLACE ANTENNA AT
-DIAGRAM
RIGHT ANGLE TO
DESIRED STATION
TO FM
ANTENNA
TERMINALS
PLASTIC
SEPARATING
WEB
27"
27"
TWISTED AND
SOLDERED
CONNECTIONS
COAT ALL SOLDERED
JOINTS WITH NAIL POLISH; DO NOT TAPE
The
construction of a temporary FM dipole antenna
THE HIGH-FIDELITY TUNER KIT: One of the more difficult kits to build,
a tuner presents a real challenge to the novice. The operation of any tuner
depends upon tuned circuits composed of inductances and capacitances. Often,
the position of a single wire lead in relationship to the metal chassis can alter
the tuning of the circuit. Other parts relationships exist which cannot always
be planned for in the construction of any kit. Pre -tuning is not always practical
in the case of such parts as IF transformers, trimming and padding condensers,
etc. Relative values of these parts may change after the parts have been
installed in the circuit. However, the printed or etched circuit boards used
in the better tuner kits eliminate, as much as possible, the necessity for care
in placement of internal wiring. If instructions are followed to the letter,
Printed -circuit wiring board has numbered holes for connections, calls for little experience
156
Above: Parts mounted on printed -circuit board
should be checked before connections are made.
Right: Underside of printed circuit board used
in
Allied tuner kit
chances are that extensive retuning will not be necessary. Care should be
taken by the builder not to adjust or tune any components withnut first having
read the manufacturer's suggestions. Since correct and complete tuning is a
job for laboratory instruments, especially where FM circuits are concerned,
it is wiser not to depend upon tuning by ear. A competent serviceman should
be called in, or the kit should be returned to the manufacturer.
THE STEREOPHONIC AM -FM TUNER: Stereophonic simulcasting on AM
and FM radio has brought some of the tuner manufacturers to market with
receivers equipped to receive two signals simultaneously and separately. In
reality, these units are two separate receivers on one chassis; each can be
-
Top view of Allied Radio
tuner kit using printed-
éircuit board with all
components wired in
place
157
Above:
Stereophonic
AM -FM tuner by H. H.
Scott features independ-
ent
tuning controls.
An
underchassis
view of the Scott stereophonic tuner
Left:
Equipment fur
158
Stereophonic AM -FM tuner with dual tuning controls by Madison -Fielding
tuned independently of the other. The outputs can be sent to separate amplifiers
for the two halves of a stereo broadcast. A switching arrangement for sending
the same AM or FM program to both amplifiers is usually incorporated.
THE MULTIPLEX FM TUNER: One unique feature of FM broadcasting is
the possibility of multiple signal transmission. In the case of multiplexing,
two or more signals can be sent out on the same carrier frequency using the
natural sidebands which occur as part of FM. An FM tuner designed to receive this type of broadcast is essentially two receivers on one chassis, with
either individual or coupled tuning. This system seems the most likely to
succeed as a means for multi -channel stereophonic broadcasting, since it involves only relatively minor changes at the transmitter level. The receiver has
a conventional FM circuit with an additional special circuit for each sideband
program to be received at the same time.
In the communications field, multiplexing already exists as an important
feature of FM. In military communications, multiplex provides a number of
channels with the same antenna and transmitter. In civilian radio, an FM
station may employ its major carrier for broadcast programming and its
multiplex for additional income through some form of police or taxi radio
service.
MAINTENANCE OF THE HIGH FIDELITY TUNER: Little preventive
Right: Fisher tuner has jack for later addition
of Multiplex unit. Below: Separate tuning sections on Madison -Fielding receiver
luiiil*liod
Lp
Fi-Lor
159
Engineer checks output stage of Bogen RB 115 AM -FM tuner -amplifier. Similar tests were performed
on all equipment included here
maintenance is needed for a modern high fidelity tuner, except for dusting
the variable tuning condensers with a small camel's-hair brush to keep dust
from forming between the rotating plates. Apart from tube replacement,
corrective maintenance on either AM or FM tuners is a job for a skilled
technician. Present-day vacuum tubes are carefully designed to last about 1000
hours, but not much more. In some equipment, either by design or by accident,
tubes are driven at excessive plate voltages; this tends to further shorten the
life of a tube. If adequate ventilation is not provided, the life of a vacuum tube
can be measurably shortened through overheating.
In operation, vacuum tubes begin to lose their ability to emit electrons from
the cathode; as this occurs, the performance of the tube suffers. Low emission
in a power rectifier could cause low operating voltages throughout a system,
and thus allow more distortion to creep into the final sound. Low emission in
the power amplifier tubes could cause additional distortion by changes in the
operating characteristics of the tubes at saturation level. In a tuner, low emission can cause changes in operating sensitivity over a wide range. Noise in
tubes due to loose elements can be very disturbing. It may produce a variety
of noises, from the familiar microphonic noises to sizzling sounds. Replacement
is the only cure, but often several tubes may have to be tried before the trouble
is found. Though a certain tube may be used in one circuit, it may not work
at all well in another circuit. When removing tubes for testing, maTk each one
and replace it in its own socket. Do not interchange tubes of the same number;
often a weakened tube in one circuit may not be as important to the operation
of the tuner as it might be in another place. DO NOT ATTEMPT TO REPAIR
OR RETUNE YOUR OWN TUNER UNLESS YOU HAVE BOTH THE TEST
EQUIPMENT AND THE KNOWLEDGE TO USE IT.
160
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7
TAPE RECORDING
TECHNICAL ASPECTS OF MAGNETIC RECORDING: Magnetic recording depends upon the fact that the magnetic characteristics of materials can
be varied. The two most important materials in magnetic recording are the
metal from which the laminations in recording head structures are made and the
metallic oxides used on recording tape. The most important characteristic of
these magnetic materials used in recording is coercivity. Coercivity is the
force of resistance of any material, once magnetized, to changes in its magnetic
state. The metal used in head structures should be capable of being magnetically
changed thousands of times a second without much trouble. The metallic oxide
used on tape should, once magnetized, resist firmly any effort to demagnetize
it without its being sent through the recorder mechanism again. The head
is said to have "low coercivity," and the tape "high coercivity."
The small oxide particles on tape can be influenced by magnetic force
and can become magnetized according to the magnitude of such force. In
magnetic recording the tape, bearing millions of particles, must pass the head
structure of a tape recorder at a constant speed. As this process occurs, different sets of magnetic particles come under the changing magnetic influence
of the head structure, and recording takes place. Since no physical change
The structure of ordinary magnetic recording tape
VARYING LENGTHS; STORED ON
REELS OF
THICKNESSES:
V000"
1/2 OF
TO 34000
DIFFERENT DIAMETERS
---OXIDES
& BINDER
FILM BASE (CELLULOSE
ACETATE, POLYVINYLS, ETC.)
MAGNIFIED PORTION
i
162
CHANNEL ONE
RECORD HEAD
STRUCTURE
MAGNETIC
GAP
DIRECTION OF TAPE
MAGNETIC TAPE
Qá'%
Dóópóa
00g_5
POINT OF
CHANNEL ONE
O
OóOóo
aOU
OD
LIa
'%1A,5:317
\\\>
Ó
C1®
Id
C=
dód
C=
óóO
0J0000 GpOvOÓol0400
oQdDó
CHANNEL
TWO
RECORDED
1
SIGNAL
UNMAGNETIZED
1=1=
=
PARTICLES
PARTLY MAGNETIZED
PARTICLES
= FULLY MAGNETIZED
PARTICLES
The effect of a recording signal on the magnetizable tape particles
takes place on the tape, the resultant recording is constituted in the changing
amount of magnetism at any one point on the tape. More simply, for a loud
recorded sound the magnetism on the tape would be heavy in comparison
with the magnetism for a soft sound.
THE TAPE TRANSPORT MECHANISM: The major job of a tape transport
system is to get tape off the supply reel, past the head structure at a very
even speed, and then wound on the take-up reel at the other side. Variations
in tape speed or tension will produce audible changes in the sound or music
being played from the tape. "Flutter" and "wow" are the terms used to
designate such variations. Complicated instrumentation is necessary to register
the amounts of flutter and wow in a properly working tape device. Less expensive tape machines employ cheap motors, which by their inherent un smoothness of operation contribute to the ultimate inconstancy of operation of
the whole machine. As motor quality is raised and as manufacturers make
more use of smooth-operating hysteresis synchronous motors, the over-all
quality of tape recorders will rise.
As tape from the supply reel is pulled into the head structure of a machine,
it first meets the erase head. The function of the erase head is to remove by
demagnetization any audible signals that might be on the tape. It accomplishes
this by means of a very small high-frequency signal. As the tape passes this
head, the magnetic particles are remagnetized according to the strength and
character of the inaudible a.c. erase signal. For all intents and purposes, the
tape then has no signal on it, and is ready to receive the recording signal. The
163
Above: Ampex 601 monaural tape recorder. Below: Ampex "A" series tape deck with cover removed. Note erase, record, and in -line stereophonic playback heads
164
latter is arranged magnetically upon the tape by the next head in the series,
the record head. Because of the magnetic nature of the recording material on
the tape, a preconditioning signal called bias must be applied to the tape at
the same time as the recording signal. The bias signal is produced in an
electronic circuit of the recorder; it is taken from the same vacuum-tube
oscillator which produces the erase signal. Its essential effect on the signal
being recorded on the tape is in establishing linearity where it would not
otherwise exist. As has been mentioned before, the essential feature of any
audio amplifier is that it simply amplifies the audio signal without otherwise
changing it. This same feature of linearity must also be a part of the recording
amplifier in a magnetic recorder to ensure undistorted output.
The audio signal provided by a microphone and a recording amplifier has
now been magnetically impressed on the recording tape. In most of the betterquality tape recorders a separate playback head is provided for monitoring
the signal on the tape during recording and, of course, for later playback
service. This head, similar to the recording head in structure but with a finer
gap in the pole piece, is influenced by the magnetic energy on the tape. It
produces a small signal voltage in an associated playback head circuit and
amplifier, later to be fed into the power amplifier and speaker system.
Because of the fact that the erase current and record bias are on only
during recording, it is possible to use a single head structure for erase and
record or, alternately, playback. The head has two gaps separated by a short
distance; the first gap provides the erase feature and the second is for record
or playback. During the recording cycle the same signal is supplied to the
erase winding and bias coil on the head structure. The recording signal is
fed to the record winding, energizing the second gap. It is not possible to
monitor the signal on the tape during recording. In playback, no signal is supplied to either the erase or bias coils, and the magnetic impulses on the tape
produce a small signal voltage in what was formerly used as the record coil,
now operating as the playback coil. Used in most non-professional machines,
this type of head structure provides an incomplete but adequate coverage of
the audible frequency range.
Four -channel stereophonic record -playback head with two pole pieces
MU -METAL
SHIELD CONTAINER
HEAD WINDING
CONNECTING
METAL CORE
TERMINALS
165
_
COIL
1
1
r
DOUBLE POLE PIECE
SINGLE POLE PIECE
(-TAPE
DIRECTION OF
OR WIRE
MOVEMENT
FLUX PATH
`SI
II
FLUX
PATHS
SOLENOID COIL
COMBINATION RECORD -ERASE HEAD
SOLENOID
NON-MAGNETIC
IRON
SPACER
A
FLUX PATH
RING
BALANCED RING HEAD
166
_. '
Opposite page: Different types of tape -recorder
heads. Right: Experimental tape magazine unit
incorporated into a stereophonic radio-phono tape console by Ampex Audio
STEREOPHONIC RECORD/PLAYBACK HEAD STRUCTURES: One of
the problems that first beset stereophonic recording and playback for home
sets was the lack of good quality stacked, or "in-line," heads. It was possible to
combine two tape channels in one head, but the cost was more than the cost
of a complete home system. Heads of this quality were reserved for the professional tape machines. Even those who called their machines professional
often used poor heads.
Due to the problem of cross talk (influence between two heads in one
structure), stereophonic sound was first recorded and played back using two
separate heads some distance apart, called "staggered" heads. The signal on
the top channel was magnetically isolated by distance from the channel on the
bottom track, but splicing and editing were almost impossible. However, it
Block diagram of a typical monaural home recorder with stereophonic playback
RECORDING
EQUALIZER
LOUDSPEAKER
TO
CHANNEL
2
AMP
CHANNEL
2 STEREO
PLAYBACK HEAD
RECORDING
MATERIAL}
DRIVE
MECHANISM
167
was cheaper for manufacturers to install a second head in their machines
to provide stereophonic sound playback.
As inexpensive stacked, or in -line, heads became available all manufacturers switched to them, and frequently supplied kits for the conversion of
older sets. Solving the problem of inductive influence between the coils in
each stacked-head section has made it possible to have four channels on one
1/4 -inch tape. By the use of certain combinations of these tracks it is possible
to double the amount of stereophonic material that can be recorded on any
tape. This step makes tape magazines practical in the pre-recorded tape industry. A complete program can be recorded on tape and put into a magazine
either in an endless loop or so that as the tape comes to the end of one pair
of channels another set of channels is switched into operation as the tape
reverses direction. No threading of reels or manipulating of the machine is
necessary to play a tape. The convenience of discs is not quite matched, but
the quality of tape will usually be better.
ASSOCIATED ELECTRONIC COMPONENTS: We have discussed the
transport systems and the head structures of tape recorders. Associated with
these two important elements are the record and playback amplifiers and the
bias oscillator circuit. In a two -channel tape machine designed for both
stereophonic recording and playback, it is necessary to duplicate both the
record and the playback amplifiers, one set for each channel. The bias -oscillator circuit can be so arranged as to supply both recording circuits with erase
and record bias.
The function of the record amplifier is to take a signal from a microphone
or other external signal device and amplify it sufficiently to drive the recording
head structure. This amplifier has essentially the same job as any audio
amplifier. The differences are mainly of a technical nature involving equalization within the amplifier circuit. The unique characteristics of the magnetic
tape call for special care to get the maximum undistorted signal output so
that background noise caused by the tape and the electronic circuitry is at
a minimum. The relationship between background noise and recorded signal
is called the "signal-to-noise ratio," which must be maintained at a maximum
level. It is not possible to cause a constant magnetic influence on the tape for a
constant input current over the audio range. Distortion does not remain
constant for all ranges of reproduction. For instance, a given signal strength
in the low range will have more distortion than the same signal strength at
some middle frequency. In the high end of the range additional distortion
occurs, due to the beating of the signal with the bias frequency. Limitations
imposed by the head structure add to the importance of limiting the amount
of recording current at both low and high ends of the audio response curve.
This is accomplished within the amplifier circuit, and is a fixed adjustment
that can only be changed with laboratory instruments. Technically, this
adjustment is a form of equalization, but it is called pre -emphasis. In this
fashion, the record amplifier and the playback amplifier are bound together in
their operation, since the playback amplifier must make up for the de-emphasis
that takes place during recording so that a flat response in the output is
obtained.
The playback amplifier receives the signal induced in the playback head
by the magnetic impulses on a tape. Its job is to amplify, and, through postemphasis, to correct for the changes made in the record amplifier. In most
professional machines the output of the playback amplifier terminates in either
a line output transformer or a cathode follower and from there connections
can be made to any external loudspeaker and amplifier combination. In
commercial home units an inexpensive power amplifier is included within the
tape machine, making a self-contained unit out of the device. The advantage
of this type of system lies mainly in user convenience rather than in quality
audio output.
168
Above: Pentron stereo
tape deck with record playback amplifiers. This
unit records and plays
back stereophonically.
Right: Rear view of the
Pentron stereo tape deck
showing drive mechanism. Below: Ampex 900
series universal tape recorder
SPEW CHANGE LEVEE
MANUAL
`:HEAD-S61°TING
LEVER
AUTOM
TAPE TENSION
IC
STOP IEJ_'R
DEVICE
CA P'S"
AND
PRf'.7SURE WHEEL
169
Above: A fine, fully portaole tape recorder used
by many professionals,
tie VU Magnemite, made
by Amplifier Corporation
of America. Left: Stereo
amplifier
constructed
f om kit. Below: Heath k t
i70
kit -constructed
corder
re-
TAPE RECORDERS IN KIT FORM: The outstanding tape recorder kit is
sold by the Heath Company. It can be constructed by anyone with a minimum
of technical knowledge. Its electronic and mechanical elements have been so
well designed and explained that it is possible for any beginner to have a high quality tape machine at minimum cost.
THE MAINTENANCE OF TAPE RECORDING EQUIPMENT: Perhaps the
most difficult member of any high-fidelity system to maintain is the tape recorder. Every function of the combined electronic and mechanical units must
be performed correctly in relationship to each of the others. Perhaps as important as anything is preventive maintenance. It is not with regard to the
machine itself that maintenance is stressed here, but with regard to the quality
of the recorded tape. As gradual deterioration takes place the quality of the
tape drops, often so gradually that it is not readily apparent. Through oversight, a valuable master tape may be seriously diminished in quality or lost
completely.
Preventive maintenance consists mainly in following instructions provided
by the manufacturer. However, some less expensive machines do not include
suggestions for lay -maintenance of any type. In these cases, the following
preventive maintenance steps can be followed by the owner:
1. Any part of the machine that is touched by the oxide side of the tape
should be cleaned regularly with grain or isopropyl alcohol. These parts
include tape guides, capstan, capstan roller, and head surfaces. Contrary to
some maintenance instructions, carbon tetrachloride should never be used
anywhere on the tape machine because of its ability to dissolve not only
plastic parts, but the resin that holds the laminated head structures together.
Other types of alcohol, such as the denatured types, often have solvents
such as acetone in them, which would also be bad for tape recorder parts.
2. During the process of recording and playback either the record head or
the playback head may become slightly magnetized. This small amount of
magnetization will add noise to the final tape, may cause part of the signal
to be erased, and will generally lower the signal-to-noise ratio. Demagnetization of head structures is of prime importance. A small demagnetizing
device is sold by several companies, among them Audio Devices, which
manufactures Audiotape. This device will help keep up the over-all quality
of your tape recordings. The head demagnetizer demagnetizes a head by
influencing it with a small 60 -cycle alternating current field in close contact
with the head. Slowly drawing the device away from the head slowly diminishes this field, leaving the head completely demagnetized.
3. Lubrication of the tape mechanism itself is very difficult since there are
specific places that must be oiled periodically, yet all other parts of the device must be absolutely free of oil or grease. Random lubrication can seriously damage or destroy rubber drive wheels, drive belts, and felt clutch
pads. Lubrication must not be attempted without specific instructions from
the manufacturer.
4. Mechanical adjustments for brake tension, tape position, correct wind
and rewind features are jobs for the skilled technician and should not be
attempted by the layman owner.
171
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meth, d f radio construction known iaa ,
a.d soldering on metal chassis, plus the on e w
your rºg,lar AC or DC house current.
'.Printed Cirenitry." These circuits operate
insixteen
THE "EDU-KIT" IS COMPLETE
build 16 different radio and clec
You will receive all parts and instructions necessary tocontain
tubes. tube sockets, v r,
Monies circuits, each guaranteed to operate. Our Kite
and paper dielectric c denser+, resistors, tie strips, coals
aule, electrolytic, mica,
Manuals, hook-up wire, solder, etc
h ardware, tubing, punched ametal chassis, Instruction
Circuit materials, including Printed Circuit chassis
In addition, you receive Printed instructions.
You also receivea useful jet of tools, a.
special tube sockets, hardware and
Dynamic Radio and Electronic!
elf -powered
professional electric soldering iron, and Instructions
and the Progressive Code Oscillator
Tester. The ' Edu-Kit'' also includes Code
for Padio Amateur License training. Tot
Answers
and
Questions
to
F.C.C.-type
addition
Signal Tracera d the Propres
for servicing with the Progressive
will also receive lessonsHigh
You receive Membership it
Fidelity Guide and a Quit ofBook.
e Signal Injector, a
Merit and Discount Privileges
Certificate
Radio-TV Club, Free Consultation Service,
Ygo receive all parts, tools, instructions, etc. Everything is yours to keep.
The Progri ive Radio "Edu-Kit" has been
y thousands of individuals, schools
d tom
.c a rg u,ìzatipns. public and private. through
rid the world. It is recognised internationally
u the ideal radia Course. Progressive Radio
BY cupola. demand, the
'Ede -Kit" is now available in Spanish u well
Ks Enyliºh
It is understood and agreed that beshould
Progressive Radio "Edu-Kit"
he
'uritied to Progressive "Edu-Kits" Inc. for any
eason whatever, the purchase price will be:
efeended
:ion. and
The
in
full. without quibble
high
'Ed -Kits"
(
Ai
delay.
recognition
which
or
goes -
rstomer
u will
learn trouble -shooting
in a propre s
e mangler.
Ill 1C practice
the sets
pairs
an
throughout thee tire
world.
and
You
.
that
nstruet. You will n learn symptoms
of troubles in home, portable
rsradios.
You will learn how to
the professional Signal Truer, the
que Si al Injector and thedynamic
d
6 Electronics Teeter. While you
learning On this practical way. you
Il be able to do many a repair tests
r
and
and
s which a will fu !excee d the pi no él
"Edu-Nit" Our Consultation Server
11 brie you
with any technical prablrma
ning
cae
u
may have.
J. Stated., of 25
ry, Conn., writes
eral sets for
y'
ney. The
é
Eau -Ki
I found yourn ad
Valerie. P. 0. Box 21
'nhe Edu-Kits are monde
answers for them.
have
v
years,
dip for the last seven
work with Radio Kits. an
iId Radio Testing Equipme
yed every m1 nute I worked
efferent kits; the Signal Tra
Also like to lit you kouW
m
a member of your
el ['feud of becoming
adie.TV Club."
Robert L. Shoff. 1534 Monroe Ave..
Huntington. W. Va.: "Thought I would
re.
drop you a few lines to say that
cinved my EduK,t. and was really amazed
that such a bargain can be had t such
a low price.
have already started
pairing radios and ehonOgraphs.
My
lends were really surprised to see me
t into the swing of it so quickly. The
Ben
m sending you the questions
1
1
1
E
ubleshooting Tester that
Kit is really swell,
and
with
finds the
ORDER DIRECT FROM AD-RECEIVE FREE BONUS
RESISTOR AND CONDENSER KITS WORTH $7
"Edu-Kit"
"Edu-Kit"
postpaid. I en:lose full payment of $22.95.
C.O.D. I will lay $22.95 plus postage.
Send me FREE additional information describing "Edu-Kit.r'
Send
Send
Name
Address
Progresºive
Inc. has earned through its many
of service to the public is due to its
nditional insistence upon the mainte lance of perfect engineering, the highest i
stroctional standards, and 100% adherence
oney-Back Guarantee
.O its Unconditional
ave suit, we do net have single dissatisfied
a
ALIGNMENT TOOL
WRENCH SET
VALUABLE DISCOUNT CARD
CERTIFICATE OF MERIT
TESTER INSTRUCTION MANUAL
QUIZZES
HIGH FIDELITY GUIDE
RADIO
TELEVISION BOOK
TROUBLE.SHOOTING BOOK
MEMBERSHIP IN RADIO-TV CLUB:
FCC
CONSULTATION SERVICE
AMATEUR LICENSE TRAINING
PRINTED CIRCUITRY
PROGRESSIVE "EDU-KITS" INC.
PROGRESSIVE BUILDING, DEPT. 5O1AK
1184-86 BROADWAY, HEWLETT, N. Y.
173
Most Highly Recommended
Enclosure in Hi-Fi
for Ultimate Fidelity
Karlson
S H ERW001)
....
PATENTED* ENCLOSURE
As Featured in Popular Mechanics, July '58
thoroughly
engineered and patented musical instrument, used to
perfect the bass response, high frequency dispersion,
tonal definition and efficiency of all full range Hi-Fi
speakers.
The Karlson Enclosure is not just a box, but a
*Pat. No. 2,816,619
--',--The
t
KARLSON
"12"
outstanding honors
For 12" Speakers
Finished Models
12CH, 12MH, 12HB, 12FR
Unfinished -12U ready for
$99.60
finish
Kit -12K
566.00
$42.00
Size: 243/4 x 163/4
Ship. Wt.: 45 lbs.
licited. by most
organizations
..
No matter where your music comes
from-FM, your own discs, or tape
-you will enjoy it at its best corning from Sherwood's complete home
music center
most honored of
them all! Sherwood tuners for ex-
x 133/4
12U
...
"8"
For 8" Speakers
Finished Models
8CH, 8HB, 8FR, 8MH
$42.60
Unfinished 8U ready for finish
same patented structure behind full grill
$26.70
$18.60
Kit -8K
KARLSON
bestowed, unsotestinc
recognized
ample
Recorder-Mate 8-assembled
unit as above with speaker-$10.00 extra.
Size: 171/4 x 113/4 x 93/4. Ship. Wt.: 14 lbs.
KARLSON
"15"
8CH
For 15" Speakers
Finished Models
M+^
15CH, 15HB, 15FR, 15MH
$129.00
$
$
output.
Model S-2000 FM -AM Tuner $139.50 net
Model S-3000 FM (only) Tuner $99.50 net
Unfinished -15U ready for
finish
Kit -15K
...
First to achieve under one microvolt sensitivity for 20 db FM quieting increases station range to over
100 miles. Other important features
include the new "Feather-Ray" tuning eye, automatic frequency control, fly -wheel tuning, output level
control and cathode -follower
87.00
57.00
Aft Prices Audiophile Net, Subject
to Change Without Notice
For complete specifications, write Dept.
S-1
Standard Karlson Finishes
15FR
FR-Fruitwood
HB-Honey Blonde
MH-Cordovan Mahogany
CH-Cherry
Outstanding High Fidelity Experts Saycan put out
Joseph Marshall, RADIO ELECTRONICS
an amazing boss, completely out of proportion to its size
and far beyond the design capacity of the speaker used."
Donald Hoefler, HI-FI MANUAL "A truly beautiful piece
of scientific reasoning, with an ultimate solution which
should please even the most highly critical." "A new
standard of performance."
Lothar Stern, ELECTRONICS MADE EASY "Offers the
utmost in versatility, outstanding bass response and un
usual styling."
"It
SHERW00-0-ELECTRONIC LABORATORIES INC.
4300 N. California Ave., Chicago 18, Illinois
The "complete high fidelity home music center."
V
write for name of nearest dealer
KARLSON
ASSOCIATES, INC.
Dept. SHF9, 433 Hempstead Ave., W. Hempstead,
174
N.
Y.
In New York, hear "Accent on Sound" with Skip Weshne.
W BAI -FM, week nights.9 P.M. In Los Angeles. KRHM-FM. 10 P. M.
FOR ADVANCED THINKING, PRECISION ENGINEERING
AND PURE, CLEAN ACCURATE SOUND REPRODUCTION
Nothing comes close to JBL precision transducers
JBL Signature loudspeakers embody feats of audio
craftsmanship never duplicated. JBL extended range
speakers are made with large voice coils of edge -wound
aluminum ribbon for brilliant highs, crisp lows,
rich mid -range, highest efficiency. Frames are rigid
castings, dural center dome is attached directly to voice
coil for clean sound, free of spurious resonances.
Left to right are the virtuoso 15" Model D130
with 4" voice coil; the versatile 12" b123, 3" voice coil;
the singular 12" 0131, 4" voice coil; precision
miniaturized 8" 0208, 2" voice coil.
Then, above, there are the famed JBL low frequency
drivers, the 130 series with curvilinear cone,
4" voice coil, for 1200 cps crossover; and the fabulous
150.4 series with 4" voice coil and straight-sided
cone for 50) cycle crossover.
acousuci. enciusures are engineered to; JbL
transducers. Pleasingly proportioned, precision built,
impeccably finished, there is one for every taste.
At top, the sensational Ranger -Paragon integrated
stereophonic speaker system. Next, the mighty Hartsfield,
universally acclaimed the finest of monaural speaker
systems, and the extremely popular C40 Harkness
back -loaded folded horn.
Below, tie JBL C37-criterion for reflex enclosures,
and the C39 Harlan corner reflex enclosure of
provocative cesign and extraordinary versatility.
Detailed prints for constructing the C37, C39, and C40
are available at $3.00 a set from the factory.
JBL
Original concapts distinguish JBL high frequency units.
Greatest of all high frequency drivers is the
massive, thirty -pound JBL 375. Of unsurpassed efficiency
and faultless coverage, the 375 is designed for
500 cycle crossover. Used in the Hartsfield and Ranger.
Paragon. Shown here with the 537.509 horn -lens
combination which gives wide horizontal and narrow
vertical coverage to minimize floor and ceiling reflections.
The 181 :75DLH assembly combines precision
driver and complex phasing plug with machined aluminum
exponential horn and the exclusive 1BL acoustical
lens which distributes highs evenly over a 90° solid angle.
The solid JBL 075 is made with the original
ring radiator and annular exponential horn for immaculate
reproduction of highs over 2500 cps.
Write fer free catalog. name of
your la. Dealer, and nformation Dopions
JAMES' B. LANSING SOUP 2, INC.
3249 Casita: Avenue,Los Angelee
175
a.
Calif.
.".
_
..
,`
\\e
iillijliiii
ilpiñ
,/ßieill,
dui:
m.
,
the incomparable
SHURE
PFiQFESk:tONAI.
The Shure Stereo Dynetic Cartridge is designed and built
specifically for the listener who appreciates accuracy and
honesty of sound. It separates disc stereo sound channels
with incisive clarity. It is singularly smooth throughout
the normally audible spectrum . .. and is without equal
in the re-creation of clean lows, brilliant highs, and true to -performance mid -range. Completely compatible
plays monaural* or stereo records. It is manufactured in
limited quantities for the music lover-is available through
responsible high fidelity consultants and dealers at $45.00,
audiophile net, complete with 0.7 mil diamond stereo stylus.
o
1 ,
dii)j.
00,9.--- 7/i
CAR TR IDGE
'.
-
For those who prefer a monaural cartridge with monaural records: Shure
Pro.
tensional Dynetic Cartridge or Studio Dynetic Tone Arm. Studio Dynetic tracks
at one gram. Ill conversion to stereo is desirable later, Shure "Planof-ProteClion" guarantees 75% trade-in allowance until Dec. 31, 1959.)
Literature available:
Dept,
M-1
N C.
SHURE BROTHERS,
Evanston, Illinois
222 Hartrey Street
a note to the technically inclined:
Shure Stereo Dynetic Cartridges are individually tested and
must meet or exceed the following specifications before being
placed on the market:
Frequency response: 2010 15,000 cps
Compliance: 4.0 a 10.6 centimeters Der dyne
Channel separation: More than 20 do trrougl,out the
critical stereo frequency range.
Recommended Tracking Force: 3 to 6 grams.
The M3D tits all 4Iead and 3.lea.f stereo
changers and arms. 10 second stylus
replacement.
I
SHURE ALSO MANUFACTURES NIGH -QUALITY PICKUP ARMS, MICROPHONES, AND
176
,
r,..NETIC
RECORDING HEAO$
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