5-04 - elektor may 1977 colofon/decoder

eLeKrur !mummer
W. van der Horst possible. The decimal point is
Editor : What is a TUN? usually replaced by one of the
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. .

Capacitance value 10n: this is the

.

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contents elektor may 1977 - 5-05

COATE111115 Volume 3
Number 5

A synthesiser, such as the Formant - the Elektor music synthesiser (1) .. 5-14
formant, consists C. Chapman
basically of a simple key- This article is the introduction to a series of articles describing
board backed up by a 'Formant', the Elektor music synthesiser.
As synthesisers are something of a mystery to many people this
boxfull of electronics! article will serve as a general introduction to them, as well as
describing the basic principles of Formant.

from quadro to surround sound 5-19

P. Fellgett and M. Gerzon
In previous articles ('Quadrille' and 'music of the spheres',
Elektor 17, September 1976, p. 9101 it was noted that
'Ambisonics' was perhaps the most promising of the quadro-
phonic systems proposed so far. This article explains the basic
theory and philosophy of Ambisonics.

stereo encoder - A. Bosschaert 5-26

This relatively simple circuit will combine two independent
Block diagram of a audio signals into a multiplexed stereo signal similar to that
used to modulate an FM transmitter for stereo transmissions.
domestic ambisonic
decoder of medium com-
plexity for 2 -channel precision V to f converter 5-29
(45JB) material.
slotless model car track (2) 5-30
This month's article describes the circuit of the multiplex
encoder, and the principle of the decoder. The board layout
for the infra -red transmitter will also be given.

missing link 5-39

op -amp frequency compensation ...

the why and the how 5-40

morse call sign generator - A. Peters 5-41

Gain/frequency response
curve of the music music cleaner 543
cleaner. The ultimate A treble and rumble filter that can be used in virtually any
slope of all filters is hi-fi system to get rid of the snaps, crackles, pops and
(glrumbles without getting rid of half the music.
18 dB/octave.
active loudspeaker -crossover filters (1) 5-46
Few things can so hold the attention of the serious audiophile
as do loudspeakers. This applies with particular strength to
those whose fingers always have the experimenter's itch.
One of the ways of sinking cash into an existing system is to
replace the 'passive' separating ('crossover') filters by 'active'
types.

albar mk I I . . . a b.f.s. 5-52

market 5-55
One of the most essential
parts of any music advertiser's index 5-64
synthesiser is the key-
board. This is what it
looks like from under-
neath ...
5-08 - elektor may 1917 advertisement

121-121' eLmiTor
FORMANT
SYNTHESISER
KITS &
P.G&8
Details in our
lists -send s.a.e.

PLUS:
The Finest KEYBOARDS
The S.K.A."' Plastic Keyboard was developed by Kimber Allen
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keyboards available.
The keys are moulded in Acrylic plastic, a material chosen for its
hard wearing properties and ideal feel to the touch. They are
moulded in two parts, the key face, which has to be perfect u,
111101000
appearance and finish, and the action. which has to be strong and
carry the mechanism. The strong section of aluminium extrusion
upon which they are mounted is specially designed to take all
& CONTACTS
the pressures of playing. Springs, felts, and contact actuators are
supplied ready -fitted.
The contact assemblies are constructed of laminated bakelite,
thus giving smooth slot walls and completely free movement of
the gold -clad contact wires. Types available as follows (Contact
pairs normally open),
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GB -2 pairs: 27p each GH-5 pairs: 57p each Keyboards: £1.50 each E25.50
GC -3 pairs: 36p each 4PS-SPCO & 3 prs: 53p each Contacts: 49 Note DC Keyboard:
Palladium wire Bus Bars - 1 octave lengths: 50p each Orders under £15.00: 25p £32.25
We also stock kits and PCBs for the P.E. Synthesiser, P.E. Joanna Orders over £15.00: 50p 61 Note C -C Keyboard:
(electronic piano), P.E. Min and other sound synthesising £39.75
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advertisement elektor may 1977 - 5-09

PHILTRON
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0001 TRANSVAAL. REPUBLIC OF SOUTH AFRICA

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5-14 - elektor may 1977 formant - the elektor music synthesiser

formant Principles of the Voltage

Controlled Synthesiser.
The concept of the voltage -controlled

the ellelictor synthesiser and related

originated by Robert A. Moog. Any
circuits was

sound can be characterised by just three

muck roynthediger time dependent parameters, namely

pitch, tone colour and volume, or to put
it in electronic terms, fundamental
frequency, harmonic content and
amplitude. If these three parameters can
be precisely controlled for the duration
of a particular sound then that sound
can accurately be synthesised. In
practice this is obviously limited to
fairly simple 'musical' and related
C. Chapman sounds.
A synthesiser thus requires three basic
circuit blocks: oscillators to generate
sounds of the required pitch, filters to
This article is the introduction to a series of produce the required harmonic content,
and amplifiers to obtain the required
articles describing 'Formant', the Elektor music amplitude. Since the three parameters
may vary during the existence of a
synthesiser. This is an instrument of advanced particular sound there must be some
means of rapidly controlling the charac-
specification that bears comparison with many teristics of these circuit blocks, which is
commercially available synthesisers, but at a where the concept of voltage control
comes in. The pitch of a voltage -
fraction of the cost. controlled oscillator (VCO) may be
varied by changing the control voltage
As synthesisers are something of a mystery to applied to it. The cutoff -frequency of a
voltage -controlled filter (VCF) may
many people this article will serve as a general similarly be varied, as may the gain of a
voltage -controlled amplifier (VCA).
introduction to them, as well as describing the
basic principles of Formant. Exponential Voltage Control
The fundamental design parameter of a
synthesiser is the control voltage versus
frequency characteristic of VCO's and
VCF's. In many applications a linear
characteristic is required, i.e. n volts per
A synthesiser may be defined as an on the other hand it may also produce Hz. However, musicians are concerned
electronic musical instrument whose sounds that cannot be produced by any not with linear frequency relationships
tonal characteristics can be varied at will conventional acoustic instrument, and but with musical intervals, the basic one
by the musician. This immediately which can be generated only by of these being the octave. For each
makes the synthesiser different from electronic means. octave increase in pitch the frequency
conventional musical instruments, The synthesiser then, is an extremely of a note doubles. This means that if the
whose tonal characters are fixed by versatile instrument, and it is a great absolute frequency in Hertz is plotted
their physical construction. It also pity that it is often used to provide against the relative frequency in octaves
makes the synthesiser different from an monotonous background accompani- an exponential curve results, as shown
electronic organ, since the latter has a ment to beat music, or as a `band in a in figure 2. It therefore seems more
fixed set of voices, generally imitative of box' to produce television advertising logical from a musical point of view to
conventional organ voices, whereas the jingles. Fortunately the capabilities of have a linear control voltage versus
synthesiser has no fixed tonal charac- synthesisers have been fully exploited octaves characteristic. Figure 3 shows a
teristics. The synthesiser may be used to by musicians such as W. Carlos, control characteristic of octave/volt
1

imitate conventional instruments, but K. Emerson, P. Moraz et al. (which is the standard generally adopted).

SAMPLE SYSTEM 4 9720-1

formant - the elektor music synthesiser elektor may 1977 - 5-15

This exponential control has several

other advantages which will be discussed
later.
An exponential (octave linear) control
characteristic may be achieved by
preceding a frequency linear VCO or
current -controlled oscillator by an
exponential generator whose output
voltage or current doubles for each one
volt increase in input voltage (figure 4).
The exponential generator can be
preceded by a summing amplifier into
which is fed the main control voltage
along with other voltages such as a D.C.
offset voltage to transpose up and down
the scale, and/or A.C. modulating
voltages to produce vibrato effects.

Keyboard Voltage Control

In order to play the synthesiser there
must be some method of feeding
varying control voltages into the
instrument. Since most (Western)
musical instruments are tuned and
played in the tempered tonic scale it
seems logical that a synthesiser should
conform to this scale, and the most 32 -0.. frequency ratio
obvious choice of 'input terminal' is a
normal organ keyboard with electrical
contacts.
The keyboard circuit (figure 5) consists
of a potential divider chain comprising
equal value resistors, fed from a current
source. Since there are twelve semitone Advantages of exponential control groups. The pitch of each VCO may be
intervals (and hence twelve key contacts) Figure 7 illustrates the principal varied relative to other VCO's within a
to an octave, each resistor has a advantage of exponential control, which group to obtain the required chord, by
potential difference of 1 /12 volt across is chording. This shows three VCO's adjusting the independent tuning
it. Depressing a particular key connects each with three summing inputs. The controls.
the voltage on that key contact out to 2. The pitch of a chording group may he
first input of each VCO is commoned varied by adjusting the chord trans-
the common bus rail, and thence to the and connected to the keyboard. The position control.
voltage -controlled circuits. second input of each VCO is connected 3. The pitch of the entire synthesiser
to an independently variable D.C. offset may be transposed by an overall trans-
voltage, while the third inputs are all position control that adds a variable
Transposition commoned to another variable D.C.
Like an electronic organ, a synthesiser that the offset voltage to the keyboard output.
offset voltage. Suppose
keyboard has only a limited compass independent offsets are adjusted so that
(three octaves in the case of Formant). the adjacent VCO frequencies are one
In an organ a wider compass is obtained octave apart, e.g. 1 kHz, 2 kHz and Voltage Controlled Filters
simply by selecting voices with a lower 4 kHz. If the keyboard input increases Voltage controlled filters employed in
register. In a synthesiser the compass is by one volt then the frequencies will synthesisers are commonly of the
extended by adding a D.C. offset increase to 2 kHz, 4 kHz and 8 kHz lowpass type. A block diagram of their
voltage to the VCO input (or to the respectively, which are still one octave operation is given in figure 8. A D.C.
keyboard output) to transpose the range apart. This would not be the case with a voltage sets the cutoff frequency
of the keyboard. This is shown in linear VCO. As an example, suppose the relative to the pitch of the VCO's, while
figure 6. An offset of +1 volt transposes first frequency increases by I kHz to a control voltage derived from the
the range up one octave, while an offset 2 kHz; the second will also increase by keyboard shifts this cutoff point up or
of -1 volt transposes it down one 1 kHz (to 3 kHz), and the third will down according to which note is played,
octave. increase to 5 kHz. This 2-3-5 kHz group so that all notes played, whether high or
no longer shows an octave relationship. low, have the same harmonic content.
Of course, with exponential control one Natural sounds are characterised by
is not confined simply to octave chords. dynamic changes of tone colour. A note
By adjusting the independent offset may start by having a 'bright' character
controls the VCO's may be set up in any with a large proportion of the higher
musical interval relationship. Addition- harmonics, but these then decay rapidly
ally any number of VCO's may be leaving only the fundamental and lower
employed. The commoned third inputs harmonics. Provision must therefore be
of the VCO's permit a common offset made to vary the cutoff point during
voltage to be fed to each VCO to the note. e.g. the cutoff point might
transpose the whole chord up or down initially start off at a fairly high
the scale. A further possibility is to add frequency, which would decrease with
an offset voltage to the keyboard time to cause the decay of the higher
Figure 1. A comprehensive commercial music harmonics. This is achieved by means of
synthesiser, the Modell 'Sample System 4'.
output to transpose the pitch of the
entire synthesiser. This may seem a little an envelope shaper which generates a
Figure 2. Showing the exponential relation. confusing at first, but is not so in varying voltage having the required
ship between relative frequency in octaves practice. To summarise: characteristics. The envelope shaper is
and absolute frequency in Hertz. I. VCO's may be arranged in chording controlled by a gate pulse which is
5-16 - elektor may 1977 formant - the elektor music synthesiser

derived from a second set of contacts on

the keyboard. The voltage versus cutoff
frequency characteristic of the VCF's is
again made exponential by preceding
the control input of the VCF with an
exponential generator.

Voltage controlled amplifiers

The VCA's are simply amplifiers whose
gain may be varied by a control voltage.
Their function is to control the duration
of a sound, and also its dynamic
amplitude characteristics, i.e. its attack,
sustain and decay. The VCA is again
controlled by an envelope shaper whose
output voltage has a form corresponding
to the amplitude envelope of the
required sound. The VCA of course has
no control voltage input from the
keyboard, since the amplitude of all
notes must remain the same and does
not depend upon the frequency of the
note being played.
The envelope shaper which controls the
VCA is itself controlled by a gate pulse
derived from the second contact set on
the keyboard, and this determines the
duration of the note played.
In the case of both the VCF's and VCA's
the output voltage characteristic of the
controlling envelope shaper (i.e. the
manner in which the envelope voltage
varies with time) can be adjusted by the
musician. This is extremely important
since the dynamic characteristics of a
sound largely determine the character of
the sound. Returning to conventional 4
musical instruments as an example, if
the attack transient at the beginning of
a note is removed and only the steady
note is played then it becomes
extremely difficult to distinguish
between many orchestral instruments. Summing amplif ier Exponential Linear VCO,
Indeed, it becomes difficult even to generator VCF or VCA
determine whether sounds so treated
belong to string, brass or woodwind
families.

keyboarc
W output

IxRT-1;'12V
formant - the elektor music synthesiser elektor may 1977 - 5-17

Figure 3. In common with other synthesisers,

Formant has an exponential or octave -linear
control characteristic of 1 octave/volt, i.e. if
the control voltage is increased by 1 volt the
frequency doubles.

Figure 4. Principle of a voltage -controlled

synthesiser module according to Moog. This
forms the basis of the voltage -controlled
modules in Formant.

Figure 5. Showing how the control voltage

is derived from the keyboard using a resistive
potential divider.

Figure 6. By adding a DC offset voltage to

the summing input of the VCO along with the
keyboard voltage, the frequency range can be
transposed. For example, an offset of +1 V
makes the note one octave higher than that
actually played on the keyboard. An offset
of -1 V would make it one octave lower.

Figure 7. To play a chord, the offset of

several VCO's may be adjusted to give the
required musical intervals. They can then be
controlled simultaneously by the keyboard,
and may also be transposed together by a
common D.C. voltage. This is known as
'chord transposition'.

Figure 8. The voltage -controlled filter (VCF)

is controlled in exactly the same manner as
the VCO. The keyboard controls its cutoff
frequency, which can also be 'transposed' by
a D.C. offset voltage. The third input allows
dynamic changes of cutoff frequency during
the playing of a note by means of an envelope
shaper.
Synthesiser Block Diagram and
Additional Circuits
Figure 9 shows the block diagram of a
basic synthesiser, which contains all the
circuits previously described plus a few
extras.
At the bottom of the diagram is the
keyboard and its interface circuits.
These consist basically of the control
voltage potential divider, the gate pulse
generator and the envelope shapers. In
addition there are low -frequency
oscillators that provide a signal for
periodic modulation of the voltage
controlled circuits (e.g. for effects such
as vibrato) plus a noise voltage generator
to provide random modulation.
Ile block containing the VCO's needs
little explanation, except for the
addition of a noise generator. Since this
generates a stochastic signal of no fixed
pitch its frequency is not controlled by
the keyboard, but the noise signal can
he processed by passing through the
VCF's and VCA's to produce effects
such as wind, water, rain, thunder,
steam trains, applause etc., as well as
totally unnatural sounds.
5-18 - elektor may 1977 formant - the elektor music synthesiser

The VCO and noise signals are passed

through the VCF and then through the 9
VCA, both of which are controlled by IIF
tone
the envelope shapers. generation
C

Tonal Character of Synthesisers NOISE

The foregoing description of the basics
of synthesisers can hardly give any
impression of the range of tonal possi- VC01 Sound shaping
bilities available with a synthesiser. To r
begin with, it should not be imagined
that the sound produced by a
synthesiser is like that produced by an
organ. A synthesiser has
electronic
control
voltage
'ill'- VCO2 VCF VCA -r -C>
output
much more in common with conven- I

I
.J
tional musical instruments. Whereas the
sound of an electronic organ has a fairly
static character, a synthesiser is much VCO3 A A
more lively and dynamic. The reasons ADSR ADSR
for this are twofold. Firstly, a synthesiser
control voltage
permits precise control of the dynamic sources
_J

characteristics of the sounds produced,

whereas an electronic organ (unless it is
an expensive one) has only fixed attack 97209
and decay characteristics that must
suffice for every voice. Secondly, keyboard and
whereas most electronic organs are fixed interface
phase, the synthesiser, with its phase - I
A - envelope generator
gate pulse
independent VCO's, can much better 3x
LFO
:

B = random modulation
produce more 'natural' sounds, which 1 generator
have varying phase patterns. C = noise generator
(To be continued) N
Figure 9. Layout of a basic synthesiser.
10 :ram.Plll Several additional blocks are shown such as
low -frequency oscillators and noise generator
to modulate the voltage -controlled modules,
and a noise generator to produce effects such
as wind, rain etc.
(.77267i la 1'1' .1". Fifure 10. Forerunner of modern disco -
lighting? An eighteenth century 'light organ'.

Literature
Moog R.A. 'Voltage -controlled
electronic music modules'.
JAES 13.3 July 1965.
'A voltage -controlled low-pass, high-pass
filter for audio signal processing'.
AES Preprint 413, 1965.
Burhans R. 'Simplified educational
music synthesiser'. JAES 19.2,
February 1971.
Orr. T. and Thomas D. W., 'Electronic
Sound Synthesiser'. Wireless World,
August -October 1973.
Simonton J. 'Build a modular electronic
music synthesiser'. Radio -
X 7 H Electronics, May -October 1973.
Shaw G. 'P.E. Sound Synthesiser'.
Practical Electronics, February 1973 -
February 1974.
Tiinker H. 'Electronic -Pianos and
.1:11..`rr,Lterij; Synthesiser'. Franzis-Verlag, 1975.
Nr. 302 (RPB Serie).
from quadro to surround sound elektor may 1977 - 5-19

OffOM quadro The aim of sound reproduction is to

give pleasure in listening. From the
earliest days of mono it was appreciated
that pleasure and realism go together.

outtound ff-ound Not only did the pioneers strive to

extend frequency -range and reduce non-
linear distortion, but they sought to
refine microphone techniques to give
the sense of space and depth of live
P. Fellgett and M. Gerzon music. Yet realism was still marred by
lack of the sense of direction of direct
and reverberant sound. Stereophony
was the first step in filling this gap,
and in the hands particularly of
A.D. Blumlein achieved great refine-
ment of technique, and an open trans-
parent sense of space of great beauty.
Even this was incomplete because
direction could be reproduced only over
a limited stage in front of the listener.
In previous articles ('Quadrille' and 'music of The final step has been taken in the
present decade by the development of
the spheres', Elektor 17, September 1976, surround -sound technology capable of
representing direction all around the
p. 910) it was noted that 'Ambisonics' was listener.
Speech, and many natural and everyday
perhaps the most promising of the quadrophonic sounds, can give listening pleasure, but
systems proposed so far. This led to a reaction music is unique in its power to sustain
repeated listening. The first requirement
from Prof. P. Fellgett, pointing out (amongst of a surround -sound system suitable for
recording is therefore to sound musical.
other things) that he preferred to use the phrase It must be able to fulfil the needs of
synthesized avant-garde or 'pop' music,
'surround sound' since 'quadrophonics' rather and equally to reproduce the sonorities
and ambience of all the music written
over -stresses the number four - quite apart from for performance from earliest times
through baroque, classical and romantic
being a condemnable mixture of classic tongues. to the present day. This requirement is
Since we had very little information on in figure 1. The test is not
conformity .with any artefact that is a
Ambisonics at our disposal, we took the technical means to an end, such as a
mixdown or a 4 -track master tape. The
opportunity to ask him to write a short article true criterion is comparison of what the
listener hears with the original live or
explaining the basic theory and philosophy of imagined music.
Ambisonics. His text reached us only three
weeks later (!), and it is presented here in full.
Where some further comments or explanation
seemed necessary, we have given this separately
as an 'editorial note'. But first let us see what The patents rights in the circuits described
in this article are owned by the National
Prof. Fellgett has to say. Research Development Corporation.

Loudspeaker
Lode
feeds
SC, .1 '1d

Loudspeakers

Imagined
sound Transmission by
broadcast or recording

Sound in
Comparison listening room
by
listener

Figure 1. Block diagram to illustrate

evaluation of a good surround -sound system.
5-20 - elektor may 1977 from quadro to surround sound

2
Direction Conversion
Live to studio
encoding
performance 0 microphone format

Steering and 01 Mono

Recording To conventional
polar of virtual processing
(optional) microphones Stereo

Recording or
4 -track
Correction transmission Surround
master tape Mixing Encoding
unit (2, 3 etc. sound
(pairwise
channel)
blended)
9833.2

Ambisonic Ambisonic
Mono sources reverb.
(multitrack, pan, tilt, etc.
potentiometers spread, etc.
multimicrophone,
etc.)

3
Pressure W
SHELF FILTERI
L

0
Speaker
distance 00
Forward
SHELF FILTER"(
compensation

Al I 00
R
cL.x-
2 cc
velocity X
00
117( 00
T 0-+.
Input channels
wc
0
Sideways
velocity Y
SHELF FILTER"(
0 II
Ft/
00
9833.3

(two or three) 90° -shifted

SHELF FILTER la
pressure -jW Phasiness
Speaker layout
compensation
All shelves have shape compensation
identical phase
responses

4
B -format inputs

Sum
RIGHT 0
V

Diff.
LEFT 0
0

9833 4

layout
control
from quadro to surround sound elektor may 1977 - 5-21

Evidently no technology can pass this It goes without saying that any design In the studio
test unless it can handle live perform- which is free from compromise is over - Ambisonic studio technology includes
ance as well as synthetic pan -pot and engineered. A most important require- the soundfield microphone for natural
echo -effects. It is found that live ment is a body of theory competent to sound, and a wide range of pan -pot,
ambient material provides the most show how specific design aims can be reverberation and spread effects con-
sensitive touchstone, for musicality is realised in a minimal design, and to forming to an accurate specification.
quickly lost if the loudspeakers are enable rational engineering choices to Correctly designed ambisonic pan -pots
audible as separate sources of sound, or be made in accordance with stated are actually simpler than pairwise
if ambience pools in particular direc- priorities among competing require- 'quadrophonic' pan -pots, or can be
tions instead of being smoothly distrib- ments. made more versatile. Existing pairwise
uted. The real problem is to leave the The earlier so-called 'quadrophonic' pan -pots can be converted by additional
listener undisturbed by obtrusion of the proposals for surround -sound fell short circuitry. There is provision for using
technical means of radiating sound into of fulfilling many of these demands, and
his room. If this can be solved, it
archive material recorded in pairwise
indeed made no attempt to meet some format, including a comprehensive range
follows almost automatically that a of them. They were conceived before
wide range of synthetic effects, from of choices for the producer to optimise
sufficient systematic understanding was the compromises inseparable from this
the simplest to some of great subtlety, available to avoid arbitrary choices, and inferior method of encoding direction.
can he made available by artificial a proliferation of mutually incompatible
simulation of elements from the A studio format, known as B -format, is
systems resulted. A new approach was recommended which facilitates pro-
response to natural ambient sound. needed.
The purely engineering requirements are cessing and reduces sensitivity to small
also exacting. 'Co he acceptable today, a errors of amplitude or phase, especially
in recording. An attraction of this
technology for surround -sound must Second generation surround -sound
cover all essential needs from studio to format is that material held in it can be
The first of the new generation of encoded at a later date into any reason-
listener. It must provide for the variety surround -sound systems commercially able surround, stereo or mono form for
of microphone techniques and artificial available was the UMX system devel- distribution, so the risk of obsolescence
processing, and be compatible with all oped by Professor D. Cooper (University is minimised. Figure 2 illustrates some
valid philosophies and aims of recording. of Illinois) and Dr. Shiga of the Nippon
It must provide a compatible series of of the facilities available in ambisonic
Columbia Company, who marketed the technology.
means of transmission beginning with a system.
basic 2 -channel implementation suitable To this has been added extensive
for dissemination by existing stereo theoretical work and experiment carried
media (disc, tape, fm) and, extending to out at the Universities of Oxford and and at home
it, effective use of multi -channel media. Reading under the auspices of the British The ear and brain do not locate sound
Very desirably it should enable the basic National Research Development by only one mechanism, but by many.
version to be enhanced by adding a Corporation, who hold extensive The goodness of localisation, and the
third channel of reduced bandwidth or patents in many countries on the impression of naturalness and correct-
amplitude suited to carrier -disc or inventions which have resulted. The ness of sonority and tone, are found to
enhanced fm stereo broadcasting. The complete technology is called improve when different mechanisms
base -band signals must have good mono 'Ambisonics', and within it a compatible agree. Ambisonic decoders are designed
and stereo compatibility. The listener series of signal formats known as 45J to give correct localisation by as many
should not be restricted to having his (a development running number) has spectral (frequency domain) and bi-
loudspeakers in a square (a shape to been internationally agreed. This engin- spectral (first order non-linear) mech-
which few domestic rooms are suited), eering standard is the subject of current anisms of the ear as possible. Decoders
and he should have freedom to enhance submissions to appropriate standards for 45113 include a 'front preference'
his system by using more than four authorities in a number of countries. option to cater for material in which
loudspeakers if he wishes. There is no simplistic 'secret' of frontal sources are predominant. Any
Ambisonics. Its capability of high number of loudspeakers in any layout
performance is due to clearly formu- (within reason) can be used. Decoders
lated engineering and aesthetic aims and for four loudspeakers include 'layout'
a body of experimentally validated and 'distance' controls permitting ad-
theory which places the realisation of justment for the ratio of length to
these aims whithin normal methods of breadth of rectangular layouts, and for
engineering design. Some unique size of room. Decoders can be provided
features may however be especially for trapezoidal layouts, or for five, six
noted. or even more loudspeakers if wished.
Ambisonics is not an isolated 'system' The inter -relations between loud-
but a comprehensive technology center- speaker feeds demanded by psycho -
ing around engineering specifications of acoustic criteria produce the useful
the signals for public distribution. bonus that four loudspeakers can he fed
Specification 45J includes a basic 2 - from only three power amplifiers, or
channel format, designated 45JB, which six loudspeakers from four amplifiers.
is of itself a carefully designed surround - Since the mechanisms of the ear, and
Figure 2. Some facilities in ambisonic tech- sound implementation with well formu- therefore the psychoacoustic criteria,
nology. lated and balanced mono and stereo change with frequency, all decoders
compatibility. This is the lowest (except the most basic for the lower end
Figure 3. Block diagram of a generalised
asobisonic decoder (courtesy MAGI.
member of a compatible series which of the market) employ special phase -
includes members using 3 channels, or compensated shelf filters. This fre-
with a 4th channel which may convey quency -dependence improves not only
Figure 4. Block diagram of a domestic height information for periphony localisation but also the subjective
ambisonic decoder of medium complexity for
2 -channel (45JB) material. This will accept
(spherical surround). Particular atten- tonal quality in comparison with naive
2 -channel or 3 -channel B -format inputs and tion has been given to fulfilling the need decoders lacking this feature.
dallier mono, stereo, superstereo or 45JB for '21/2 -channel' specifications in which Loudspeaker emphasis and signal -
decoded outputs. It can drive rectangular or the 45JB basebands are enhanced by a dependent gain devices can he incorpor-
regular hexagon loudspeaker layouts via 3 or third channel of restricted bandwidth ated in ambisonic decoders for those
4 power amplifiers. Loudspeaker distance and or amplitude suited to carrier disc or occasions when a more gross effect
Psyc-hoacoustic compensation are provided. fm broadcasting. may be deliberately sought. Ambisonic
 +I X W.
format Inputs
jzy. PHASECOMPENSATED
SHELF FILTERS WITH
RESISTIVE INPUT OUTPUT
IMPEDANCE & GAIN MATRIX
SUM & DIFFERENCE MATRIX RESISTORS
8 POLE /ZERO PHASE SHIFTERS
MATRIX
L_
59413E
37.
speaker signal outputs
--I .PE
-Ili 2131
(Dat
CB
Intukki-
100.
1
-.1 ROE I I Wl 90
01.
O
0. MT
i
9)M1.1
G..1
° LB
I'
07 21
-...i 21--... IRKRI
--- --1
-(2922.1- E- _ N.,i
T
LEFT 20th
0, 22
CS
layout
control
3
AT F- 33A
RF
Iasi} I au I
1 SOB CTI/44... PRESET .UR RACE. PANEL SVP PC9
WAV
NORMAL
11.1,ORMAT
NORMAL 0 CS: 3TOR
POLE 3 WAV
I
3 POLE 2 WAVI O
FiTii RB
SPARE 0 s
NORMAL
rpR T''SEO' I
SPARE 0
MONO
STEREO
surtesu aro o
ON, pm, IWAV L Note Many 'precoion'resistors are achieved
by series or parallel pairs
AS JO 0
component values shown are ideal salves
0a 0-1 Tin I--
,a\- LOUDSPEAKER LAYOUT
TTDLL J WAY
O°
from quadro to surround sound elektor may 1977 - 5-23

45J -encoded material can if wished

even he played through the Sansui
`Variomatrix' decoder, with simple
ea
added circuitry. An ambisonic decoder
with signal -dependent gains has been
designed having improved conformity
with psychoacoustic requirements. RIGHT
Figure 3 is a block diagram illustrating
a generalised ambisonic decoder; of
course not all of the features need be
present in a particular consumer
product.
These features of ambisonic decoders
are more or less realisable (by suitable
modifications of circuitry) with source
material encoded in any reasonable way, LEFT
although best results require correct 45J
(or equivalent) encoding. BBC 'Matrix II' +75O13-04._
encoding is close enough to 45JB to be NORMAL -0
preset or back -panel switch

playable directly into ambisonic 45J

decoders with psychoacoustic enhance- Notes: Resistors marked * may be 5%, **may be 10%
Others shown should preferably be 2% or better
ment of performance. There is effective
compatibility with Nippon Columbia
`llenon' UMX, and a simple switching
enables RM, QS and SQ material to be
handled with a degree of approximation
depending on the source encoding. A
`stereo decode' mode is available. This
eb
does not deliberately give a synthetic
pseudo -surround effect, but enables
localisation in the frontal stereo stage to
be improved by the use of four (or
more) loudspeaker playback. 22k 1---41--
Figure 4 is a block diagram of a
0--1 56k 47
domestic decoder of medium
complexity for 2 -channel (45JB)
material. The blocks are expanded into
II-I 100n
actual circuit configurations in figure 5
and these are in turn shown in more
detail in figure 6 (a), (b), (c) and (d).
Figure 7 shows various ways of using
the outputs for four or six loudspeakers
using three or four power amplifiers.
47k 1--

Decoder functions 0---1 22k 1-0- 12,

** 0i
The decoder circuit given here has a
four -position mode switch, labelled
'mono', 'stereo', `superstereo' and II IMIMP470P
`45JB'. It should be noted that the 47n 47n 470p

`mono' and 'stereo' positions are not

obtained by simply switching off
unwanted channels or shorting outputs
together.
*
-1 I--
47k

Mono reproduces mono material through 9833- O.

4 or 6 loudspeakers so that the sound

image appears to be straight ahead, but Notes: Resistors marked *may be 5%,** may be 10%
at a larger distance than the speaker
distance. It avoids the 'coming from a
box' effect of single -speaker mono
reproduction, and the 'in -the -head' and
tone -colour distortions associated with Figure 5. Overall circuit diagram of domestic worsen it). Good material will obtain a
2 -speaker mono reproduction. In ambisonic decoder corresponding to the block further sense of spaciousness without
practice, 'mono' decode helps unlock diagram of figure 4. any gimmickry of directional effect.
the spacious quality contained in good The effect is almost completely
Figure 6a. Detailed circuit of input stages
mono recordings. (sum/difference matrix/.
unobtrusive and yet capable of
Stereo reproduces conventional stereo 6b. Detailed circuit of phase shifter stage. Two considerable reduction of listener
material over the conventional front - of these stages are required, one for the E and fatigue through presenting the ears with
quadrant stereo stage, but uses the extra one for the A signal. Nominal performance: a more natural sound. 'Stereo' decode
speakers to reduce undesirable qualities phase shift 90° ± 1'/3 ° from 30 Hz to also tends to enhance the reproduction
of 2 -speaker stereo reproduction - 16 kHz. A 2% component spread gives 90° of SQ-encoded records.
notably excessive speaker prominence, ± 3° over this range. Stage gain is 1.33 dB. Superstereo gives a wider frontal stage
unstable images and tone -colour from stereo material for those who wish
distortions. 'Stereo' decode does not for a more spectacular effect, without
enhance the quality of badly -recorded descending to mere gimmickry. This
stereo material (but neither does it position also provides excellent decoding
5-24 - elektor may 1977 from quadro to surround sound

of Regular Matrix encoded records, with

6c B.FORMAT \ _S24 MONO
STEREO 0
E,_ 51
a full 360° reproduced stage.
45JB decodes system 45J recordings
NORMAL 0
SPARE 0
SPA RE 0
SURE RSTE REO 0
45 JB 0 and broadcasts. It is also suitable (but
MeakWoro.rnake not optimised) for decoding 'Matrix H'
for good ambient reproduction, and it is
reasonably compatible with BMX
recordings.
B -format enables those with a 4 -channel
recorder to get optimal studio -quality
reproduction from master recordings
made in the ambisonic B -format studio
mode.

Editorial notes.
Let's be honest: we are impressed by the
capabilities of ambisonics. To sum it up
briefly:
- an ambisonic system will accept practically
any input.
- an ambisonic system makes the fullest use
of the available number of transmission
channels.
- an ambisonic system can drive almost any
loudspeaker layout, using a minimum
number of power amplifiers.
- the designers of ambisonics have evidently
done their homework.
Great!
Now let's be practical. At the moment, no
ambisonics recordings are commercially
available. There is very little point in building
an ambisonics decoder until some record
company starts supplying these recordings, or
until some broadcasting corporation starts
transmitting in ambisonics. From a technical
point of view (please note: not necessarily a
commercial or even a consumer point of view)
we wish that this situation were different.
In theory, and even - as we have heard on
first-hand authority - in practice, a surround -
sound demonstration using ambisonics is
unbelievably convincing. Speaking only from
the theoretical point of view, we are prepared
to go along with this: we have favoured UD-4
so far, since this system seemed to make full
use of the available transmission capabilities.
In our opinion, ambisonics is an extension of
UD-4: technology and mathematics have been
complemented with psycho -acoustic research.
We have tried to give all systems an equal
chance in the pages of this magazine, and we
have noted the good points of each system. In
previous articles we have even noted that
there may be room for more than one system.
However, if only one system is to be chosen
for universal application, our money is on
ambisonics - on a purely technical and
theoretical basis. We can't wait to hear a
practical demonstration!
Finally, some comments on this article. It is,
of course, incomplete. It is sufficient to whet
the appetite without giving something to
11011 MEIN bite on. The author explains that they are
still negotiating with various parties and
cannot release more information.
The circuits are quite basic, but they are
approved by the inventors. There are no
alignment points, to our relief, so we don't
need to give alignment instructions. So long as
SEM 6c
there are no commercial releases of ambisonic
recordings, we do not feel obliged to offer a
printed circuit board.
For that matter, Mr. Gerzon (who designed
this decoder within 48 hours!) has pointed
out that there is room for improvement and
simplification. When the time seems ripe for a
constructional article with a p.c. board, we
Components marked *may be 5%, **may be 10%
hope to take him up on this. However, the
circuit shown here does give an idea of what
an 'ambisonics' decoder looks like, and that is
sufficient for the present.
from quadro to surround sound elektor may 1977 - 5.25

Figure 6c. Detailed circuit of resistor matrix +

6d 150.1--
shelf filter stage.

Figure 6d. Detailed circuit of output stages,

J 15-
Y including loudspeaker distance compensation,
I a 1 ---OCR or CB
oo loudspeaker layout control and amplitude
matrix,

Figure 7.Three or four of the outputs from

figure 6 may be fed to power amplifiers to
drive four- or six -loudspeaker layouts. The
'+' sign indicates the positive loudspeaker
terminal in each case.
-f 1

-4 611.-71-
ea6 335
0-4 F5.-40
I-
CR

O
CAO

33. j --
aen

02-1 LIM MEI I PF

BASIC LOUDSPEAKER LA,OuP

-1 335
Notes: 1i Resistors marked *may be 5%, **may be 10%
216/30 nF capacitors may be implemented as
220 nF and 470 nF in paralel
1200 k may be replaced by 180 k and 22k in series
4,11 potentiometer a 4k7, make up to 5 k
by two 150 0 series resistors, one at each end

7c 6 SPEAKERS 1:7:)
4 SPEAKERS Ei 4 AMPS
4 AMPS

RFEO

SQUARE
OR
RECTANGLE
LF00
141. REGULAR
HEXAGON

'0.1)0

CR(y)
9533.73

9833_7c
POWER
POWER AMPS
AMPS

7b
4 SPEAKERS 1:1
3 AMPS

SQUARE
OR
RECTANGLE

9533_70

POWER AMPS
5-26 - elektor may 1977 stereo encoder

toio® one o der

subject of mono compatibility.
Since the total mono signal is the sum
of the left and right signals it seems
logical to modulate the mono or L + R
signal direct onto the carrier. This is
identical to a mono transmission and is
thus fully mono compatible. To
separate the left and right signals in a
stereo receiver it is then necessary to
modulate the difference between the
two signals (i.e. L-R) onto the
subcarrier. At the receiver, having
demodulated the FM signal and the
subcarrier to give the L+R and L-R
signals, the L and R signals can be
retrieved by performing two simple
algebraic operations i.e.
A. Bosschaert L= ( (L+R) + (L-R) )
R = 1/2 ( (L+R) - (L-R) )
This is the basis of the GE -Zenith
multiplex stereo system, which has been
This relatively simple circuit will combine two universally adopted for FM stereo
independent audio signals into a multiplexed transmissions. A mono receiver will, of
course, demodulate only the L+R signal,
stereo signal similar to that used to modulate an the modulated subcarrier being rejected
by a lowpass filter in the receiver.
FM transmitter for stereo transmissions. It can, Figure la shows the frequency spectrum
occupied by a multiplex stereo signal.
amongst other things, be used to check the The L+R signal occupies the normal
audio spectrum from 30 to 15 kHz. The
operation of stereo decoders in FM receivers. L-R signal is amplitude modulated
onto a 38 kHz subcarrier and so
occupies two sidebands extending
To those unfamiliar with stereo FM This is where the concept of frequency 15 kHz above and below this frequency.
transmissions 'multiplex encoding' and multiplexing comes in. The next The 38 kHz carrier is itself suppressed,
`decoding' may seem a little mysterious. obvious step would be to modulate say but in order that it can be regenerated
The immediate answer to the problem the left signal direct onto the FM carrier, at the receiver for demodulation
of transmitting a would but the other signal would first be purposes a low level 19 kHz pilot tone is
seem to be to transmit the left signal on amplitude modulated onto a high - transmitted, which is phase -locked to
one FM channel and the right signal on frequency subcarrier which itself was the original 38 kHz subcarrier.
an adjacent FM channel. However, this modulated onto the carrier along with Figure 1 b shows how the L-R signal is
solution is unattractive for several the left channel audio signal. In the modulated onto the subcarrier. In this
reasons, for one thing it would occupy receiver the total FM signal would be example the L and R signals are shown
two FM transmitters to send out one demodulated to give the left channel as simple sinusoidal signals of 2 kHz and
programme, and two receivers would be plus modulated subcarrier. Lowpass 4 kHz.
required to receive it. Nevertheless, filtering to remove the subcarrier would
experiments on these lines were carried then give the left signal, while highpass
out, and occasionally are still carried filtering to remove the left channel plus
out for quadrophonic transmissions, the AM detection would yield the right
front channels being transmitted by one signal. Figure la. Spectrum of a multiplex encoded
transmitter and the rear channels being This idea unfortunately falls down on stereo signal.
transmitted by another. mono compatibility, since people
Figure lb. Showing modulation of the L-R
For a practicable system, some method equipped with only a mono receiver signal onto the 38 kHz subcarrier.
had to be found of modulating both left would be able to receive only the left
and right signals on to a single channel! Since the majority of listeners Figure 2. Pre -emphasis of the signal above
transmission in such a manner that they possess only mono receivers the broad- 3 kHz improves the high -frequency signal-to-
could be separated at the other end. casting authorities are very keen on the noise ratio.

la

amplitude
L+R L L R
! .20 .30 ,35 ao ,50 .55 60
15 110 is ,25 10
9791 to

10Hz 15 kHz 19 kHz 15 kHz 38 kHz 15 kHz

frequency
stereo encoder elektor may 1977 - 5-27

lb left, 2 kHz right, 4 kHz

EC A 220
::,.L.,

16 15 14 13 12 11 10 9

IFI0111111
elka.
1 2 3 4 5 6 7 8
.1 c9a> i22 E ,
-
10.12.16 - cur9R6,94N,-,

11%e
man -2200 06 to ../..

38 kHz subcarrier 0 CD 4069 BE

14 13 12 11 10 9-1.51-1mr
__I I_ _J L __I l --

>c__.
1 1 II)

1 2 3 4 5 6 7

CD 4016 AE
Oy CD 4066 BE

14 13 12 11 10 9 8

modulated subcarrier
1111

,- -
A
1 2 3 4 5 6 7

CD 4013 AE

0
multiplex signal 14
02

13
52

12
C17

11
B7

10 8

L 9I

4W11,1b _
'

1=6.81116, 7111 II - _
R

flif MOW
thM
-1
2
_

3 4
P
-I
5
1,
6
Si y
7

9791_ 1 b

Pre -emphasis and De -emphasis Pre -emphasis is generally quoted as the

It has long been known that most of the time constant of the RC network
energy content of naturally produced necessary to produce the required pre -
emphasis curve. In Europe pre -emphasis
audio signals, including speech and of 50 ps is used, while in the USA
amplitude musical instruments,
(dB)
conventional 75 As is the standard.
16, 1406 pre -emphasis occurs below about 3 kHz. If an audio At the receiver, in order to produce a
signal were frequency modulated onto a
flat audio frequency response, a de -
carrier with no processing then, because emphasis network must be employed
of the lower amplitude of high - having an inverse characteristic to the
frequency signals, the signal-to-noise
pre -emphasis network, i.e. a fall of
ratio would tend to be worse at the top
end of the audio spectrum. For this 6 dB/octave above 3 kHz. This network
reason signals above 3 kHz are boosted comes after the stereo decoder, so in
or given a pre -emphasis before the audio fact each channel has its own de -
36
30
frequency signal is modulated onto the carrier. The emphasis network.
Hz
pre -emphasis curve is given in figure 2
9791
and is simply a 6 dB/octave lift above Block diagram of the encoder
3 kHz up to 15 kHz (for European Having examined the composition of
Broadcasts). the multiplex signal it is now possible to
5-28 - elektor may 1977 stereo encoder

formulate the requirements for a stereo modulated subcarrier, pilot tone and
encoder, and a block diagram is given in L+R signal are summed together before
figure 3. Firstly, the L and R signals being fed to an output amplifier.
must be fed through pre -emphasis It should be stated at this point that the
networks to boost the high frequencies. multiplex signal produced by this
The two signals are then summed to give encoder does not comply exactly with
the L+R signal, the right channel is broadcast regulations. For one thing
inverted and summed with the left sinusoidal waveforms are specified for
channel to give the L-R signal. The the subcarrier and pilot tone. Since the
38 kHz subcarrier is derived by dividing encoder described here uses digital
down the output of a 76 kHz oscillator, division the waveforms produced are
process ensuring a Figure 3. Block diagram of the stereo encoder.
the division squarewaves. However, the signal
waveform with a 1:1 mark -space ratio. Figure 4. Complete circuit of the stereo generated is quite adequate for simple
The 19 kHz pilot tone is similarly encoder. testing of stereo decoders, and the cost
derived by dividing the 38 kHz signal. of an encoder to the full official specifi-
The 38 kHz subcarrier is then amplitude Figure 5. Pinouts of the IC's used in the cations would hardly be justified for the
modulated by the L-R signal, and the encoder. home constructor.

4
142 N3

9V
C4. P1 R17
R24
47k
R22
f, 50k

Ift.MI CIE 330k H P3

I 18011-7:-. so. NI C11 835
R5
0-.-4 47k R34 C12
100k
R19 R23
270p
, 615
47k

4i2

R3
Al 614
100k
A2

618
621
33k

R.14 P2
-deo-
ES ES 1 CI OCK

FF 1
0 CLOCK

FF 2
j
a
50k R

CI. C21 C9 .1.

C10JIOV

T220P 710p T220P

9V
C7 .25 R31 #3
630 0.17
900 4
R10 810

R37

(-.14 MPX
LN aU
2p2
10V
A3
R12
33k
R26
SEM
R29
-000- 22 03
(117
tUV
0
ES 3 ES 4 BC 548
C13
840
87 y 89 827 3x1N414B
BC 548 100711-i
p 111
R36 R38 39
'005
10V
Si
CO

7 In T2200
0 FF 1,2= CD4013AE
s
9751-4

Nl. N4 .=. CD4069BE

Al . A3= TCA220
ES1 .ES4= C04066AE
stereo encoder precision v to f converter elektor may 1977 - 5-29

Complete Circuit pulses whose duration (T2) is a constant

The various functional blocks of the
circuit can easily be identified in the
complete diagram of figure 4. Al and
precibion and whose spacing (T1) depends on the
output voltage of IC2.
The output voltage of ICI is filtered by
A3 operate as input buffer amplifiers
having unity gain below 3 kHz. Above v to ic R6 and C3 to give a DC voltage (V2)
equal to the average value of the IC1
3 kHz the feedback networks R5/C3
and RI 0/C8 provide the rising pre -
emphasis characteristic. The outputs of
Al and A3 are summed by R12 and
converter waveform i.e.
V2 = 15 x -.
T2
Ti
Since T2 is constant V2 is proportional
to 1/T1 i.e. proportional to the output
R13 before being fed to an electronic frequency of IC1.
switch ES3, which can be used to switch The input voltage V1 is applied to the
the L+R signal in and out. inverting input of IC2, which functions
The R signal is inverted by A2 and the as an integrating comparator. If V1 is
L and -R signals are summed by R21, less than V2 the output of IC2 will
R20 and P2. With this simple summing
arrangement there exists the possibility ramp positive. Cl will thus discharge at
of crosstalk, but this can be minimised a slower rate, making the interval T1
by the adjustment of P1 and P2. longer and reducing V2. If V1 is greater
than V2 then the output of IC2 will
The 76 kHz oscillator is constructed ramp negative (i.e. towards zero)
around two CMOS inverters N2 and N3, Although this Voltage Controlled
Oscillator uses only two CA3130 op making Cl discharge more quickly and
the frequency of oscillation being reducing T1. When V1 and V2 are
adjustable by P3. If desired C12 could amps the linearity of its voltage/fre-
quency transfer characteristic is better equal the output voltage of IC2 will
be replaced by a 76 kHz crystal to remain constant.
improve the oscillator frequency than 0.5% and its temperature coef-
ficient less than 0.01%/°C. The circuit will thus always reach
stability, but this expense is probably equilibrium with V1 = V2. However,
not justified. Two CMOS flip-flops FF1 The circuit operates as follows: ICI
functions as a voltage controlled multi - since V2 and the output frequency of
and FF2 divide down the 76 kHz signal ICI are proportional then the output
to provide the subcarrier and the pilot vibrator. Assuming that the output volt-
age of IC1 is initially +15 V, Cl will frequency of IC1 is also proportional
tone. S3 can be used to inhibit the pilot to the input voltage V1 , since V1 and
tone by disabling FF1. charge via D3, R4 and P1 with a time
constant (R4 + P1) CI until the voltage V2 are equal.
Since the subcarrier must be completely A few small refinements are added to
suppressed a balanced modulator is on the inverting input of IC1 exceeds
that on the non -inverting input. improve the temperature stability of the
employed. This consists of two electronic circuit. The temperature coefficients of
switches ES1 and ES2, part of a Neglecting DI and D2 for a moment
this is approximately 10 V, set by the D3 and D4 could introduce errors by
4066 IC, and a CMOS inverter Ni, varying the charge and discharge times
connected as an inverting linear potential divider Rl, R3 and R2. The
output of ICI will now swing down to of CI, so these are compensated by
zero and the voltage on the non - including identical diodes in series
The modulator output passes through with R3 to produce a similar variation
another electronic switch ES4, which inverting input will fall to about 5 V
due to the hysteresis introduced by R3. in the reference voltage at the non -
can be used to switch in and out the inverting input of IC1.
modulated L-R signal by means of S2. Cl will now discharge into the output
of IC2 at a rate determined by R7 and P2 is included to null the offset voltage
The L+R signal, modulated L-R signal of IC2 which would otherwise cause a
and pilot tone are all summed at the the output voltage of IC2 until the volt-
age on the inverting input of IC1 falls zero error.
base of T1 , T1 and T2 acting as an P1 provides fine adjustment of the
output buffer amplifier to provide a low below 5 V, when the output of ICI will
swing up to +15 V again and the cycle conversion ratio, which with the values
output impedance. P4 adjusts the gain given is about 1 kHz/volt.
of T2 and hence the amplitude of the will repeat. The output waveform of
multiplex signal. IC1 thus consists of a series of positive Lit.: RCA Application Notes.

Adjustment f011kHz/V1 VI= 0...10V

To check the operation of the encoder,

first set the oscillator frequency to
aO
76 kHz using a frequency counter, then
adopt the following procedure. 0 V+.. -15V
1. With only the L+R signal switched in, R5
connect the encoder output to the
external input of an FM test generator C4

and use P4 to adjust the deviation. 10n

2. Connect the generator output to a RI R3

properly aligned FM receiver, tune to

the generator frequency and switch in T 100n

the pilot tone and L+R signals. The

ICI
stereo beacon on the tuner should light.
CA3130
3. Alternately switch off the left and cl R6
right channels and adjust P1 and P2 for -.4 1M
maximum separation. i.e. with left 47017

signal applied adjust for minimum D3

P1
00T
output from the right channel and vice II
10k DM
82p

seTsa. MIR Al
4. Alternatively the multiplex signal can 1
1k

be fed direct to a stereo decoder. P4 C3

should be adjusted so that the multiplex Ton
signal level is within the operating range 9715
0
of the decoder. Then follow pro- D1 D5 DUS
cedure 3.
5-30 - elektor may 1977 slotless model car track

(23

As described in last
month's article, the
model cars are
powered from the special race
track, but steering and speed
commands are provided by an
infra -red communication link. The speed and direction of up to four
cars can independently be controlled using a 9 -channel time -division
multiplexed 'digi-proportional' control system. This month's article
describes the circuit of the multiplex encoder, and the principle of the
decoder. The board layout for the infra -red transmitter will also be
given.

Last month it was described how, in a in the correct sequence. The multiplex to 'freeze' its operation at a certain
digi-proportional control system, the decoder at the receiving end must be point in the cycle, and to make certain
position of servo's was proportional to capable of 'picking out' each pulse and assumptions. Firstly, it is assumed that
the width of a rectangular control pulse. feeding it to the correct servo amplifier. R4 is much less than R3 and RI is much
The range of pulse width chosen was less than R2. Secondly, assume that T1
1 ... 2 milliseconds so that, for example, is presently turned off, but is about to
the extreme left position of the steering Multiplex encoder turn on (how it turns on will become
servo corresponds to a pulse width of The multiplex encoder uses circuits that apparent). Since T1 is turned off Cl is
I ms, the extreme right position to 2 ms, will be familiar to the experienced con- charged via RI and the base -emitter
and the centre position to 1.5 ms. structor, namely, an astable multi - junction of T2 to Vb - 600 mV. If TI
Since several servo's are to be controlled vibrator to generate clock pulses to now turns on the left hand end of Cl
the control pulses for each servo are control the whole sequence, monostable will be grounded through the collector -
time division multiplexed into a pulse multivibrators to produce the control emitter junction of T1, hut since the
train. A particular pulse in the train pulses and a NOR -gate to collect all the voltage across Cl cannot change instan-
corresponds to a particular servo, and pulses into a pulse train. taneously the right hand end falls to
the width of each pulse can be indepen- For the less experienced constructor the - (Vb -- 600 mV), turning T2 hard
dently controlled. The multiplex encoder basic circuit of an astable multivibrator off. C2 will now charge rapidly to
must be capable of producing the is explained in figures 1 a and lb. Since (Vb - 600 mV) through R4 and the
different pulses to control each servo an astable multivibrator is an oscillatory base -emitter junction of T 1. The large
and assembling them into a pulse train circuit, to understand it, it is necessary pulse of base current delivered to T1
slotless model car track elektor may 1977 - 5-31

will turn Ti hard on, thereafter T1

remains turned on by current flowing
through R2.
Cl will discharge slowly through R3 and
the collector -emitter junction of T1
until the voltage on the base of T2
exceeds 600 mV, when T2 will turn on.
The base of Ti will be pulled down to
- (Vb - 600 mV) and T1 will turn off.
Cl will charge rapidly through RI and
the base -emitter junction of T2. C2 will
charge slowly through R2 and the col-
lector -emitter junction of T2 until the
base voltage of T1 exceeds 600 mV,
when T1 will turn on, which is where
we came in.
The time for which each transistor is
cut off is the time taken for Cl or C2
to charge from - (Vb - 600 mV) to
600 mV, and is determined by the time
constants R3 -C1 and R2 -C2.
The time tl for which T1 is turned off
is given by R2C2ln2 and the time t2
for which T2 is turned off is given by
R3 -C1 -1n2, where 1n2 is the natural
logarithm of 2 (approximately 0.7).
Figure lb shows the waveforms at
various points in the circuit. Waveform
A shows the collector voltage of T1.
The base waveform of T1 is shown in
B and the exponential rise from a nega-
tive voltage can clearly be seen.
Waveforms C and D show the voltages
on T2 collector and base.
The period of the total waveform is
t 1 + t2 and the frequency, being the
reciprocal of the period, is
1

ti + t2
An astable multivibrator may he made
either symmetric or asymmetric i.e.
with t 1 equal to t2 or t 1 not equal to
t2, and it may be useful at this point to
introduce the concept of duty -cycle and
mark -space ratio. The duty -cycle of a
rectangular waveform is the time for
which the waveform is positive divided
by the total period times 100%. Thus
the duty -cycle of the waveform at the
collector of T1 is
t1=R2.C2In 2 0,7. R2.C2
t
x 100%. t2= R3C1 1n 2 0,7413.C1
ti + t2
On the other hand the duty -cycle of the
T2 collector waveform is
t2
X 100%. Figure la. Basic circuit of an astable multi - almost zero so the base of T1 is
ti + t2 vibrator, used as a clock generator in the grounded and Ti is turned off. C2 is
The mark -space ratio of the waveform is multiplex encoder. charged to (Vb - 600 mV). If a short
simply tl/t2, or t2/t1 looking at the positive pulse is applied to the base of
Figure lb. Waveforms at various points in the
collector of T2. astable circuit.
T1, TI will turn on, grounding the left
hand end of C2. The right hand end of
C2 will take the base of T2 negative,
Monostable multivibrator turning off T2. The collector voltage of
Unlike the astable multivibrator, which T2 will rise and T1 will be kept turned
is an oscillator with no stable or 'rest' on by current flowing into its base
state, the monostable, as its name through R4, even after the trigger pulse
implies, has one stable state. It also has has disappeared. T2 will remain turned
an unstable state into which it can be off until C2 has charged via R2 and the
flipped by a trigger pulse, and it will collector -emitter junction of TI to
return to the stable state after a preset +600 mV, when T2 will turn on again
name. and T1 will turn off.
Figure 2a shows the basic circuit of a The time for which T2 is turned off is
monostable. In the stable state T2 is exactly the same as for the astable
nursed on by current flowing into its multivibrator i.e. R2.C2.1n2.
base through R2. Its collector voltage is Figure 2b shows the various waveforms
5-32 - elektor may 1977 slotless model car track

in the monostable circuit. Waveform A settings of P1, and waveform B shows Figure 3b shows a multi -input NOR -gate
represents the trigger pulse and B shows how this affects the pulse width. using resistor -transistor logic (RTL)
the base voltage of Ti. Waveform C is fabricated from discrete transistors. If
the collector waveform of T2 and NOR -gate all inputs are low then all transistors will
waveform D the base waveform. The final main circuit element used in be turned off and the output will be
Figure 2c shows a modification to the the multiplex encoder is a NOR -gate. pulled high by the common collector
monostable multivibrator to allow the This collects together the pulses from resistor. However, if any input goes high
pulse width to be varied, which is the different monostables and converts the corresponding transistor will turn
essential in this application. P1 controls them into a pulse train. Figure 3a on and pull the output low.
the voltage to which C2 is connected shows a two input NOR -gate together
with its truth table. It can be seen from Differentiator and integrator
while T 1 is turned on, and hence the
negative voltage level applied to the base the truth table that if either input goes Two other simple networks are exten-
of T2 when T2 is turned off. This in high then the output will go low. sively used in the multiplex encoder.
turn varies the time it takes C2 to A multiple input NOR -gate can thus be These are the differentiating network
charge back up to +600 mV, and hence used to collect pulses from the outputs shown in figure 4, and the integrating
the time for which T2 is turned off, i.e. of several monostables. If all inputs are network shown in figure 5. The most
the pulse width. This is shown in fig- normally low then if any input goes useful property of the differentiating
ure 2d. Waveform C shows the negative high a corresponding low -going pulse network is that of producing short
base voltage applied to T2 for different will appear at the NOR -gate output. pulses from step inputs. If a step input,

2b

vinf

Vb

vB1t1Vs

voutlVB
t3

vB2f

Vb.

Vs

-Vb
t3 = R2.C2.1n2 z 0.7 R2.C2 9796-26
 elektor may 1977 - 5-33
slotless model car track

such as the leading edge of a square -

wave, is fed to the network then the
output voltage across R will initially
be the same as the input voltage. How-
ever, as C charges the voltage across R
will fall exponentially according to the
equation
-t
\Tout = Vb eRC
When fed with complex AC signals the
differentiator functions as a simple
high-pass filter, attenuating the low
frequency components of the signal.
As might be expected, the integrating
network functions in a more or less
opposite manner. When fed with a step
input the voltage across C is initially
zero and rises slowly as C charges
through R according to the equation
-t
Vout = Vb (1 _ eRC)
The integrator is thus useful as a sort of
delay network, and indeed is used as
such in the astable and monostable
multivibrators.
When fed with an AC signal the inte-
grating network becomes a simple
lowpass-filter.

Multiplex Encoder -
block diagram
Having discussed the various elements
that make up the multiplex encoder, the
block diagram is given in figure 6a. The
clock generator (an astable multi -
vibrator) produces a 5 ms pulse every
25 ms, which triggers monostable MF1.
This produces a pulse whose width can
be varied by P1, and when it resets it

Figure 2a. Basic circuit of a monostable

multivibrator.

Figure 2b. Waveforms in the monostable

circuit.

Figure 2c. P1 can be used to adjust the mono -

stable pulse length.

Figure 2d. Showing how different positions of

P1 vary the time for which T2 is cut off.

Figure 3a. Symbol and truth table of a two-

:riput NOR -gate.

Figure 3b. A multi -input NOR -gate using

nesistor transistor logic.

Flure 4. A differentiating network, used to

?e-werate short pulses from step inputs.

ure 5. An integrating network, used in the

r.7chronising circuit of the decoder.
5-34 - elektor may 1977 slotless model car track

CLOCK MF' K1 MF2 K2 MF3 K3. MF4 P<,U MF5 MF6 M17 KT ME8 .KS. MF9 KS

O MF10
0jL. MPX

Do-
9796 K.S.

triggers MF2. When MF2 resets it

triggers MF3 and so on up to MF9. Each
clock pulse thus produces a train of nine
servo control pulses. Each monostable is
equipped with a control potentiometer
Inn to vary its pulse width and thus control
its assigned servo.
L Since each monostable pulse starts as
KI KI the previous one finishes it is not poss-
ible to collect the monostable pulses
0 directly and feed them to the trans-
0 mitter. Because the pulses follow so
closely the NOR -gate output would
0 simply be a continuous low level, or at
0 best a low level with extremely short
positive spikes at the switchover point
O K6 KS between one monostable and the next.
Such a spiky waveform would require a
very large transmitter bandwidth.
0 KS To overcome this difficulty the outputs
0 K9
of the clock and the even -numbered
76r 9K8 K K. K6 Ka
monostables are fed into one NOR -gate,
while the odd numbered outputs are fed
0 21 rl K,r1 KS 61 r1K9 into a second NOR -gate. At this point it
0 may he useful to refer to figure 6b.
Waveform (a) is the clock output, and
waveforms (b) to (j) are the monostable
0 outputs. (k) and (1) are the outputs of
the two NOR -gates and it can be seen
0 1 141.1951.11., 1991 1 II I I I 11 1 1 that (1) is almost an inverted version
of (k).
0 The next step is to differentiate the two
NOR -gate outputs to produce a series of
91 laK31 K.I KS 1.91 short spikes (m and n). Referring to
figure 6b it can be seen that these spikes
define the leading and trailing edges of
the monostable pulses. Combining
waveforms (m) and (n) in a third NOR -
gate gives waveform (o), which is simply
 slotless model car track elektor may 1977 - 5-35
Figure 6a. Block diagram of the multiplex
encoder 6V3
8..12V 20,nA BC 5476 0c4Lc4.6.1.
Figure 6b. Timing diagram for the multiplex 100700n
encoder.

Figure 7. Complete circuit of the encoder.

The transistors are not discrete devices, but
IC transistor arrays.
T 1 T5 IC1 = CA 3086
T6. T10 = IC2 CA 3138,6
711..T15 - IC3 = CA 3086
T16 .. T20 IC4 = CA 3086
T21 ... T25 = IC5 - CA 3086

a train of pulses the same lengths as the

original monostable pulses with short
negative -going spikes between them.
However, this waveform is still too
`spiky' for the transmitter, so the nega-
tive going pulses are used to trigger yet
another monostable, which 'stretches'
the spikes to a constant 400 ps width.
Both normal and inverted versions of
the multiplexed signal are available at
the output of MFIO.
It will be noted that, since the maxi-
mum monostable pulse width is 2 ms
the entire pulse train occupies a maxi-
mum of 18 ms which, since clock pulses
occur every 25 ms, leaves a gap of at
least 7 ms at the end of each pulse train,
more if the monostables are set to a
shorter pulse length. This gap is used to
synchronise the receiver decoder with
the encoded signal, so that each servo
receives only the correct pulses.
I
I Complete circuit
Tne complete circuit of the multiplex
encoder is given in figure 7. The clock
generator is constructed around T1 and
T25, and the only difference between
this and the basic astable circuit is the
inclusion of diodes DI and D2 and
resistors R31 and R32, which help to
speed up the leading edge of the rec-
tangular waveform. The monostables
are formed from transistors T2 to TIO,
and it will be noted that these are in
fact only 'half monostable' circuits, in
fact simply the pulse width determining
portion of the circuit. Two of the NOR -
 slotless model car track
5-36 - elektor may 1977
gates are made up from transistors T11
to T20, while the third NOR -gate
comprises T21 and T22. The 400µs
pulse -stretching monostable consists of
T23 and T24, and the MPX and MPX
signals are available at the collectors of
T23 and T24 respectively. Capacitors
C 1 to C25 are included to reduce the
susceptibility of the clock generator,
monostables and NOR -gates to spurious
pulses, which could upset the timing
sequence. Note that these capacitors
should be ceramic types. All other
non -electrolytic types should be MKM
polycarbonate or other good quality
components, and resistors should have
a tolerance of 5%.
The transistors used in the encoder are
not discrete devices but are contained in
five CA 3086 transistor arrays. Since
the encoder is intended to be universal
in application it has been made possible
to use any or all of channels 4 . .. 9
for switched functions instead of pro-
portional controls by including single
pole change -over switches at points A
and B. Also, the IC pinning is laid out so
that if a maximum of only four chan-
nels is required for any application then
T6 to T10 (IC2), T16 to T20 (IC4) and
their associated components may be
omitted. However, for the slotless car
track all the components shown are
required to control up to four cars, and
resistors R15 to R21 should be soldered
to position A on the p.c. board.
The encoder is provided with a simple
stabilised supply to enable it to operate
from an unregulated DC voltage from
8 V to 12 V.

Decoder principle
Referring now to figure 8, it is apparent
that the decoder circuit should be
capable of extracting from the multi-
plexed signal the original control wave-
forms K1 to K9 so that these can be fed
to the appropriate servo channels 1 to 9.
This can be accomplished by a simple
decade counter having ten outputs. If
the counter is initially at zero then on
the leading edge of the first (K1) pulse
in the multiplex signal the counter will
clock and output 1 will go high until

Figure 8. Showing the principle of the

decoder.

Figure 9. Practical circuit of a decoder using

a CMOS counter IC.

Figure 10. Showing how the joystick control

potentiometers are connected to the multi-
plex encoder. One joystick control is required
for each car - four in all.

Figure 11. Circuit of the infra -red transmitter,

which was described last month.
 slotless model car track elektor may 1977 - 5-37

the leading edge of the second (K2)

pulse, when output 2 will go high until
canoes
the third pulse and so on. From figure 8
RCA On it can be seen that the pulses thus
obtained at the counter output corre-
spond to the original control signals
generated by the monostables in the
encoder.
However, some means must be found of
ensuring that the counter is in the reset
state at the beginning of each pulse
sequence, otherwise the pulses might be
wrongly decoded and fed to the wrong
servo channels.
Fortunately this is easily achieved by
connecting the MPX signal to the reset
input of the counter via an integrating
network with a switch S connected
across the integrator capacitor. During
the positive periods of the pulse
sequence the integrator capacitor will
charge, but if switch S is closed during
the spaces in the pulse waveform then C
will be discharged and the integrator
output voltage will not reach the
threshold level of the counter reset
input. However, during the space at the
end of the pulse sequence C will charge
until the voltage on it exceeds the reset
threshold level, and the counter will
reset ready for the next pulse sequence.
Figure 9 shows a practical circuit for a
decoder operating on this principle.
Here the function of the switch is
performed by diode DI. While the input
signal is high the capacitor can charge
through resistor R, but if the input goes
low C will discharge rapidly through Di.
The decoder circuit will be discussed in
more detail in the next article in this
series.

Encoder - p.c. layout

A printed circuit pattern and com-
ponent layout for the multiplex encoder
are given in figure 12. Figure 10 shows
the connections made to the joystick
controls, one of which is required for
each car. For example, the first joystick
a
controls car number one and comprises
P1 which controls steering and P2 which
controls speed. The second joystick con-
it
trol comprises P3 and P4 which control
car number 2 and so on.

15V
1N4148

O
:4106
25V 8 .LD 241

32 BC 557B
1804nA
Ie 2x
BC 5578
1 N4148
"26
226
P3*
RI I
Rini WWI,.
MPX
44/ BCi451 BC141
BC 547 8 , BC 5478 C

DI 1N4148 1N4148 100n ICOn

2.1N4148 2 x 1N 4148 51,96- C)

see text
5-38 - elektor may 1977 slotless model car track

12

0.7 Oxo Oxl TP6

oo
vo.
oo
0,t9
oo oo oo
ys z
oo
0)(5
00
01.8
00
y2
y7 Z7:2 00 OX 6
ys z6 ys z y4 z
OW4
y3 z
Ox
7-1 u
C46 0 0
0
i p19 plEt
1
P1 P1 P14

ICI
110

icn
P12

2_0
P11
cG)
P10
44
I 21632
OrsApx

Cu
cr N I
TO To -L' W 0
cio C9 41 0--0-0---0 C5 C4 C3
0-0-00110 011420° 8 OBOE 08,2,0110 011"114"
A
BOB 013 A AA A
a

01:11
-6- WIICK4ics
C15
C20 C1B C17

C21 C19 c,97,4

D

Figure 12. Printed circuit board and corn

ponent layout for the multiplex encoder.

Parts list for figures 7 and 12. Capacitors:

Figure 13, Printed circuit board and com-
C1 ... C25 = 1 n ceramic ponent layout for the infra -red transmitter.
Resistors: C26,C27,C28,C29,C30,C31,
R1,R22,R23,R26,R28,R29, C32,C33,C34,C35,C40,C42,
R30,R31,R32,R34,R36 = 4k7 C45,C46 = 100 n MKM
R2,R3,R4,R5,R6,R7,R8, C36,C37,C38 = 10 n MKM
R9,R10,R11 = 47 k C39 = see text
R24,R25 = 10 k C41,C44 = 10 µ/22 V tantalum
R12,R13,R14,R15,R16,R17, C43 = 47 g/22 V tantalum
R18,R19,R20,R21 = 33 k
R27 = 68 k
R33 = 1 k Semiconductors:
R35 = 100 k T1 ... T25 = 5x CA 3086
P1 ... P8 = 5 k joystick controls transistor -array
P9 = 5 k lin or 4k7 fixed (see text) T26 = BC 547 B
P10,P11,P12,P13,P14,P15, D1 ... D4 = 1N4148
P16,P17,P18,P19 = 50 k (47 k) preset D5 = 6V8/400 mW zener
slotless model car track elektor may 1977 - 5-39

13

Modifications to
Additions to
Improvements on
Corrections in
Circuits published in Elektor

LED VU/PPM
April 1977, E24, p. 4-32. It was perhaps
not made sufficiently clear that the
display itself (figure 5) works off an
asymmetrical power supply (+15 V). No
negative (-15 V) supply is required:
points '0' and on the display board
(figure 7) are both connected to supply
common. The rectifier section (figures 2
and 6) does require a symmetrical
(+1-15 V) supply, of course.
Speech shifter
July/August 1976, E15/16, p. 742. In
some cases the TBA 120 has been found
to be so asymmetric that the oscillator
signal is audible at the output. The sol-
ution in this case is to add a 22 k or
If the unit is used to control four cars 25 k preset potentiometer between
in this way, P9 will be redundant. If this pins 7 and 9 of the IC; the slider is con-
Parts list for figures 11 and 13. nected to supply common through a
channel is not used at all, P9 can be 10 k resistor.
Resistors: replaced by a fixed resistor between
R1 = 18 k points X9 and Z9.
R2 = 10 k Testing and fine adjustment of the
R3,R6,R10,R11* = 2k2 multiplex encoder will be given in a
R4,R5 = 5k6 later article.
R7 =180 St
R8 = 1k8
R9 = 470 n Infra -red transmitter
R12,R13* = 1118
R14 = 10011 Finally, the p.c. board and component
R15 = 22 k layout for the .infra -red transmitter, the
P1,P2,P3" = 5 k preset circuit of which was given last month, is
given in figure 13. The circuit itself is
Capacitors: reproduced in figure 11. To adjust the
C1 = 470 p/25 V infra -red transmitter, first set P2 and P3
C2 = 100 n MKM to their central positions and allow the
C3,C4 = 1 n MKM circuit to warm up for about ten 9134

C5,C6" = 560 p ceramic minutes. Insert an ammeter between

C7 = 4)17/25 V
Cx = 100 n MKM (2x)
points A and B and adjust P2 to give a
T6 collector current of about 180 mA.
Semiconductors: Repeat this procedure with P3 for T7
T1,T4 = BC 547 B (meter connected between points A'
T2,T3,T5 = BC 557 B and B').
T6,T7" = BC 141 With some specimens of the LD 241
D1 ... D8 = 1N4148 LED it may not be possible to reduce
D9 ... D16 = LD 241 the collector current to 180 mA. In this
D17" ... D24" = LD 241 case R10 and R 11 should be increased
to 3k3. Precision timebase
Miscellaneous: Microprocessors
2 x heatsinks TO 5 type 'Wireless' intercom
(To be continued.)
NB. = see text Elektor SUMMER
CIRCUITS issue
5-40 - elektor may 1977 op -amp frequency compensation ... the why and the how

amp ftequene9 local feedback and biassed in class B).

The rolloff time -constant is normally
inserted between the first and second
stages or as a 'Miller' integration net-

c nn pencation work in the second stage itself.

It is not difficult to see that an op -amp
with the figure 1 dashed
obtained by a 'slow' second stage, will
response,

the whg have its input stage driven progressively

harder above f1, due to the failing
feedback (6 dB/octave above f1, 12 dB/

and the h®w

octave above f2).
There is a distinct danger that rapidly -
changing high -amplitude signals
cause the input stage to momentarily
saturate, at the steepest part of the
will

waveform - usually the zero -crossing.

This results in bursts of gross distortion
- in audio amplifiers - known as
Transient Intermodulation or TIM (see
Equin part 1, Elektor 12, April 1976,
When an operational amplifier is used in a p. 448).
The solution to this problem is to insert
negative feedback circuit its frequency response the compensation network at the ampli-
fier input. Figure 3a shows how this is
requires 'compensation', a high -frequency rolloff done for a non -inverting amplifier and
that may be 'internal' or 'external'. With figure 3h gives the inverting circuit. The
figure also gives rules for determining
inadequate compensation the circuit will usually the resistor and capacitor values
required. Some op -amps will misbehave
misbehave or even oscillate. This article will with nothing connected to their 'com-
pensation' pins; it is not always immedi-
explain the reasons for frequency compensation, ately apparent why - so that no general
rule can be given. A trick that usually
describe the usual 'simple' approach and then works is to insert a series RC -pair that
show how an 'external' type can be fitted with reduces the open -loop gain by 6 dB or
so, in a step, at some frequency above
an improved compensation -arrangement. the highest input but well below fl, at
the usual compensation position.
The latter approach results in a circuit that The insertion of the compensation
ahead of the input stage removes the
responds far better to large and fast excursions cause of slew -rate limiting and TIM; the
drive level of the input stage proper no
of the input signal. longer rises with frequency during the
sloping part of the open -loop response.
Why does an operational amplifier need at 6 dB/octave. A step -network that There is however a price to be paid,
compensation? The story starts with the 'flattens' again at f1 (drawn curve in quite apart from the extra mess around
observation that parasitic capacitances figure 2) is the standard trick. It will the input pins. Noise from the input
in the IC itself cause the 'open -loop' result in the dashed curve of figure 1. stage is no longer attenuated by the
(i.e. without -feedback) response of the The situation is in fact that the 'loop compensation network - it receives the
device to roll off more or less sharply gain' falls below the amount that would full open -loop gain up to f1. The kind
above a certain high frequency. This is enable oscillation, if the feedback were of circuit in which TIM is a problem
illustrated by the drawn line in figure 1 to become positive, at a point where (high level) is however not usually so
- the 'uncompensated' response. The there is still 90° phase -margin. critical in respect of noise. Furthermore,
actual curve is bounded by asymptotes, Note that many integrated op -amps the low source impedance at higher
6 dB/octave (20 dB/decade) above have their frequency compensation frequencies 'seen' by the input stage will
12 dB/octave above the second turnover built-in. An internal capacitor then tend to reduce its noise level anyway. N
point f2 and even 18 dB/octave above displaces one of the 'stray' rolloffs so
f3 in cases where there is a third turn- far downward in frequency that it
over. The open -loop gain is constant dominates in the response, automati-
from DC to f1 (the real curve 3 dB cally providing the figure 1 dashed
'down' at f1), equal to the value A01. curve. Perhaps the best known example Figure 1. The drawn line shows the op -amp's
of this is the '741'. frequency response without 'compensation'.
Figure 1 also shows the desired gain -
The dashed line shows the compensated or
with -feedback -operating, Ad (in deci- Those op -amps that are intended for use
rolled -off response. At its point of inter-
bels). If the slope of the open -loop with external compensation are supplied section with the horizontal dotted line
response at the intersection with the with data on how this should be done, through Aci (the so-called 'closed -loop gain',
horizontal through Ad exceeds 12 dB/ given the required values of closed -loop i.e. the amplication obtained with feedback
octave, the actual feedback will start to gain, phase -margin etc. For most appli- operating), this response curve slopes at
become positive, as the total phase -shift cations the instructions err on the 'safe' 6 dB/octave (20 dB/decade) - and the
will have exceeded 180°. With the side. system is unconditionally stable.
values assumed in the figure the op -amp That concludes the review of the basics
of frequency -compensation. It is now Figure 2. A so-called 'step -network' compen-
would certainly be in business for itself! sation will cause the drawn line in figure 1 to
The only way to 'dump' enough open time to take a closer look - preferably follow the 'compensated' response curve
loop gain before the phase -shift in the inside the IC! It will be convenient to shown dashed in figure 1.
IC exceeds 180° is to provide HF assume the usual op -amp circuit of
rolloff, starting early enough - so that differential input stage, second stage Figure 3. Basic circuit and 'design rules' for
the intersection between the open -loop with gain and some form of wideband improved compensation of a non -inverting (a)
and closed -loop response curves occurs unity -gain output stage (usually with and an inverting amplifier (b).
op -amp frequency compensation ... the why and the how elektor may 1977 - 5-41

I m©e
A
id814
AOL,
eon air
96(10rd:or
ACL

f0 fl
2 f3 log f
9782 -1
A.J.B.M. Peters

2 Readers may
remember that a
idAB,1

0
design for an
automatic callsign
generator was
featured in Elektor
20 logACL
AOL
a little over a year
ago (Elektor 11,
fo
log f
February 1976).
?7i32
This new design
2

offers the same

facilities with
considerably simpler
circuitry, though at
the expense of a
slightly more
complicated
programming
ACL
1
procedure.
f AOL
ACL

It may be remembered that the previous

3b design for a callsign generator used
R2 CMOS shift registers whose outputs
were connected via diodes to two
1

If°
2(R, ,$R2*.e1CLI
AOL '01 RldR2
programming lines. This made for very
simple programming but made the cir-
cuit fairly complicated. The program-
ming of the new design is accomplished
by storing the callsign in a 100 bit read
only memory consisting of a diode
matrix. A dot is stored in the matrix by
inserting one diode in the required
9782-30 position. A dash, which has a duration
ACL = -
equal to three dots, requires three
diodes. A space within a character is of
AOL one dot duration and occupies one
ro -JACO blank space (no diode) in the matrix. A
space between letters is the same
duration as a dash and thus occupies
5-42 - elektor may 1977 morse call sign generator

three blank spaces in the matrix. To counter counts the column outputs 0 to diodes, the exact quantity depending on
generate the callsign the contents of 9 of IC1 go low in turn. Whenever a the actual callsign.
the matrix are read out row by row. position is reached where a diode is To programme the generator, start with
100 bits may seem excessive, but it connected from a column output to row 'a' of the matrix. Leave position
is possible for a single figure (digit cb) row `a' then the second input of NI is `AY blank as this is the rest position.
to occupy 19 spaces in the matrix. This, pulled low and the output goes high. Work along row 'a' and connect a diode
combined with long European call signs, At the end of the first row the D output for each dot with its anode to row 'a'
soon uses up the spaces in the memory. of IC3 will go low, causing IC4 to and its cathode to the particular column
British callsigns of 4 or 5 characters will, advance one step. One input of N2 will you have reached. For a dash a diode
of course, not use as much of the now be low and as IC3 counts from 0 to must be connected to each of three
memory capacity. 9 again the information on row 'b' will successive columns. For a space the
The complete circuit of the callsign be read out via N2. This is repeated appropriate number of columns must be
generator is given in figure 1. The diode until all the rows of the matrix have left blank. When the end of row 'a' is
matrix is in the top left corner of the been read out. reached then return to the start of
diagram. Readout is accomplished by The diodes connected to the outputs of row 'b' and continue.
addressing the rows and columns of NI to N10 form an OR gate to route The callsign example shown in the dia-
the matrix using two 7490 decade the information to the inputs of an gram is the author's, DE PAOARR,
counters and 7442 decoders. audio tone generator SI and a relay which in morse is
The rate at which the callsign is repeated driver Tl. When a dot or dash is present . . . . . .

is determined by ICS, a 555 timer the tone generator is activated and the
connected as a monostable multi - relay is energised. During spaces This is laid out in the matrix as follows:
vibrator. Assume that initially the between character elements there is no
monostable is in the triggered condition. tone and the relay drops out. The tone column
The output, pin 3, is high, so both the generator may be used to modulate a
0 2 3 4 5 6 7 8 9
counters IC3 and IC4 are held in the transmitter or the relay may be used for 1

reset condition. Output 0 of ICI CW keying. row a

(column 0) and output 0 of IC2 (row When a count of 100 has been reached b -0-
a) are thus both low and all other all the rows of the matrix will have been S
outputs are high. One input of NOR read out. The D output of 1C4 goes low d -0-
gate NI is low and the other is high, on count 100. This negative -going edge
e -111-1110
since no diode is connected in position is differentiated by the 1 n capacitor
`a0' in the matrix as this is the rest and 10 k resistor to produce a short *111-4141"
position. The output of NI is thus low. pulse which triggers the 555, inhibiting 9 -0- -0-
When the monostable (105) resets, the the counters until the 555 resets again. h
reset inputs of IC3 and IC4 go low and The repetition rate of the callsign can be
.41-t-11111040.
IC3 begins to count pulses from the varied by means of P1.
clock generator built around S2. As the Programming requires a fair number of

Figure 1. Complete circuit of the callsign

generator. The desired callsign is stored in the
all diodes in matrix are DUG
diode matrix in the top left-hand corner.
10x DUG
3315q,5VI
680tD1OVI
0
a*. y5LED
1 N4148
I I
I I
10- MEM TUN

13

I ;0- 470 SI

47n
470 10k
1L.

?(it i(( Db. 0

10). 01
®5V
t 39%
4 10 2 3 4 5 6 7 9 10 11
1 Vl
2 3

7442 ICI
7 B 9 3

IC2
7442
4 5 6 6 9
AC)
!-C)
) 11.1

A B C A 9 41

15 14 13 12
DUS
15 14 13

IC5

12 9 12 9 11
It 555
---1330 A B C 8
BIN IC4 . S2 7413
IC3
7490 4 7490
A IN Ni . N10 =3X7402
4 AIN
9911) 9912) 9011) 11113121
5
9919 199171 ROID 90 21 L C)
10 7 2 3 6 3
11 21 2

T6V- OR
3

9695
music cleaner elektor may 1977 - 5-43

quency of 25 Hz was chosen, but again

mud@ elleanet this can easily be altered if so desired.

The ultimate slope of the filter is also
important. Some filters have a slope of
as little as 6 dB/octave. This means that,
to be effective in suppressing rumble or
noise the turnover point of the filter
must be within the frequency range of
the wanted signal leading to partial loss
of the signal or, to use another
metaphor, `throwing the baby out with
the hathwater'.

Choice of Response
To avoid this both the filters used in
the present design have an ultimate
slope of 18 dB/octave. There still
remains, however, the choice of filter
design. The magnitude/frequency re-
sponse of the so-called Butterworth
In response to popular demand, here, at last is type of filter is maximally flat in the
passband. The Chebishev filter attains
a treble and rumble filter that can be used in a sharper cutoff at the turnover point
virtually any hi-fi system to get rid of the snaps, than does the Butterworth, but at the
expense of a magnitude frequency
crackles, pops and (g)rumbles without getting response that is not flat in the passband.
However, neither of these types are
rid of half the music. concerned with phase distortion of the
signal, which is important when dealing
with complex waveforms such as music.
There has been much speculation about The treble filter in the present design To minimise phase distortion the phase
who first coined the phrase 'The wider has a choice of two cutoff points. A
you open the window, the more the cutoff point of 25 kHz is chosen to
muck blows in'. However, there is no prevent ultrasonic signals from reaching
disputing its truth with respect to audio the power amplifier, as these can lead
systems. The bandwidths of pickup to the (now) well-known transient
cartridges, amplifiers and loudspeakers intermodulation distortion (TIM). For
are now so great that the imperfections elimination of record and tape noise a
of records and turntables are often cutoff frequency of 10 kHz was chosen.
glaringly exposed, even by a relatively The circuit can easily be adapted to give Figure 1. Theoretical circuit of the third
inexpensive cartridge, amplifier and many other different cutoff frequencies order lowpass Bessel filter used for the treble
loudspeakers. according to personal taste, calculation filter.
Quite apart from the argument as to of the required component values being Figure 2. Theoretical circuit of the third
whether these phenomenal bandwidths a relatively simple matter. order highpass Bessel filter used for the
actually serve any useful purpose apart For the rumble filters a cutoff fre- rumble filter.
from churning up one's innards at the
bass end and annoying the neighbour's
(R7 resp. (R8 resp. (R9 resp.
dog at the treble end, there is obviously R7i R101 R8,0" R11) R9 R12)
a need to `chop off the top and bottom
end of the audio spectrum under certain
circumstances. Turntables, for example,
are not always as rumble -free as one
might wish, and being mechanical
devices tend to get worse with age.
Records, too are rarely perfect, suffer-
ing often from rumble and even more
frequently from surface noise, which
latter tends to become worse with wear.
Then, of course, one rr,.,st consider the
enthusiast who possesses treasured col-
lections of older records or non -noise -
processed tapes.

Design requirements
Unfortunately, few amplifiers, except
the most expensive ones, possess effec-
tive treble and bass filters, simply
because the requirements for such
filters are rarely properly formulated.
Firstly, the turnover (-3 dB) points of
the filters should be chosen carefully.
Many treble filters cut off at much too
low a frequency, leading to loss of part
of the wanted signal. Similarly, rumble
filters often cut off at too high a fre-
quency.
5-44 - elektor may 1977 music cleaner

shift p, produced by a filter should The configuration finally chosen was may be altered simply by changing the
vary linearly with frequency. This that using a voltage follower. This is values of the three resistors. e.g. if the
condition is fulfilled by a filter having easy to realise with low -noise audio resistors are halved the turnover fre-
a Bessel type of response. This produces transistors, and the overall gain of the quency is doubled, if they are doubled
minimum phase distortion, though the filter within the passband is unity, the frequency is halved.
cutoff at the turnover point is not as which means that it can be inserted For the highpass filter the three capaci-
sharp as either the Butterworth or in any amplification chain without tor values are equal and the resistor
Chebishev filters. Three -pole Bessel affecting the gain. values are calculated from the equations
filters were thus chosen for this design. Figure 1 shows the theoretical circuit given. In this case halving the capacitor
Having decided on the best type of of the lowpass (treble) filter, while value will double the turnover fre-
filter response one is then faced with figure 2 shows the theoretical circuit quency, and doubling them will halve it.
the problem of practical realisation. of the highpass (rumble) filter. The The equations for obtaining the filter
There are various realisations, some three resistors in the lowpass filter are parameters are reproduced from
of which require infinite open -loop gain identical, and the values of the three `Electronics' August 18th 1969.
from the active part of the circuit. capacitors required for a given turnover
frequency can be calculated from the Practical Circuit
While these requirements can be ap-
proximated using IC operational ampli- three equations given. Once the capaci- To obtain a treble and rumble filter in
fiers, such ICs are not ideal from the tor values for a given frequency have the same unit the two circuits of figures
point of view of distortion and noise. been found then the turnover frequency 1 and 2 are simply cascaded. The com-
plete practical circuit of one channel of
the filter is given in figure 3. Here the
theoretical voltage followers have been
replaced by transistors connected in a
`super emitter -follower' configuration.
The rumble filter is built around T1 and
T2. R3 and R4 perform a dual function,
forming part of the rumble filter but
also providing the base bias for TI. The
effective value of these resistors as far
as the filter is concerned is R3 paral-
R3 X R4
lelled with R4 or
R3 + R4'
The treble filter is built around T3 and
T4. Selection of the turnover point is
carried out by a three way switch. With
the switch open only R7, R8 and R9
are in circuit, and the turnover point is
about 10 kHz. With the switch closed
these resistors are connected in parallel
with R10, R11 and R12 respectively,
and the turnover point goes up to about
25 kHz. Capacitors C4 and C8 are
included to guard against the possibility
of r.f. instability.
The circuit has a high input impedance
and low output impedance, so that it
may be connected virtually anywhere
in an audio system. For use with an
existing amplifier the unit may be
connected to the tape socket if this is
music cleaner elektor may 1977 - 5-45

Parts list to figures 3 and 4

As the p.c. layout is for a stereo
system all components except S1
are duplicated.

Resistors:
R1 = 100 k
R2 = 68 k
R3,R4 = 820 k
R5,R13 = 5k6
R6,R14 = 2k2
R7,R8,R9 = 3k9 (see text)
R10,R11,R12 = 2k7 (see text)
R15 = 47 k

Capacitors:
C1,C2,C3 = 68 n
C4,C8 = 4n7
C5 = 3n9
C6 = 5n6
C7 = 1 n
C9 = 22 µ/35 ... 40 V
C10 = 10 µ/35 . 40 V
C11 = 100 n
01-. to
Semiconductors:
T1,T3 = BC547B,BC107B or
equivalent. 01-2 lo
T2,T4 = BC557B,BC1 57B or pi -6' I°
equivalent.
04Fulcilo
Miscellaneous:
S1 = six pole on/off with
normally closed contacts.

°01043
7'
Figure 3.Complete circuit of the rumble 04 R14' 1.0
and treble filter.
Figure 4. Printed circuit board and com-
()Hi Pa
ponent layout for the filter unit. 00115 143

Figure 5. Gain/frequency response curve of

the filter unit.
not already in use. The input to the
filter is taken from the 'tape' output
of the amplifier and the output from
the filter is fed back to the 'tape' input.
Depressing the 'Tape Monitor' button
will then bring the filter into circuit.
Distortion introduced by the filter is
extremely low so that it may be used
with the highest quality systems.
Printed Circuit Board
A printed circuit board and component
layout for the stereo filter are given in
figure 4. Provision is made on the p.c.
hoard for mounting a pushbutton
switch for S1. Alternatively Si may
be mounted remote from the board
provided the leads are not more than
a few centimetres long. If only one
turnover point is required for the treble
filter then S1 and the associated
resistors may be omitted.
Response Curve
The gain/frequency response curve of
the filter unit is shown in figure 5. As
mentioned earlier, with S1 depressed
(open) the treble filter turnover point
is around 10 kHz, while with SI closed
it is around 25 kHz.
5-46 - elektor may 1977 active loudspeaker -crossover filters (1)

human being already knows how to do.

active totudopealicer A multiway system is necessarily more

complicated and more expensive to
produce than a single -driver system.

croot.-riover kiittero That is a clear disadvantage. There is

however a second objection to the
multiway approach - a more funda-
mental objection: how does one tackle
the fact that frequencies near the
crossover point are radiated by both
drivers? The two radiating diaphragms
cannot he at the same position in space
- although they often can be spaced
quite closely - so that 'interferences'
between the two radiated waves can
cause irregularities in the response
characteristics and in the radiation
pattern of the system.
`Dividing' is one thing; 'ruling' is quite
another ...
Most of the interference effects can be
Few things can so hold the attention of the avoided when the two frequency -
adjacent drivers are mounted concentri-
serious audiophile as do loudspeakers. This cally - one within the other. This is
applies with particular strength to those whose usually no problem, since an optimal
tweeter can be made smaller than a
fingers always have the experimenter's itch - so woofer. The past has known designs -
many of them still very popular - in
that they cannot or will not without reserve which a tweeter of one kind or another
has been built into a woofer (or, more
accept somebody else's idea of a loudspeaker accurately, a woofer -midrange) cone -
system. This can lead to the expenditure of loudspeaker. The crossover can he
mechanical in nature (as in the 'good
considerable sums, if only on wooden panels, old' Philips 9710M), or a more advanced
twin -driver -plus -electrical -crossover sys-
and it will sometimes also lead to frayed tempers tem can be employed (as in the famous
Tannoy Monitor Gold and certain
at home . . . Goodmans and Isophon units).
One of the ways of sinking cash into an existing
system is to replace the 'passive' separating Passive or active?
Having decided that a good loudspeaker
('crossover') filters by 'active' types. This of system, at the present state of the art, is
going to need at least one crossover
course involves the provision of a separate power filter, we have to decide whether this
amplifier for every driver in the system. filter should be a 'passive' or 'active'
design. (For our purposes, an 'active'
This article on Active Crossover Filters (ACF's) filters is one in which the inductors have
been eliminated by the application of
will describe a universal filter circuit, capable of capacitors and amplifiers).
Figure la illustrates a typical passive -
producing a vast number of filter characteristics. filter three-way system. The passive
filter is built up with inductors,
capacitors and any matching networks
that may he necessary (e.g. to reduce
High -quality loudspeaker systems are proper), the system is said to have a the drive to a too -sensitive tweeter).
invariably designed on the basis of passive filter. Figure 1illustrates a Figure 1 b illustrates the bare bones of a
`divide and rule' principles. [he typical three-way system. The low -to - three-way passive filter.
incoming audio spectrum is split up into midrange crossover frequency is f, and One difficulty is immediately apparent.
two, three or even four sub -spectra, the midrange -to -high crossover occurs at The woofer section requires an inductor
each of which is then passed to a loud- f2 .The representatives of the animal in series with the driver voice -coil. The
speaker specially designed for that kingdom shown have had their typical considerable inductance involved means
particular frequency range. The change- calls `borrowed' to provide a classifi- that there will be power loss in the
over from one loudspeaker to the next cation of the drivers into the categories copper -resistance of a many -turn air -
higher in frequency range is ac- low -range (woofer) midrange (squawker) cored coil, or else that there will be
complished by a complementary and high -range (tweeter). distortion due to the non -linearity of a
filter -pair whose roll -off response -flanks The big idea behind the multiway low -loss coil that has a ferromagnetic
`cross over' each other at a point some approach is the fact that an optimally - core. Neither of these effects should
decibels below the `full power' level. designed 'woofer' is - for basic design however be viewed out of proportion:
The filter -pairs are therefore called reasons -- a sub -optimal loudspeaker at the often -cited effect of the series -
'crossover filters'. higher frequencies. This does not mean resistance on the woofer's electrical
A loudspeaker system that uses such that a 'new' design method may not damping is completely swamped by the
filters is usually called a 'multiway' someday produce a first-class full -range effect that the voice -coil resistance has
system. driver; it simply hasn't been done yet. - and one can design iron -core inductors
When the filter sections are inserted The problems to be faced are quite with a level of distortion that is
between the single power amplifier and formidable - and a computer is only insignificant compared to that of the
the individual 'drivers' (i.e. loudspeakers useful to quickly do the sums that a actual driver.
 -4 -14 -

active loudspeaker -crossover filters (11 elektor may 1977 - 5-47

Another source of difficulties is more

la awkward to eliminate. Normal electrical
wave -filters assume a pure -resistance
load -termination. When you connect a
loudspeaker to such a filter the final
characteristic may not be quite what
you intended - it may even be wildly
off. The trick of connecting an RC
network across the speaker terminals to
compensate the high -frequency rise in
impedance (due to the coil's inductance)
certainly works and should be better
known; but the fun really begins when
the speaker impedance contains signifi-
cant components 'reflected' from the
9786 -la mechanical 'circuit'. That usually
happens in the neighbourhood of the
driver's fundamental resonance; it can
be a very expensive nuisance in the case
of midrange and tweeter units that have
lb a resonance (as is usual) at or just below
their high-pass crossover frequency.
T27
Now, a well -designed commercial
SP1032 `passive filter system' will invariably
work very well - but that success is due
to a combination of design experience
IrrH and available facilities beyond the reach
of the 'do it yourself' audiophile.
r -X-) 8110 Although it would be possible to say a
SP1003
great deal more about passive filter
arrangements and matching networks,
5rrH
this article is supposed to be about
-/-1,10M- active arrangements. Having implied,
P.K.) above, that the amateur is better off
j 8139 tackling his problem with an active
t"\_, system, we must now try to explain
9786 lb how.

Active Crossover Filters

Figure I c shows the block diagram of a
three-way active (`electronic') crossover
filter. It is immediately clear that each
of the loudspeakers requires its own
power amplifier. This need not be so
expensive as one might think, since the
total power required (and hence the
amount of mains transformer, reservoir
capacitor and heat sink) is not increased
by subdividing the amplifier. As a rule,
the woofer will need the most powerful
amplifier (perhaps 50 . . 70% of the
total), with the midrange unit handling
perhaps two-thirds of the remainder.
Much will obviously depend on the
individual drivers used. When drivers are
obtainable with varying rated im-
pedances, the power distribution over
the output stages can be achieved by
using a single supply voltage together
with a low -impedance woofer (say
4 ohm), a mid -range unit of higher

'Id

Figure la. Block diagram of a three-way

system with passive crossover filter.

Figure lb. As an example: the KEF type

DN 12 SP 1004 three-way passive filter.

Figure lc. Block diagram of an active -filter

three-way system.

Figure ld. An active -filter two-way system.

5-48 - elektor may 1977 active loudspeaker -crossover filters 11)

impedance (say 8 ohm) and a tweeter of

still higher impedance (15 ohm).
A major advantage of the active -filter
approach is the ease with which sensi-
tivity differences between the drivers
can he eliminated. In figure lc this is
accomplished by adjustment of the
presets P1 , P2 and P3. Figure 1 d gives a
simpler two-way circuit, suitable for use
with smaller diameter woofers that are
also well-behaved throughout the mid -
frequency range. Still another possibility
is shown in figure le, a 'hybrid' three-
way system. In this case the woofer to
midrange crossover is done with an
active filter and two power amplifiers;
the frequency ranges for the midrange
and tweeter drivers are however
separated by a passive filter set.
What are the other advantages of the
active filter approach?
If - the design is far more flexible; a
change of crossover frequency or
drive level can be quickly and
conveniently achieved by changing
f2 one or two R's and C's or adjusting a
preset potentiometer.
- there is no complication in the filter
design caused by the awkward
termination (the loudspeaker
impedance).
- it is relatively simple to produce
complicated filter characteristics
whenever this is thought desirable or
1 2 34 5 6 78 necessary.
11211181 1181 1121 112) 118) (18) U21
9786 - since the power amplifiers will usually
be installed in the loudspeaker cabi-
net, the individual drivers can be pro-
tected from overload by suitable
choice of the power rating of the
amplifier concerned.

2a The filter circuits

Figure If shows a set of filter charac-
teristics, as would be required for a
R17 021 R18 021
8161121
R 5 011 R 6 111 1 R 7 1f11 three-way system. The frequencies fl
and f2 are the `-3 dB' points, at which
the response curves of a complementary
1 392
Ca2nfR filter -pair actually 'cross over' each
other. Half of the power at a crossover
3,546 frequency is transmitted through each
=Cb 2nfR
filter of the pair. For a three-way
0,202
CC - 27f R 9786 2a system fi will frequently lie between
C 31f1) C 4 1111 C 5)111 300 and 600 Hz (sometimes as low as
C14 021
C12 1(2) C13 1(21
100 Hz, or as high as 800 Hz). The
other crossover will then usually be
found between 2 kHz and 8 kHz -
typically near to 5 kHz. The single
crossover in a two-way system is usually
between 1 kHz and 3 kHz (typically
around 2 kHz).
2b The slope of the various filters well into
their respective 'stop -bands' is a
multiple of 6 dB/octave (i.e.
C 8111) C 91f1) C101111
C17 021 C18 1(21 C19 1(2) 20 dB/decade). The figure if curves are
drawn for 12 dB/octave (1,4,5,8) and
00A for 18 dB/octave (2,3,6,7). If we assume
n 0 718
that either slope may be used for each
ria = 2 -;71 -c - of the four filters, then there are sixteen
possibilities for a three-way filter. It is
n, b =0,282
ri not always desirable to make the filters
4,95 of a crossover -pair with the same slope
Rc = grit
R10 011 R11 011 812 H13 011
9786.26
- a so-called asymmetrical crossover
821 (f2) R22 1121 R23 r R26 021 may be needed when the response of
one of the loudspeakers is not flat
through the crossover point. Table 1

lists the possibilities.

active loudspeaker -crossover filters (1) elektor may 1977 - 5-49

The last four alternatives apply to two-

way systems. We will refer in this article 3a
to the single crossover as fi
An electric wave -filter is characterised
C 5 (f = f1)
not only by the 'ultimate slope' of the C14 If = f21
rolloff curve, well into the 'stop band'
R17 If = f2) R18 if = f2)
but also by the 'sharpness of transition' Fl 6 (f = fl) R 7 lf = VI)
between the pass -band and the stop - III 11111
band. A number of Famous Names are
associated with a classification of filters
C 4 If =f1)
into categories with increasing sharpness C13 If = f2)
(once again: note the distinction
between sharpness and steepness). Cb1
= 414
72- tr3
Almost all loudspeaker crossover filters 3786 3a
are of the Butterworth 'maximally flat 0707
amplitude' type. We will therefore
illustrate the workings of the practical
circuits by Butterworth responses.
When the 'pass -band' is defined as the
frequency range up to the -3 dB point
(low-pass) or from the -3 dB point up-
(high-pass), then Butterworth
wards
gives the lowest possible 'pass -band 3b
attenuation' that can be obtained with-
out allowing 'ripples'. R12 1 R13 (f=f1)
The figures 2, 3 and 4 give the design C 9 (f - fl) C10 If - f1) R23 R26 = f2)
information for Butterworth low-pass C18 (f - f21 C19 If -- f21
filters (`a' figures) and Butterworth
high-pass
ultimate
filters
slopes
(`b'
of
figures), for
18 dB/octave
(figure 2), 12 dB/octave (figure 3) and
6 dB/octave (figure 4). The two sets of
component numbers refer to the two
00
R11 If - fl)
R22 (f = f2)
I

t
different crossovers. We will come back 0,707
Rb 27fC
to this when referring to the parts list.
The active element in the circuits of 1,414
c 27fC
figures 2, 3 and 4 is a voltage follower.
The best known AC voltage follower is
the so-called 'emitter follower'. Since
a voltage gain of unity can only be
closely approximated by an amplifier
with extremely high current gain, the
total circuit diagram of figure 5 shows
'super emitter followers' using two
transistors each. The derivation of the
component values always assumes the
use of an ideal voltage follower; any
attempt to 'make allowances' is fraught
with great uncertainties - and the
assumption that a one -transistor
follower is ideal is just too optimistic!
This is not the place to go into the

Figure le. A hybrid active/passive three-way

system.

Figure if. A few frequency -response plots,

with slopes of 12 and 18 dB/octave and one
or two crossovers, as an aid to interpretation
of table 1.

Figure 2. Circuit diagram and values for a

Butterworth low-pass (a) and high-pass (b)
18 dB/octave filter.

Figure 3. Circuit diagram and values for a

Butterworth low-pass (a) and high-pass (b)
12 dB/octave filter.

Figure 4. Circuit diagram and values for a

low-pass (a) and high-pass (b) 6 dB/octave
filter.
 acme mu esmaxer-crossover filters let .

5-50 - elektor mey 1977

available. Note that the idea was to find three-way systems is not inevitably one
details of the derivation of design
ormulae. something to play with! of cost,with three-way always better if
One practical consequence of the There is one fundamental guideline, you can afford it. On the contrary
however. Loudspeaker are meant to be some of the best -sounding system
derivations must however be noted hem.
That is the fact that it is not always used for listening to music, not the around use a woofer -midrange unit plu
possihle to design filters in which all the other way round. if it sounds right, then a tweeter. These woofer -midrange unit
frequency -determining R's and C's have never mind what it looks like on paper. do however tend to need rather more
Assuming that o n's musical taste is than a simple closed -cabinet if they are
convenient values. We have chosen
reasonable, any discrepancy between to do a really good job at the deep -bass
circuiets with either three equal C's
(high pass) or three equal R's (low-pass), the theory and the actual result will end.
usually be due to an oversight or The frequencies and ultimate slopes of
the other components hopefully coming the crossover filters can be taken, at
fairly close to standard E12 values. incompleteness in the theory.
It will simplify this story if we introduce least as a starting point, from the
Filters with low kIll values (such as of the passive filter
two further 'boundary conditions'. Let parameters
Butterworth) will, fortunately, not recommended by the speaker manufac-
immediately go haywire when some of us assume that (1) we are going to do
the job properly - no skimping on turer. If one is combining speakers from
the components are a few percent out. various sources, then some experiment
That is not to say that a fusspot with parts - and (2) that the reader already
knows how to design his enclosure. may be +recess.), (greet fun!). There are
access to 1% R's and C's should not
The question that should be tackled one or two guidelineshere, more 'don'ts'
indulge a craving for 'precision' ... than 'do's'. In the first place, beware of
So much for the general aspects of first is the choice of the loudspeaker to
be used. This usually will involve a dig the 'power handling capacity' ratings of
active crossover filter design. It is n
time to try working out a specification. into the manufacturer's literature - or tweeters. It is in the nature of things
One way to tackle this problem is to use at least a good look into a distributor's that their smaller coil systems cannot
a check -list. catalogue. Unless one knows precisely handle the mive
ass amounts of Input
- Active filters only (figure lc or Id) or what one wants, it is a good idea to power that will not damage woofers.
select a combination recommended by The temptation to suppliers is to quote
hybrid (le)?
- Three-way or two-way? the manufacturer, replacing only the a high power rating for a tweeter in
inevitable passive filter by circuits combination with a specified high-pass
- Which speakers?
- How steep the filters? covered in this article. Information on filter. The 'power density' of normal
- Which amplifiers? how to construct special woofer music spectra certainly becomes
Do not try to find complete 'paper' enclosures, such as folded horns or significantly lower as the frequency
questions. A great 'transmission line' types, can often be increases; but this no longer applies
answers to these
found in the literature. when the amplifier is driven into
deal will depend on one's individual
The basic choice between two-way and distortion (accidentally or on purpose).
taste and on whatever happens to be
active loudspeaker -crossover filters (1) elektor may 1977 - 5-51

Figure 5. Complete circuit diagram of an

active filter set for two symmetrical
Table 1. 18 dB/octave crossovers (three-way).

The different possible combinations of symmetrical or asym-

metrical crossovers and 12 or 18 dB/octave slopes

filters slopes at filters slopes at

fl to be f2 to be combine from refer to
figure 1f figures
18 12 18 18 2, 4, 6 & 7
18 12 12 12 2, 4, 5 & 8
18 12 18 12 2, 4, 6 & 8
18 12 12 18 2, 4, 5 & 7
12 18 18 18 1, 3, 6 & 7
12 18 12 12 1, 3, 5 & 8
12 18 18 12 1, 3, 6 & 8
12 18 12 18 1, 3, 5 & 7
18 18 18 18 2, 3, 6 & 7 5 & 6*
18 18 12 12 2, 3, 5 & 8
18 18 18 12 2, 3, 6 & 8
18 18 12 18 2, 3, 5 & 7
12 12 18 18 1, 4, 6 & 7
12 12 12 12 1, 4, 5 & 8 7* & 8*
12 12 18 12 1, 4, 6 & 8
12 12 12 18 1, 4, 5 & 7
18 18 2 &3 9*&10*
12 12 1 & 4 11*&12*
12 18 1 & 3

18 12 2 &4

* Note: figures 6 to 12 will be given in part 2.

Put another way: (1) the high-pass filter The amplifiers on the choice of (high-pass) crossover
associated with a certain tweeter will frequency. This seems to indicate that
invariably have a `protection' function We come now to one of the great the amplifier's 'music power.' rating,
as well as its effect on the response and sources of endless discussion. How may together with the choice of crossover
(2) don't try to get a quart out of a pint watts need one provide for each loud- frequency, should be matched to the
pot! speaker? There are many ways of (higher) instantaneous rating of the
The other guideline worth mentioning looking at this question, depending on individual speaker. This applies literally
concerns the fact that a given loud- the kind of music you have in mind for to the midrange and to the tweeter; for
speaker invariably will have a frequency instance, or depending on which 'trade- the woofer something similar applies -
response extending far higher than the off' you prefer. but now with the 'box' design setting
recommended crossover low pass cutoff. We have already noted that the the high-pass cutoff frequency.
The response in the non -recommended (continuous) dissipation of which a Having taken a look at the limiting
range is however usually ragged or typical tweeter is capable will be less amounts of power that an amplifier
`peaky' due to the cone (or other than that of a mid -range and significantly should not be able to exceed, we still
diaphragm) 'breaking up' into patterns less than that of a woofer. That is have no answer to the real question:
of flexural resonance. This effect will simply a question of the physical size of how much do we need? The answer is,
impair the transient response. When a the respective `motors'. It would seem for normal domestic listening, 'surpris-
high-pass rolloff is recommended, quite obvious that the continuous -power ingly little'. Simply read off from the
apart from the input power consideration ratings of the associated amplifiers manufacturer's literature low much
given above, there may be a mechanical should reflect this fact. All one can input will produce about 96 dB SPL
limitation on the obtainable sound hope to achieve with some 'reserve (`sound pressure level) at 1 metre from
output in the non -recommended range. watts' is an increased risk of sometime the loudspeaker (usually specified for
This would apply in particular to dome - needing a 'reserve speaker'. There is a free -field -room measurement). It will
type tweeters and squawkers. bit more to it than this; but let us break usually prove that 10 or 20 Watts
The filter slope of 6 dB per octave is off at this point. already offers a very comfortable safety
rarely used, although there is consider- Every loudspeaker has a certain margin!
able evidence that a slow woofer - `instantaneous' power rating, referring So much for the design considerations.
midrange rolloff combined with a tc how much driving force it will handle Next month we will give circuits and
steeper tweeter slope can give excellent (quite apart from the dissipation printed circuit boards for 6-, 12- and
results. It is included here for involved) before some moving part hits 18 dB/octave filters for use in both 2 -
completeness sake, since 'asymmetrical' an end -stop. Since, at a given sound and 3- way systems.
crossover filter design really requires level, the diaphragm amplitude will be
access to acoustical measurement greatest at low frequency, the actual
facilities. useful instantaneous rating will depend
5-52 - elektor may 1977 albar mkll-a bfs

mounted, not the ones shown dotted.

allbar mlic1111 a Wm in II Alignment has been considerably

simplified by the addition of a head-
phone output. Components C14 and
R17 can be mounted on the board, and
any relatively high -impedance head-
phone or earpiece can be used.
The alignment procedure is as follows:
- connect the unit to a 12 V supply and
set P2 at minimum resistance (fully
clockwise).
move a hand to and fro in front of I
the transducer. This should produce a
rising and falling tone in the earpiece.
- set P1 for maximum level (loudness)
of this tone.
- now adjust the required sensitivity for
the unit with P2.
Two -transducer version
If the first version proves to be less
In November 1976 the circuit was described of sensitive than desired, or too temperature
dependent, it can be extended to a two -
an ultrasonic intruder alarm, the 'Albar'. By transducer version.
slightly mis-using the ultrasonic transducer it got The original circuit is modified
according to figure 2, to work as a '

away with using only one transducer as both receiver, and a transmitter is added
(figure 3). The modifications to the
transmitter and receiver. basic circuit are marked with asterisks.
Take particular note of the fact that T2
Unfortunately, this system has proved only is replaced by a PNP type.
80%-90% reliable, i.e. 10%-20% of the units did Construction and alignment
not work as well as they should. As briefly The components shown dotted on the
component layout for the main board
noted last month, there are two possible (figure 4) are used in the receiver circuit.
This means that where only one possi-
solutions to the problem. Both circuits are bility is shown, it is valid for both
circuits; where a component is shown in
discussed in greater detail here. full and another is shown dotted, the
dotted version is required here. Note
Lab tests have shown that the problems the original Albar circuit (Elektor 19, however that several of the other
arise from the wide variation in the November 1976, p. 1112, figure 6) in component values are also changed, as
characteristics of the transducers. several details: marked in the circuit and shown in the
Specifically, in the original version the The numbering of the components has parts list.
transducer was used as the frequency - has been changed. The transmitter and receiver hoards
determining element for the transmitter The zener diode has been changed with their respective transducers should
oscillator. In practice, most transducers from 9V1 to 10 V, since 10 volt be mounted in the same box, with the
have several possible resonance modes zeners are more readily available. transducers facing out in the same
some in the desired frequency band One of the T2 emitter resistors has direction. It is advisable to mount the
and some outside. To make matters been replaced by a fixed 2.2 mH transducers in foam rubber rings, to
worse, they are temperature dependent. inductor. avoid acoustical coupling through the
The result of all this is that the sensi- A supply decoupling capacitor (C13) box. The distance between them should
tivity can vary between a few inches and has been added. be approximately 2 in. (5 cm).
15 to 20 ft. The latter value is what we In the original circuit, the alarm Point X on the transmitter output is
measured on our original prototypes . latched on until the reset button was connected to the receiver 'X' input, and
There are two solutions for those operated. In the new circuit the alarm the two units are run off a common
constructors who have built a unit with resets automatically a few seconds 12 V supply.
a sensitivity closer to the former value. after it ceases to detect movement (it The headphone output is once again
The first is a minor modification to the operates continuously as long as a used for alignment. Since this unit is
original circuit, still using a single moving object is within range). The more sensitive than the single -transducer
transducer: the second is the Brute reset delay depends on the value of version, alignment may be difficult if
Force Solution if all else fails: it uses C12. If required, the original circuit there are any large reflecting surfaces
two transducers, one as transmitter and can also be used: C12 is replaced by a within 6 ft. (2m) of the transducers.
one as receiver. series connection of two diodes (the
The 'old' Albar printed circuit hoard has anode of the first to the collector of
been replaced by a new version that can T5 and the cathode of the other to
be used both for the modified single - CI I) and a reset button is added
transducer circuit and for the receiver parallel to C11.
section of the two -transducer circuit. A The working voltages of several
small additional board is also available electrolytics have been changed to
for the transmitter circuit of the two - more readily available values.
Figure 1. Modified circuit for the single -
transducer version. transducer version.
Construction and alignment
Single -transducer version The new printed circuit board is shown Figure 2. Receiver circuit for the two -
Figure 1 gives the circuit of the new in figure 4. For this version, only the transducer version. Point 'X' is connected to
single -transducer version. It differs from `normal' components should be the transmitter.
 albar mkll-a bfs elektor may 1977 - 5-53

I
910 12V

CI 0,
210

C113
100
16V 400 mw
61
2709
16V

T3 I
0I ICI
T4
10
C4 741
T c6
14

Pt 1017,
7177
mos

-f 412 10
n0..1006
16V

7.2, 4 d
220n
C14

479
-( 1115

142 11
413
l--] 100,0
02 ,11

47771
16V

A
C2
mlm 16V 160

17
4k7

D2 .. 05 = IN4148
T I ... T5 = BC5478 BC1078

R10
2 TO SA

C 3

10V RI R4
4007TTV r R8

C1 .--

ICI
292
10
63V 741 1i 675

C6.6
.60
1007, -3
10.1009

1611
P7'5
7115

3306
6-4 47k
C9 02 C11
CS 813 '30
C_.
Ht H6 618 7117
111
4M1= A
22p 009 479
0 63
C16
O 10V
16V
P7
16V

72p
6V 4k 7

9428. 943766 4444 9836 2

D2 05=1N4148
T 1,T3 T5 = BC5478 BC 1078 C14 1s7600T,
T2= BC55713
NB! 'Modification for use as 611-4-7-77-1
979
receiver 16v

r Parts list for single -transducer version R14 = 2k2 C12 = 10 ... 100 4/16 V
f (figures 1 and 4) R16,Rv = 4k7 C13 = 470 µ/16 V
S P1 = 2k2
P2 = 4k7 Semiconductors:
IC1 = 741
Resistors: Capacitors: T1 ... T5 = BC547 B or BC107 B
R1,R9,R11 = 100 k Cl = 22 g/16 V D1 = 10 V/400 mVV
R2 = 68 k C2,C6 = 100 n D2 . . D5 = 1N4148
.

R3,R4 = 5k6 C3 = 5n6

R5,R17 = 1 k C4 = 1 n Sundries:
R6 = 56 k C5 = 22 g/10 V L1 = 2.2 mH
R7 = 10 k C7 = 2µ2/63 V US = Murata MA 40L 1R or
R8 = 33 k C8 = 220 n MA 40L 15
a-
R10 = 270 St C9 = 100 g/16 V LS = headphone or earpiece
R12,R15 = 47 k C10 = 10 g/16 V IR 3= 600 n)
R13=1052 C11,C14 = 47 g/16 V Re = 12 V/20 mA relay
:o
5-54 - elektor may 1977 albar mkII-a bfs

3
®12V
CS,

IV° 16V

Ox
R3
lk
OY
P1 555
L1 68..nR

C3 Ca
NMI
72n2 T°on 173n 4n7

9836 3

Figure 3. Transmitter for the two -transducer

version.

Figure 4. Printed circuit board and com-

ponent layout for both figure 1 and figure 2.
01R3 f0 ( EPS 9815-1).

1.° (0) 0-1F00 Figure 5. Printed circuit board and com-

R4 r 1=0-14) ponent layout for the transmitter in the two -
to cis transducer system. (EPS 9815-2).

Figure 6. The number of transmitters (or

receivers, for that matter) can be extended at
will, provided the transmitters are arranged to
run at the same frequency.

The procedure is as follows:

- set P2 at maximum resistance (fully
anti -clockwise).
- as before, move a hand to and fro in
front of the transducers, and set P1
on the transmitter board for maximum
level of the tone in the earpiece.
- adjust P2 until the correct sensitivity
is obtained.
One final tip: in some cases one
transducer is better than the other, so
it is worth while interchanging them to
see which one is better for the receiver.

Three -transducer version .

In theory it is possible to extend the
two -transducer version by adding one or
more extra transmitters.
For this it is essential that they should
all oscillate at the same frequency.
Although we have not tried this in
practice, the principle is relatively
simple. Output 6Y' from one of the
transmitters (the `master' oscillator)
is used to drive or synchronise the
other(s).
There are two ways to do this. Provided
the cable capacitance of the link
between the transmitters is not too
high, components R 1 , R2, P1 and Cl of
the 'slave' transmitter can be omitted
and output 'Y' from the master is fed to
pins 2 and 6 of the IC on the slave
transmitter. Alternatively, all com-
ponents are mounted on the `slave'
board, but the lower end of C2 is
disconnected from supply common and
the `Y' signal is fed in via this capacitor
to pin 5 of the IC. P1 on the 'slave'
transmitter is now adjusted for 'zero
beat' in the earpiece.
albar mk1I-a bfs market elektor may 1977 - 5-55

2 -hour room -temperature drying

period, which can be accelerated
to 1 hour at I70°F. When
elevated -temperature drying is
used, the coating should first be
allowed to dry tack -free for
20 minutes at room temperature
to avoid solvent entrapment.
For evaluation and propotype
purposes, 4900 is available in 6 oz.
aerosol cans. However, for best
control of uniformity and
thickness, conventional spray
equipment should be used.
A thicker, non-sprayable version
of 4900 is also available, and is
especially useful for brushing on
areas difficult to coat with a spray.
Chomerics,
66 Rue la Boitie, 75008 Paris,
France
(449 M)

Recording audiometer
Parts list for receiver C6 = 100 n An audiometer for statutory
(figures 2 and 4) C7 = 2112/63 V audiometry in connection with
C9 = 100 µ/16 V health investigations has been
C10 = 10 µ/16 V developed by Briiel & Kjaer.
C11 ,C14 = 47 4/16 V Conductive spray paint Designated Type 1800, the
Resistors: C12 = 10 100 p/16 V Chomerics' 4900 conductive paint recording Bekesy-type audiometer
R1 = 22 k C13 = 470 p/16 V is a one -component, air -drying provides a pure tone signal
R2,R7 = 10 k C15 = 330 p coating which will provide (continuous or pulsed) at 7
R3,R19 = 560 5-2 C16 = 22 4/6 V effective EMI attenuation when different frequencies from
R4 = 5k6 applied in a 2 -mil coating on non- 500 Hz to 8000 Hz and records
R6 - 56 k conductive enclosures. It can also the patient's response automati-
R8 = 33 k Semiconductors: be used improve the shielding cally. The audiometer is basically
R9,R11 = 100 k
ICI = 741 characteristics of metal enclosures an x - y recorder - the x- axis
R10 = 270 .S1 by reducing the contact resistance representing the test -frequency
R12,R15 = 47 k T1,T3 ... T5 = BC547B or and the y- axis representing the
of flanges; to create or improve
R13 = 100 it BC1076
grounding surfaces; to fill minute hearing -threshold of the patient
R14 = 2k2 T2 = BC5576 or BCI 778
D1 = 10 V/400 mW
voids in porous metal castings; (in the range - 10 dB to + 90 dB
R16,Rv = 4k7 and to make conductive patterns HL) as registered with the help of
R17 = 1 k
D2 . D5 = 1N4148
on nonconductive substrates. a patient -operated handswitch.
R18 = 1k5 This paint is a solvent -based The patient is instructed to hold
P2 = 4k7 system consisting of a silver filler the button on the handswitch
Sundries: in an acrylic resin. It is specifi- pressed down as long as he hears
cally intended for spray - the signal in the matched
US = Murata MA 40L 1R or
Capacitors: application, but can also be earphones, thus causing the signal -
MA 40L IS applied by dipping or brushing.
C1 = 22 µ/16 V LS = headphone or earpiece
level to decrease, and to release
C4 = 1 n Regardless of application method, the button when he no longer
(R 6600 S2)
C5 = 22 p/10 V 4900 must be agitated often to hears the signal. In this way he
Re = 12 V/20 mA relay
prevent the dense filler from tracks his own threshold level.
settling out and causing resin -rich Remote control facilities and
areas in the coated surface. lamp indication of patient
Coverage is approximately 70 sq. response outside normal range
ft. per pound in a 1 -mil thickness. makes the audiometer ideal for
When used as a shield, the 2 -mil group testing.
thickness should be built up in
2-3 applications, allowing 15-20 Briiel & K/aer,
minutes after each application for 23 Linde alle , DK-2850 Naerum,
Parts list for transmitter (figures 3 and 5) solvent evaporation. Optimum Denmark
properties are developed after a (452 M)
Resistors:
R1,R3 = 1 k
R2 = 4k7
P1 =4k7

Capacitors:
C1 = 2n2
C2 = 100 n
C3 = 33 n
C4 = 4n7
C5 = 10 4/16 V

Semiconductors:
IC = 555

Sundries:
LI = 6.8 mH
US = Murata MA 40L 1R or
MA 40L IS
 15 240 WATTS !
1-IY5 The HY5 is a mono hybrid amplifier ideally suited for all applications. All common input functions Imag Cartridge.
tuner. etc I are catered for internally, the desired function is achieved either try a multiway switch or direct connec-
tion to the appropriate pins. The interval volume and tone circuits merely require connecting to external potent.,
mete, S OW included, The HY5 is compatible with all I.L.P.power amplibers and power suppties. To ease eon.
preamplifier Reaction and mounting a P.C. connector is supplied with each preamplifier.
FEATURES Complete preamplifier in single pack - Multi function equalization tow noise Low distor Don
High overload - Two sorriply cOinbined fur stereo
APPLICATIONS Hi -Fr - Mixers - Disco - Guitar and Organ - Puolic address.
SPECIFICATIONS,
INPUTS. Magnetic Pick-up 3 mV, Ceramic Pick-up 30 mV. Tuner 100 mV. Microphone 10 mV. Auxiliary 3.100
mV; input impedance 47 kit at 1 kHz.
OUTPUTS. Tape 100 my. Main OUSOUt 500 mV H.M.S
ACTIVE TONE CONTROLS. Treble 12 dB at 10 kHz Bass , at 100 Hz.
DISTORTION- 0 1% at I kHz SionaEhloise Ratio 68 dB.
OVERLOAD. 38 t-113 on vai.ehi- Pick up. SUPPLY VOL TAG,- -'6 SO V
Price £ 5.22 + 65 p. VAT P&P free.
HY5 Mounting Board B1 48 p + 6 p VAT P&P free.

HY30 The HY...50 is an exciting Niew v.1 fruits :.L.P. it features a virtually inoestructible I.C. with short circuit and theonai
protection. The kit Consists 01 IC. heatsink, P.C. hoard. 4 resistors, 6 CetiaCilois, mounting kit, together with easy
to follow construction and operating instructions. This amplifier is ideally suited to the beginner in audio who
wishes to use the most up -to date technology available.
15 Watts into 811 FEATURES: Complete Kit - Low Distoi non - Shor I. Open and Theo.' Protection - Easy to Build.
APPLICATIONS, Updating audio eouipment - Guitar practice amplifier - Test amplifier - audio oscillator.
SPECIFICATIONS:
OUTPUT POWER 15 W R.M.S. into 8 11 DISTORTION 0.1% at 15 W
INPUT SENSITIVITY 500 mV. FREQUENCY RESPONSE 10 Hz -16 kHz -.3 dB.
SUPPLY VO- I AUL ! 18 V
Price £ 5.22 + 65 p. VAT P&P free.

HY50 The HY50 leads I.L.P.'s total integration approach to power amplilier design. The amplifier features an integrul
heatsink together with the simplicity of no external components. Outing the past three years the amplifier has been
refined to the extent that it must be one of the most reliable and robust High Fidelity modules in ilia World.
FEATURES: Low Distortion Integral Heatsink Only rise Connections 7 Amp output transistors - No ester
25 Watts into en net components
APPLICATIONS- Medium Power Hi.Fi systems - Low power disco - Guitar amplifier
SPECIFICATIONS:
INPUT SENSITIVITY 500 mV.
OUTPUT POWER 25 W R.M.S_ into 8 O. LOAD IMPEDANCE 4 16 11 DISTORTION 0.04% at 25 W at 1 kHz.
SIGNAL! NOISE RATIO 75 dB. FREQUENCY RESPONSE 10 Hz -45 kHz -3 dB.
SUPPLY VOLTAGE t 25 V_ SIZE 105x50025 mm.
Price £ 6.82 + 85 p. VAT P&P free.

HY120 The HY 120 is the hates of 1.1__P_'s new high power range, designed to meet the most exacting requirements
load line and thermal protection this amplifier sets a new standard in modular design.
FEATURES. Very low distortion - Integral heatsink - Load line protection - Thermal protection - Five corner,
non - No external components.
50Watts into SR APPLICATIONS- HeFi - High quality disco Public address - MonilOr amplifier - Gilder and organ.
SPECIFICATIONS:
INPUT SENSITIVITY 500 mV.
OUTPUT POWER 60 W FLM.S. into 811. LOAD IMPEDANCE 4.16 D. DISTORTION 0.04% at 60 W al 1 kHz.
SIGNAL1NOISE RATIO 90 dB. FREQUENCY RESPONSE 10 Hz.45 kHz 3 dE, SUPPLY VOLTAGE 35 V.
SIZE 114050x85
Price £ 15.84 + £1.27 VAT P&P free.

HY200 The HY200 now improved to give an output of 120 Watts has been designed to stand the most rugged conditions
such as disco or group while still retaining true HiFi performance
FEATURES. Thermal shutdown - Very low distortion - Load line protection - Integral heatsink - No external
components_
120 Watts into 811 APPLICATIONS, Hi-Fi - Disco - Monitor - Power stave - Industrial - Public Address.
SPECIFICATIONS:
INPUT SENSITIVITY 500 inV.
OUTPUT POWER 120 W R.M.S. into 8 D. LOAD IMPEDANCE 4.16 Si_ DISTORTION 0.05% at 100 W at 1 kHz.
SIGNALiNOISE RATIO 96 dB. FREQUENCY RESPONSE 10 liz45 kHz -3 dB. SUPPLY VOLTAGE 45 V.
SIZE 1140100x85 mm.
Price £ 23,32 + £1.87 VAT P&P free.

HY400 The HY40.3 sILF 's '91.-; Oddly of the range o,ctd..., 740 W into 4 11' it has been designed 10i high IJONItt (i13.1
or public address applications. II the amplifier is to be used at continuous high power levels a cooling fan is rec.
nrnnlended. The amplifier includes all the qualities of the rest of the family to lead the market as a true high power
hi -fidelity power module -
240Watts into 411 FEATURES. Thermal shutdown Very low distortion Load line protection - No external components
APPLICATIONS. Public address - Disco - Power slave - Industrial
SPECIFICATIONS:
OUTPUT POWER 240 W TENTS. into 411 LUAU IMPEDANCE 4 16 11 DISTORTION 0 at 24005 at kHz. 1

SIGNAL,NOISE RATIO 94 dB. FREQUENCY RESPONSE 10 Hz.45 kHz 3 dB SUPPLY VOLTAGE 45 V_

INPUT SENSITIVITY 500 mV SIZE 114x 100x85 mm
Price £ 32.17 + £ 2.57 VAT P&P free.

power PSU36 suitable for two HY30's E 5.22 plus 65 p VAT P&P free
111$1.150 suitable for two H VSO's 8.82 plus 85 p VAT P&P free

supplies PSU70 suitable for two HY120's 13.75 plus E 1.10 VAT P&P free.
PSU90 suitable for one HY200 12.65 plus E 1.01 VAT P&P free.
P5U180 suitable for two HY200's or one 110400 E 23.10 plus 1.65 VAT
P&P free.

TWO YEARS' GUARANTEE ON ALL OF OUR PRODUCTS

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 !IlH J Jill iii
This high quality Graphic Equaliser will enhance even the
most sophisticated system ate fraction of the cost of a
ready-made equaliser of equal quality. You can get all the
Parta from us 'except woodwork' including drilled and
.1.1111101003
printed metalwork. Consumption details in our leaflet: 15p.

wkim is extremely
ve got toggle mos..,

IT'S A FANTASTIC BESTSELLER!

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Over 30 pages of complete projects to build!
Thousands and thooaands of useful components described
Ilex.!

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NAME
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