The Big Book of Sound - ADC Sound Design (2003) PDF

The Big Book of Sound:
ADC Sound Design
by
Andrew Love, Simon Parry,
Matt Johnson and Rob Loxley
Version 2002.b
The Big Book of Sound
Contents
List of Illustrations .....................................................................................................................................................5
Introduction ................................................................................................................................................................6
Help! .....................................................................................................................................................................6
Thanks! .................................................................................................................................................................6
ADC General Information .........................................................................................................................................7
Use of Equipment .................................................................................................................................................7
Access to ADC Sound Facilities ...........................................................................................................................7
Sound System Power ............................................................................................................................................8
Points to Remember ..............................................................................................................................................8
ADC Sound Equipment ..............................................................................................................................................9
Microphones and where to put them ......................................................................................................................10
Windshields and Popshields ...............................................................................................................................12
Feedback .............................................................................................................................................................13
Radio Mics..........................................................................................................................................................14
Sound Media .............................................................................................................................................................16
Cassette Tape ......................................................................................................................................................17
Compact Disc......................................................................................................................................................19
Digital Audio Tape .............................................................................................................................................19
Open Reel Tape ..................................................................................................................................................21
Vinyl ...................................................................................................................................................................22
Sound Material .........................................................................................................................................................23
PPL and PRS.......................................................................................................................................................24
Sound Processing ......................................................................................................................................................25
The Reverb Unit..................................................................................................................................................26
The Multi-effects Unit ........................................................................................................................................26
The Graphic Equalisers.......................................................................................................................................28
Dynamic Processing............................................................................................................................................30
The ADC Sound Desk (Mixer).................................................................................................................................31
Input Channels ....................................................................................................................................................32
Other Inputs ........................................................................................................................................................35
Master Section ....................................................................................................................................................35
Mute System Operation ......................................................................................................................................35
ADC Sound Output ..................................................................................................................................................37
Auditorium Speakers...........................................................................................................................................37
Stage and Band Foldback....................................................................................................................................37
Sound to Light ....................................................................................................................................................38
Induction Loop....................................................................................................................................................38
The Perfect Cue Sheet ..............................................................................................................................................39
Recording Shows.......................................................................................................................................................40
Audio ..................................................................................................................................................................40
Video ..................................................................................................................................................................40
The Law ..............................................................................................................................................................40
Mixing the Recording .........................................................................................................................................41
The Band ...................................................................................................................................................................42
A Happy Band... .................................................................................................................................................42
Page 2 of 107
Miking the Band .................................................................................................................................................42
Mixing the Band .................................................................................................................................................43
MIDI ...................................................................................................................................................................43
ADC Sound Wiring...................................................................................................................................................45
Fixed Wiring .......................................................................................................................................................45
Cabling................................................................................................................................................................46
Patchbays...................................................................................................................................................................48
Unusual Patchleads .............................................................................................................................................49
Hum and Interference ..............................................................................................................................................50
Hum ....................................................................................................................................................................50
Interference .........................................................................................................................................................51
Balanced Connections ..............................................................................................................................................53
Sound out and about.................................................................................................................................................55
Touring ...............................................................................................................................................................55
Sound in the Open...............................................................................................................................................55
Further Reading .......................................................................................................................................................57
The Internet.........................................................................................................................................................58
Appendix A Sound in Other Cambridge Venues ...................................................................................................59
Arts Theatre ........................................................................................................................................................60
Christ’s New Court Theatre ................................................................................................................................61
Corn Exchange....................................................................................................................................................61
Downing College Theatre ...................................................................................................................................61
Emmanuel URC ..................................................................................................................................................62
Fitzpatrick Theatre, Queens’...............................................................................................................................62
Holy Trinity Church............................................................................................................................................63
The McCrum Theatre, Corpus ............................................................................................................................63
Mumford Theatre, APU ......................................................................................................................................64
Peterhouse Theatre..............................................................................................................................................65
Playroom.............................................................................................................................................................66
Queens Building, Emmanual...............................................................................................................................66
Robinson Auditorium..........................................................................................................................................66
St. Chad’s Octagon, St Catherine’s.....................................................................................................................66
School of Pythagoras, St. John’s.........................................................................................................................67
Appendix B Sound Connectors................................................................................................................................68
BNC ....................................................................................................................................................................69
DIN .....................................................................................................................................................................69
Jack .....................................................................................................................................................................71
Lemo ...................................................................................................................................................................72
Phono ..................................................................................................................................................................72
SCART ...............................................................................................................................................................72
Speakon...............................................................................................................................................................73
XLR ....................................................................................................................................................................74
Appendix C Sound Cables........................................................................................................................................76
Helical Screened Twin (HST).............................................................................................................................77
Musiflex ..............................................................................................................................................................77
Oxygen Free Copper ...........................................................................................................................................77
Starquad ..............................................................................................................................................................77
Page 3 of 107
Multicore Cables.................................................................................................................................................78
Speaker Cables....................................................................................................................................................78
Mains Cable/flex.................................................................................................................................................78
Appendix D MiniDisc Instructions..........................................................................................................................79
Playback..............................................................................................................................................................79
Recording............................................................................................................................................................80
Editing.................................................................................................................................................................80
Appendix E Reverb Unit Instructions.....................................................................................................................83
Appendix F ADC Patchbay Layouts .......................................................................................................................86
Appendix G ADC Sound Effects Library ...............................................................................................................90
Appendix H Sound levels, frequencies and other figures ......................................................................................91
Sound Levels.......................................................................................................................................................91
Metering Sound Levels .......................................................................................................................................93
Hearing Damage .................................................................................................................................................94
Octaves and Frequencies.....................................................................................................................................94
Appendix I A guide to compression........................................................................................................................95
Overall Compression ..........................................................................................................................................95
The Controls .......................................................................................................................................................96
Multi-band Compression.....................................................................................................................................96
Glossary of Sound and Theatre Terms ...................................................................................................................97
All rights reserved. This document may be freely copied, loaned and electronically transmitted in
Postscript and Portable Document Formats, on the condition that the document (including this
message) is kept whole. Excerpts may only be used with the authors’ prior permission, and should
clearly identify the authors as the holder of copyright on the excerpt.
Please note that this guide was originally written for the UK market; any references to “mains”
electricity are taken to mean a 230 Volt 50 Hz AC supply, and all legal references (e.g. copyright,
royalties) are based upon the authors’ understanding of current UK law.
Whilst we have taken the greatest possible care to ensure the accuracy of the advice contained in it,
the authors accept no liability for any loss, damage or injury, howsoever caused, arising from the
use of this guide.
© Andrew Love, Simon Parry, Matt Johnson and Rob Loxley

9 February, 2003
Page 4 of 107
List of Illustrations
Figure 1 - Block diagram of sound power system.......................................................................... 8
Figure 2 - Microphone Polar Patterns .......................................................................................... 10
Figure 3 - A Shure SM58............................................................................................................. 11
Figure 4 - An AKG D190............................................................................................................. 11
Figure 5 - A MiniDisc.................................................................................................................. 20
Figure 6 - The ADC processing & effects units........................................................................... 25
Figure 7 - Dynamic processing in a typical recording setup ........................................................ 30
Figure 8 - Main signal and sidechain signal paths in a dynamic processor ................................. 30
Figure 9 - The ADC sound desk .................................................................................................. 31
Figure 10 - Block diagram of a simple mixer .............................................................................. 31
Figure 11 - Input channel facilities on the ADC sound desk ....................................................... 32
Figure 12 - MIDI device 1 controlling MIDI devices 2 and 3 via a thru port .............................. 44
Figure 13 - MIDI device 1 controlling MIDI devices 2 and 3 via a splitter box.......................... 44
Figure 14 - Plan of ADC Stage showing connection points (not to scale)................................... 45
Figure 15 – The Sound Shelf Cupboard....................................................................................... 46
Figure 16 - How to fit a cable rubber........................................................................................... 47
Figure 17 - How a hum loop is generated .................................................................................... 50
Figure 18 - An audio signal.......................................................................................................... 53
Figure 19 - An audio signal and its inverse.................................................................................. 53
Figure 20 - The signals pick up interference................................................................................ 53
Figure 21 - ...and then it is cancelled out ..................................................................................... 53
Figure 22 - Connecting an unbalanced output to a balanced input .............................................. 54
Figure 23 - Connecting a balanced output to an unbalanced input .............................................. 54
Figure 24 - A 100 Volt line system .............................................................................................. 56
Figure 25 - A BNC connector ...................................................................................................... 69
Figure 26 - Pin layouts of common DIN connectors.................................................................... 70
Figure 27 - “A” gauge three pole quarter-inch jack .................................................................... 71
Figure 28 - “A” gauge two pole quarter-inch jack ...................................................................... 71
Figure 29 - A Phono connector .................................................................................................... 72
Figure 30 - Pin layout of a SCART connector ............................................................................. 72
Figure 31 - A Speakon connector................................................................................................. 73
Figure 32 - An XLR connector, also showing pin layout ............................................................ 74
Figure 33 - The microphone and main patchbays, also showing the cable tester ........................ 87
Figure 34 - The ring intercom, amp rack and speaker patchbays................................................. 87
Figure 35 - The ear's logarithmic response .................................................................................. 91
Cover Illustration: The Sound Box, ADC Theatre, Cambridge
Page 5 of 107
Introduction
This guide is intended to turn newcomers to sound into competent sound designers, and as a general
reference for everyone. It explains both technical and artistic considerations, and covers design,
fault-finding and operation. It is also hoped that this guide will form a useful reference work for
anyone undertaking sound design in the ADC Theatre.
Although this guide was originally written for use in the ADC Theatre, the principles described hold
true for any theatre; for reference, basic descriptions of other venues in Cambridge have been
included as many sound designers will work in several of them.
Sound isn’t the “black art” that most people will have you believe. It simply requires patience, a
logical mind and several very strong cups of coffee to achieve some truly excellent results.
Finally, note that the latest version of this guide can be downloaded from the web in several formats
at http://www.robloxley.co.uk/bbos/. Comments or suggestions can be mailed to the authors (see
below).
Help!
If you have problems with a sound system, and have worked though all the information contained in
this guide, or if you need ideas for a sound design, or even if you just want to know more about
sound, and how it all works, feel free to contact the following people:
Name Telephone E-mail

Simon Parry sparry@iee.org
Andrew Love 07941 098483 aghlove@iee.org
Matt Johnson 07802 719981 mij20@cam.ac.uk
Rob Loxley 07939 072342 ral32@cam.ac.uk
The Technical Manager at the ADC is also a useful fountain of knowledge, and will have dealt with
the vast majority of problems before.
Thanks!
Thanks to: Laura Hill for the title, James Irvine, Duncan Wood, Alan Morgan and Adrian Cresswell
for answering endless silly questions, everyone who supplied information about venues in which
they had worked (a list is given in Appendix A
Sound in Other Cambridge Venues) and everyone who reviewed and proof-read this guide!
Page 6 of 107
ADC General Information

Use of Equipment
As a general rule, if the ADC Theatre owns a piece of equipment, it can be used by any show. The
current exceptions are the six radiomics, which need to be hired from the theatre if required. Also,
Cambridge Footlights own the two Technics CD players, the Sony MD player and the EMO DI box;
which normally live in the theatre except during the Footlights Summer Tour. Anything that is not
fixed down and is not being used by the current shows in the theatre can also be hired for an event
anywhere else. This includes:
• microphones
• speakers
• adapters
• cables
• small mixer
• flightcased amplifier
Therefore, if you are involved in a show at the ADC, let the technical manager know which pieces
of equipment you intend to use as early as possible, and preferably two weeks before the start of the
show - otherwise, you may well find that the equipment has been hired to another production!
Certain pieces of equipment are locked in the “cans cupboard” in the technical office corridor.
These include:
• microphones and clips
• sound effects CDs
• intercom sets
• direct inject boxes
Access to ADC Sound Facilities

The sound and lighting boxes are joined together, and can be reached from stage over the
auditorium. However, use the fire escape in preference to this route during the show - the audience
can hear your footsteps as you walk over their heads! Ensure that this escape is unlocked when the
control boxes are in use, and likewise that it is locked when the boxes are unattended.
The amplifiers are located in a rack on the Prompt side of stage. The black door on the first landing
(“The Sound Shelf”) leads to the sound storage cupboard, where leads, stands and adapters may be
found. Please remember to leave it tidy!
Page 7 of 107
Sound System Power

The ADC sound system is powered through a special mains distribution board located opposite the
amp rack, and is switched on and off with control panels located on the amp rack, at stage level
under the amp rack, and in the sound box (see Figure 1.) When powered up, the sources and “sound
power” sockets will be switched on instantly, and the amplifiers will power up 20 seconds later.
When powered down, the amplifiers will be switched off instantly, and the sources and “sound
power” sockets will power down 20 seconds later. This arrangement minimises the thuds associated
with switching sound equipment on and off, which can cause damage to speakers.
An LED at each station lights up in red when the system is off and in green when the system is on.
Sound Box Stage Level Amp Rack

Control Panel Control Panel Control Panel
Control Unit
Supply to Sound Box equipment

Distribution Board Amplifiers
and Contactors
Supply to Sound Power sockets
Incoming Mains Supply
Figure 1 - Block diagram of sound power system
Points to Remember
• Be very aware that EVERYTHING which is said in the sound box during a show can be heard by
the audience; this is clearly unprofessional. If you must speak - speak very quietly. If you need to
talk normally, go into the lighting box and close the curtain. If you need to talk loudly, go
outside. This also applies to the L.X. Op. and anyone else in the box.
• Beware of monitoring at high levels for extended periods of time - it can damage your hearing!
• Do not use the monitor speakers in the sound box during a show - unless the audience are meant
to hear them!
• Remember that your show has to share the theatre with other shows; at best, there will be a main
show and a late show, and there will sometimes be several main shows in rep. Show some
consideration to other shows by only using what you need, labelling all faders (and cables, if long
or complex runs are used), and by leaving a note if you have an unusual setup which might affect
them. Liase with other shows before repatching their rig for whatever reason, and always
leave a note showing what you have done and how to restore their patch.
• Even if you have nothing to do, it is polite to turn up at least 40 minutes before a show. This
stops the Stage Manager from panicking, and means that if you can’t get there for some reason,
there is time for them to find someone else!
It’s all obvious really - just use some common sense...
Page 8 of 107
ADC Sound Equipment

The theatre has a wide range of equipment available. This includes:
Mics: 1 x Shure SM48

4 x Shure SM58
4 x AKG D190E
1 x Beyer M201
2 x AKG C1000s
1 x Tandy PZM
2 x Trantec S3000 &
4 x Sennheiser 1083 VHF radio microphones
Mic stands: 4 x Mic stands

2 x Boom arms
4 x Telescopic boom arms with 2” clamps to attach to scaffold bars
Sources: Denon MiniDisc DN-990R wired remote control in sound box.

Sony MDS-JE510 minidisc recorder owned by Cambridge Footlights
2 x Technics CD players owned by Cambridge Footlights
Denon DN-770R twin cassette deck can be used as two separate decks
Ferrograph “Super” open reel tape machine 1/4 inch tape format - in technical office
corridor
Mixer: Soundcraft SeriesTWO 24 channel desk
Effects: Alesis M-EQ230 graphic equaliser

Behringer Ultracurve equaliser/ spectrum
analyser/ delay/ feedback destroyer
Yamaha REV100 reverb unit
Behringer Modulizer Pro DSP1224 multi-
effects processor
Four channel ARX Quadcomp compressor
Behringer Composer Pro MDX2200 stereo
compressor/ limiter/ expander
Amplifiers: 1 x C-Audio RA3001 (625W / channel)

1 x Bose 1800 (250W / channel) 4 channels in amp rack on sound shelf
1 x Crest V350 (350W / channel) 2 channels in flightcase on sound shelf
1 x Monitor amp 2 channels for monitors in sound box
Speakers: 6 x Bose 802 (240W) 6 channels Bose EQ available

2 x Bose 101 monitor speakers fitted in sound box
Other: 5 x Stereo direct injection boxes with earth lift switch; one owned by
Cambridge Footlights
Spirit Folio Notepad portable mixer 4 mono channels, 2 stereo channels
Allen & Heath SC Plus 24-4-2 desk Old sound box mixer; 24 channels
Similar details for other venues in Cambridge are given in Appendix A

Sound in Other Cambridge Venues.
Page 9 of 107
Microphones and where to put them

At a first glance, microphones are simple to use - you point them at what you want to pick up! But
what will they pick up? Microphones come in several “polar patterns” which define in which
directions they are sensitive. These are shown in
Figure 2 and described below:
• Omnidirectional - picks up sound from all directions.
• Cardioid - picks up sound that is pointing towards the top surface of the windshield. Used for
most vocal applications.
• Hypercardioid - picks up sound that is pointing through the very top of the windshield. Used in
many vocal applications. Note that although these microphones pick up a tighter beam of sound
from the front, they also pick up slightly more sound from the rear than a cardioid, so beware
feedback!
• Shotgun - the “spotlight” of microphones. Useful for picking up sound from the back of a stage
or from a long way off.
• Rifle - picks up a narrower “spot” of sound that a shotgun - otherwise similar.
• Lemniscate / figure of eight - picks up an even lobe of sound from each side. Not often used in
the theatre - more useful for interviews and advanced stereo recording techniques.
Omnidirectional Cardioid Hypercardioid Lemniscate/

figure of eight
Shotgun Rifle
Figure 2 - Microphone Polar Patterns
Microphones can also be divided into two categories according to construction:

• Dynamic or moving coil microphones are just like speakers in reverse - sound waves make a
diaphragm bounce about and this wiggles a coil in a magnetic field which generates a signal.
These microphones are cheap to make and easy to use.
• Condenser, electret or capacitor microphones use the diaphragm as part of a capacitor.
Changes in the voltage across this capacitor are amplified by circuitry within the microphone and
the signal from this amplifier is fed out, usually via an impedance matching circuit. These
microphones always need a power supply to run the amplifier; this is either provided by internal
batteries or by “phantom power”, where either the mixer or an external unit feeds the microphone
power along the signal cable. These microphones exhibit a wider and smoother frequency
response and are physically smaller than dynamic microphones, but they are not so rugged and
are generally more expensive.
Page 10 of 107
The ADC possesses several types of microphone. These are:
• Shure SM48 - a general purpose cardioid dynamic vocal microphone, characterised by a bright,
lively sound, and producing quite a high output level. As well as vocals, this is a good mic for
drums, brass and strings.
• Shure SM58LC - the classic vocal microphone. A higher quality version of the SM48, this
cardioid dynamic microphone makes vocalists sound better than they actually are by “peaking”
slightly at just the right place in its frequency response and therefore giving a brighter, punchier
sound. See Figure 3.
• AKG D190 - this dynamic microphone is very honest and impartial, with a flat frequency
response and an even cardioid pickup pattern. Useful in the pit, for stage miking at short ranges
and for recording shows. See Figure 4.
• Beyer M201 - a high quality hypercardioid dynamic microphone most useful for picking out
individual instruments in the orchestra pit at close range. Small and inconspicuous.
• AKG C1000s – this is a cardioid condenser microphone powered either by a 9V PP3 battery or
phantom power. An excellent quality vocal microphone, also great used for instruments
including strings, flutes and woodwind. A special converter (PPC 1000) turns the microphone
characteristics from cardioid into hypercardioid, if it is mounted on the mic top. The PB 1000
Presence Boost Adapter, if fitted, provides an additional 5-dB high-frequency peak in the
cardioid mode adding brilliance in the 5 to 9 kHz range.
• Tandy PZM (Pressure Zone Microphone) - a cheap “boundary layer” microphone, which
consists of a steel plate with a lump on the top containing an electret microphone. Fix them to
flat surfaces with gaffer tape and experiment with them to se how they pick up much of the
sound hitting that surface! They work well taped to the floor under pianos (so long as the
mechanics of the piano or “action” is smooth) and pick up a surprising amount of dialogue if
taped to the front of the stage.
Figure 3 - A Shure SM58 Figure 4 - An AKG D190
Rifle and shotgun mics also useful for large musicals, where dialogue and singing need to be lifted
to a level where they can be heard above the band; they work well hiding in the perches or hanging
overhead from bar zero or even further back in the grid. However, these microphones need to be
hired in, as the theatre does not currently possess any. A more conventional approach is to place two
or three microphones across the front of the stage (in the “float” position); PCC-160s or the D190s
work well in this arrangement. Note that if the show uses a band in the pit, noise from this is likely
to be picked up by float microphones and may drown out sound from the stage, rendering them less
useful.
Page 11 of 107
There should be datasheets for all the microphones that the ADC Theatre owns, in the Sound File in
the sound box.
Don’t forget that microphones also pick up some sound due to the body being knocked. Don’t put
them where they are likely to get kicked or hit, and avoid using them in floor stands if the stands
could pick up vibration from the floor. To reduce the amount of noise being picked up from stand
vibration, a “microphone suspension” can be used. This is an elastic sling into which the
microphone fits, and which acts like a shock absorber by damping movement.
Where several wired or radio handheld mics are being used at once (e.g. stand-up comedy), it is
worth making the mics separately identifiable to aid control. This can either be done by wrapping a
ring of coloured LX tape around the body of each microphone, or simply by using microphones with
different body colours. If the mixer channels are also labelled with the colour code, the sound op
can then identify which channel corresponds to which microphone quickly and over a reasonable
distance.
Windshields and Popshields

Other accessories to prevent unwanted noise from being picked up by microphones are windshields
and popshields.
Windshields are moulded foam hoods which fit snugly over the top of microphones and prevent
wind from causing a hissing sound when microphones are used outdoors. Many microphones have a
basic windshield build in under the “mesh” top.
Popshields are circular hoops over which a fine fabric is stretched which clip onto the side of a
microphone so that the fabric is between a vocalist’s mouth and the microphone head; they prevent
consonants (especially at close range) causing an exaggerated “popping” sound, firstly be breaking
large blasts of vocal air into many smaller puffs, and secondly by physically preventing the vocalist
from getting too close to the microphone! A cheap but effective popshield can be made by
stretching stocking fabric over wire hoop formed from a coat hanger. Fore visual reasons, these are
very rarely used in theatres, but are very common when recording.
Page 12 of 107
Feedback
Feedback (“howlround”) is caused by sound from a speaker finding a path back into a microphone
and “chasing itself” around the audio system. Sound design involves juggling the position of
microphones and the levels at which they can be run to achieve sufficient clarity and volume
without risking feedback. The following tips will help to get rid of feedback:
• The golden rule: don’t put microphones too close to speakers! By all means experiment to see
what you can get away with, but stick well within the limits which you discover.
• If microphones are to be used for vocals, it is advisable to put them in a stand wherever possible;
if the performer is allowed to walk around the stage with the microphone when the levels are
cranked right up, it is almost certain that they will put the microphone somewhere that will cause
feedback!
• If you get problems, don’t forget that you can move the speaker as well as the microphone!
• If it is artistically acceptable, try panning the microphone away from the speaker which is causing
the feedback. This will reduce the amount of that microphone’s signal coming out of the speaker
(so reducing the chances of feedback) and compensate for this by increasing the amount of that
microphone’s signal coming out of other speakers.
• The graphic equaliser will sort out most feedback problems. Patch it into the microphone channel
giving problems and try “notching” out the frequencies until the feedback disappears. To find
these frequencies, insert the graphic equaliser (“EQ”) into the system and then turn the gain up
until the system is just “ringing” - this is the stage just before feedback starts where the sounds
produced by a microphone take longer than normal to decay, producing an almost metallic,
chiming sound; this is the tone that will feed back if the gain is turned up any more. Now you
have the tone, find the offending frequency by dropping sliders on the EQ until it is removed;
listening to the pitch and some guesswork is required to find the correct slider, but with
experience, it is quite easy to find the correct “band” within a few attempts. This can then be
repeated for other feedback frequencies and so on. Do not let the feedback get overloud or the
speakers might be damaged; pure tones damage speakers more readily than other forms of sound!
Once more than half the sliders on the EQ have been moved, you will not get much more of an
improvement because the EQ is then reducing the gain of the whole signal rather than just at
specific notch frequencies.
The new Behringer Ultracurve equaliser can help automate some of this process with it
‘feedback-destroying’ mode. It can also be used with a reference microphone to set up a room’s
EQ. See the section in this guide, or the User’s Manual for details.
You may find that the feedback frequencies are multiples of each other (e.g. 100, 200, 400Hz). This
is perfectly normal, and occurs due to different resonant “modes” of a space. You may also find
some really complicated feedback paths that don’t just loop around the system once, but twice -
perhaps through different microphones. The classic example of this is the sound which loops
between the orchestra pit and the stage.
It is also worth remembering that the acoustic properties of a stage will change depending on the
audience size, the positions of set, and even large props. A microphone gain setting that is perfectly
acceptable in one act may cause feedback in another due to a scene change! On one occasion, a
large flat which was flown out close to overhead mics which were operating “close to feedback”
caused a new feedback path in one position whilst they were travelling; this caused a very short but
noticeable “blip” of feedback. The moral of such stories is that gain cannot be set to avoid feedback
and then forgotten, but must be checked throughout an entire rehearsal wherever possible!
Page 13 of 107
Radio Mics
“The most expensive radio mic in the world is only
ever almost as good as a piece of cable.”
Remember those words, and your life will be much easier. There aren’t many occasions that you
will need to use radio mics; when you do, they can cause you problems with interference, batteries
running out and physical fragility.
There are two frequency ranges in which radio mics work: VHF (173 - 175 MHz) which can
accommodate five channels, and UHF (usually channels 69 and 72) which can accommodate over
64 channels. UHF radio mics are less prone to interference but the same transmission power will
travel less far; they are also more expensive!
There are five VHF radio mic frequencies available for use without a licence. These are:
• 173.80 MHz (Yellow)
• 174.10 MHz (Red)
• 174.50 MHz (Blue)
• 174.80 MHz (Green)
• 175.00 MHz (White)
Unfortunately, Green cannot be used with any of the other channels as it causes intermodulation
problems, where the “carrier” frequencies interfere with each other. In rare cases, Red has proved
troublesome as well. Therefore you only have four channels available, unless you start spending on
licensing other frequencies. (If you hire radio mics then another two frequencies are available from
the hiring company, which may only be used indoors.)
If you use radio mics, ensure that the cast are well briefed in how to use them. Remind them that
they can be heard after they have left the stage and until they have switched the mic off (even
though it should be muted at the mixer by this point - sound ops sometimes forget!) and have a
stage hand ensuring that mics are in place and switched on before actors go on stage. If there are a
limited amount of mics to go around and the cast keep swapping them between acts, this will have
to be very carefully planned.
Radio mics can cause all sorts of problems with feedback - because the microphone keeps moving
around, the feedback frequencies keep changing, making it very difficult to remove. If you are
having problems, then one solution is to use a “feedback exterminator” : this piece of technology
automatically scans the microphone signals for signs of feedback, and “notches” it out before it
becomes noticeable. However, the output sounds slightly processed, and the scanning technique
only works well with stationary mics (e.g. floats); it is not fast enough to cope with fast-moving
radio mics as it takes a moment to recognise the feedback occurring. Top West-End shows don’t use
them...
Watch out for situations where two radio mics get close together - two miked vocalists embracing is
the classic example. Instead of singing into their own microphones, they will now both be singing
into both microphones. Not only does this cause the combined level to shoot up, but as there is still
a small separation between the microphones, phase cancellation occurs degrading the frequency
response.
Page 14 of 107
There are several places that a radio mic pickup can be worn. One of the best places is in the centre
of the forehead above the hairline; this allows the mic to be pointing at the actor’s mouth, without
picking up too much from the surrounding. If the mic is too obvious there, try placing it over one
ear, as far forward onto the cheek bone as you can - glasses make another excellent hiding place.
Stage make-up can easily find its way into pickups work on the head; one solution is to cut the teat
end off a condom (this requires steady hands and very sharp scissors!) and place this over the
pickup, as the thin material lets the sound in but keeps everything else out. It is not a good idea to
place the mic on an item of clothing, as damage can easily occur during a rushed costume change.
The transmitter is best worn on a waistband or belt, with the pack inside the clothes and the aerial
wire hanging loose.
The location of the radio mic receivers can make a huge difference. Ideally they want to be as close
to the cast as possible, but it is also useful to be able to see the signal indicators on the receivers to
aid in fault-finding; this usually means that receivers are best placed in the wings. In the ADC
Theatre, the receivers are in the sound box, though connected to the aerials by low-loss coaxial
cable, and via an amplifier.
Don’t be tempted to use a set of batteries for more than one, or one and a half shows. If the budget
can stretch to radio mics, it can also stretch to lots of batteries. Batteries can be bought very cheaply
from Sundries in the ADC Theatre. Always take the batteries out of the units when they are not
being used, as the residue from leaking batteries is very difficult to remove!
One overriding factor is that radio mics are very expensive. As an example, to hire one VHF radio
mic for one week costs around £60 plus VAT; though the ADC Theatre currently hires its 6 out for
£30 per week to internal productions, including new batteries for every one-and-a-half
performances. Add the cost of licences to this if you wish to use more than four channels, though
the ADC Theatre is licensed for a number of VHF frequencies.
Page 15 of 107
Sound Media
There are many sound media available. Each is a compromise between quality, recordability,
flexibility, availability, cost and ruggedness. The two main classes of media are analogue media,
where the sound is recorded as a continuously varying signal, and digital media, where the sound is
converted into a string of ones and zeros and then stored as blips of signal. Digital media are more
versatile and free from background noise, whereas analogue media can theoretically give better
sound quality, although this is difficult to achieve.
The sound quality on analogue media is highly dependant on how fast the media is passing the pick-
up - a higher speed improves frequency response. Different speeds are offered on vinyl and tape to
give the best compromise between sound quality and recording time from a given amount of media.
Don’t forget to mark recordings with the speed and to select the correct speed for playback!
When recording onto analogue media (cassette tape, open reel tape etc.), the best signal to noise
ratio will be achieved when the recording level shown in the unit is about 0dB - this level will get
enough signal onto the media to make the background noise negligible but not enough to make the
recording distort. When recording onto digital media, background noise is less of a problem, but
distortion sets in much more suddenly because digital media clips above 0dB; it is therefore wise to
aim for a lower recording level - anywhere from -10dB to -6dB is about right. This “margin of
error” between the operating level and the maximum recording level without distortion is called
“headroom.”
All media have a defined “bandwidth”, which is the range of frequencies which they will record.
For analogue media, this depends on the circuitry used, the design of the head / stylus / pickup, and
the quality of the media. For digital media, the main factor is the “sampling rate”, which is the
number of times per second that the signal is measured and stored in digital form. The theoretical
bandwidth of a digital media is from the very bottom of the frequency range (i.e. 0 Hz) to a
frequency of half the sampling rate (e.g. CDs use a sampling rate of 44.1 kHz, and can record
frequencies of up to 22.05 kHz); therefore, the higher the sampling rate, the wider the bandwidth
and the better the sound quality will be.
Always label recordings as soon as they have been made for easy identification. Many of the
recordable media also have a mechanism to prevent recordings from being erased. This normally
takes the form of a tiny slider (e.g. on a MiniDisc or a DAT) or a breakable tab (e.g. on a cassette.)
Whichever media is being used, ensure that the mechanism is clean. Dirty heads throw sound
quality away on analogue media, and can cause “skipping” and other errors on digital media, whilst
dirty rollers can cause the tape to be dragged into the innards of the unit causing irreparable damage
to the recording. The mechanism can be cleaned either by using a specialist head cleaning cassette
(some of which are better than others), or with cotton buds dipped in surgical spirit and a great deal
of care - tape decks do not like having tufts of cotton adrift in their mechanism!
Page 16 of 107
The following is a description of many of the media, how to use them and their strengths and
weaknesses:
Cassette Tape
There are several different types of tape used inside cassettes; they can be referred to by a “type”
number, by the material used to coat the tape, or by the “bias” setting required for that tape type.
The common types available are:
“Type” Coating Bias Advantages / Disadvantages

setting
I Ferric (“normal”) 120µS Cheap, but with limited sound quality, especially
at high frequencies. “Ferric” is the lowest grade,
“microferric” is higher quality and “high energy”
(cobalt doped) types are as good as cheap type II
tapes.
II CrO2 (Chromium 70 µS Better sound quality than type I tapes, but with a
dioxide) lower maximum recording level.
III Ferrochrome An obsolete combination of type I and type II
tape coatings.
IV Metal 70 µS Excellent sound quality, but expensive.
Few people realise the sound quality that cassette tape can achieve; a good metal tape with noise
reduction and a good cassette deck can sound better than the digital (but data compressed) sound of
a MiniDisc!
Cassettes are one of the hardest media to cue. Ideally, the recording should be made with a spoken
introduction, such as “Sound effect take 14 after three, one... two... three...” so that the sound
operator has a positive identification of what is being cued up and when it will start. It is relatively
easy to put a cassette deck into pause mode the correct time after the word “three” so that the effect
starts as soon as pause is released.
This technique is not possible when using pre-recorded cassettes, so a different method must be
used:
• ensure that the output from the cassette deck is muted on the mixer and select headphone
monitoring.
• play the cassette up to the desired starting point.
• as soon as the first sound of the desired section is heard, stop the tape.
• eject the tape and use your finger or a pencil to wind the tape back a quarter turn; to do this, hold
the tape the correct way up with the current side facing you, and rotate the left-hand spindle
clockwise for a quarter turn.
• replace the cassette in the deck and enter pause mode
• unmute the output from the cassette deck and check that the output level is correctly set
• release pause, and playback will start instantly.
Page 17 of 107
Some experimentation is required to determine exactly how far the cassette should be wound back;
this varies from cassette deck to cassette deck, and even between different places on the tape! Some
cassette decks are better that others for this sort of work - in particular, older, mechanical cassette
decks which hold the head and pinch roller back against spring pressure when paused start faster
than newer, electronic cassette decks, which use solenoids to move the heads.
For reference, cassette tape uses a speed of 17/8 ips.
To overcome the problem of background noise, several forms of noise reduction have been devised.
The best known and most standard of these are the “Dolby” systems developed and licensed by the
Dolby Laboratories, which boost high frequencies where tape hiss occurs on recording, and then
reduces them to normal levels (reducing hiss at the same time) on playback. There are several
versions of this system:
• Dolby A, which is normally only used in professional recording studios.
• Dolby B, which was the most popular version provided on domestic equipment.
• Dolby C, an improved version of Dolby B.
• Dolby S, a development which uses compression as well as pre-equalisation. Its frequency
characteristics are similar to those of Dolby C, and useful, but slight compression can be
achieved by recording in Dolby S and replaying in Dolby C.
Some manufacturers have developed noise reduction systems which do not need the tape to be pre-
encoded (e.g. DNL by Philips), but these are less standardised.
Most domestic cassette decks are referred to as “two head” machines, as they have a combined
record/playback head and an erase head. On many professional cassette decks, the record and
playback functions are assigned to separate heads, giving a “three head” machine. The advantage of
this arrangement is that the quality of a recording may be monitored as the recording is being made
by immediately playing the recording back via the separate play head.
Residual magnetism is an invisible enemy which both degrades sound quality and gradually erases
recordings. It is usually found on tape heads but can occur on any part of the tape “transport”
(mechanism), and is caused by the magnetic field on the tape being drawn past metal parts, the
effect of erase heads, or sometimes by external equipment. It can be removed be using a
demagnetiser, which is a unit with a long tip which can be rubbed over all metal parts inside the
tape transport. Remember to lift the demagnetiser as far away from the tape deck as possible before
switching it off, as this produces a pulse which is strongly magnetising, and keep it away from all
tapes, or it will demagnetise those as well!
Page 18 of 107
Compact Disc
Compact discs were introduced in 1984, and provided digital sound quality in the home for the first
time. Whilst not as indestructible as was first claimed, they are very robust, and are easy to work
with in a production environment - most radio stations use CDs for the vast majority of music. The
sampling frequency used is 44.1 kHz.
In normal use, compact discs suffer from a pause between the operator pressing the “play” button
and the playback starting. To overcome this, use the following sequence to cue up a track:
• ensure that the output from the CD player is muted on the mixer!
• select the track and press “play.”
• when the time display for the track appears and is counting, press pause.
• press the “reverse skip” (|<<) button to return to the start of the track
• unmute the output from the CD player and check that the output level is correctly set
• press “play” to release pause, and playback will start instantly.
This trick works by ensuring that the disc is already spinning and that the optical head is in the right
place - on fancy CD players with a built in memory, it also means that the first chunk of data has
already been read off the disc before playback is required! Note that this seems to work on virtually
all CD players - this means that even the cheapest players can be used as sound sources in the
theatre due to the inherently high quality of CD sound!
Some CD players (including those in the ADC) have an “Auto-cue” function which automatically
cues a track up and then pauses the player. After playing the track, the next track will be cued in the
same manner. This function can make CD-intensive work much easier. However, note that some
CDs are recorded with the breaks between tracks in the wrong places, leading either to an unplanned
silence or losing the first couple of notes off a the start of a track. Always check that “Auto-cue”
will work on a particular track before using it live!
It is now possible to record CDs cheaply. Single copies and small batches can be recorded using
CD-R technology; the recordable discs cost less than £1. This is an ideal way of taking effects on a
tour where MiniDisc players may not be available and cassette tape would be too fragile.
Digital Audio Tape

Although unlikely to be “the” new digital audio format, DAT
has filled the hole for providing recordable digital sound in
semi-professional applications. Many radio stations now use
DAT for convenient, high quality location recording and DAT
machines are becoming standard pieces of equipment in many
studios, but poor cueing accuracy and the time taken to find a
cue make it less suitable for live work in the theatre. The
sampling frequency used is 48 kHz, giving a theoretical sound
quality improvement over CDs.
Page 19 of 107
MiniDisc
The MiniDisc format is the leading digital
recording standard in theatre sound. The
discs themselves are similar in construction
to computer 31/2 inch discs, but smaller, and
with an optical disc instead of a magnetic
disc (see Figure 5.) They can be written to
and read many times without wearing out,
and are very robust. As with CDs, the
sampling frequency is 44.1 kHz. The discs
are available from audio specialists with a
recording time of either 60 or 74 or 80
minutes, and cost less than £2 each.
Figure 5 - A MiniDisc
Although there is an absence of background
hiss due to the digital recording system, data compression is used to squeeze all the sound onto a
limited space. Normally, this is inaudible, and the quality is easily good enough for theatre sound
effects; however, when recording music with a wide frequency range, and especially with pipe
organ music, this compression may dull the sound.
A MiniDisc stores sound in a way that is a cross between a CD and a computer disk. The sound is
recorded in blocks that last 1/75 second, each called a frame, in that same way that a CD is recorded.
However, on a MiniDisc the sequence of frames that makes up the track can be moved around; this
allows you to divide, move and combine tracks. All MiniDisc editing can be accomplished with
these three operations.
The following points are worth noting:

• MiniDiscs are limited to a maximum of 255 tracks.
• Consumer MiniDisc players employ SCMS (Serial Copy Management System), which prevents
digital copying of a ‘second generation’ recording. Professional players such as the ADC’s
Denon 990 ignore SCMS data.
• The MiniDisc cannot combine very short tracks; a group of small tracks can be combined by
recording them onto a good quality cassette and then returning them to the MiniDisc as a single
track.
• Chopping the silence off the beginning of a track will reduce the time between pressing play and
hearing something.
• Labelling the tracks with the cue numbers as well as a description of the track contents helps to
ensure that the correct sound effect happens at the right time!
• Wherever possible, avoid having to use continuous play mode during a show, as it is all too easy
to forget to put it back into single play mode again afterwards...
There is a guide to using the ADC Denon MiniDisc player / recorder in Appendix D (see page 79); a
copy of the manual is kept in the sound box for reference.
Note that “budget” MiniDisc recorders are now commonly available, but as with everything in this
world, you only get what you pay for! These machines will certainly have inferior editing features,
and are likely to have comparatively poor response times as well. However, if editing is not required
and tight cueing isn’t crucial, these machines can represent excellent value for money.
Page 20 of 107
Open Reel Tape

This was the original “high quality” media, and until recent developments in digital audio storage,
was the only serious way of recording and editing sound. It was popular for the following reasons:
• it could be easily cut and spliced to edit material
• it could be run at different speeds to give different compromises between length of play and
sound quality
• very high sound quality could be achieved on a decent machine and decent tape.
• the tracks on the tape could be used in different ways.
• it was what everyone else used.
Many ways of splicing tape have been devised, from using a razor blade and block, reel of splicing
tape and steady hands to jigs that cut the magnetic tape, and apply a pre-cut piece of splicing tape at
the pull of a single lever. The method used can only be determined by budget and personal
preference.
Plastic leader tape with no capability to store audio information is spliced onto the magnetic tape to
form the ends of a reel and the breaks and cueing points between items. This tape protects the
recording from dust and dirt and to identify the orientation of the tape. Many types can be written
upon to provide further identification and information. The following colours of leader tape are in
common use:
Leader Colour Use

Green Start of tape
Red End of tape
Yellow between items
Clear cue-point on optically cueing machines
Most open reel machine in theatres will use a quarter-inch tape. This can be split into either two or
four tracks; four tracks are more versatile (quadraphonic, two separate stereo tracks, dubbing etc.),
whereas two tracks give better sound quality. The common speeds are 17/8, 33/4, 71/2 and 15 inches
per second (ips); for reference, a 1,200 foot tape at 71/2 ips will run for 30 minutes. The BBC
standards are 15 ips stereo for music, and 71/2 ips mono for speech. The reason that mono is used for
speech is to eliminate the effects of two channels carrying the same information getting out of
phase, either partially (due to a misaligned tape head) or totally (due to a cross-connection.)
Finally, there are two pre-equalisation standards for open reel tape: NAB (National Association of
Broadcasters) and IEC (Internationale Electrotechnique Commission.) Ensure that the playback
equalisation matches that used in the recording for the optimum frequency response, and don’t
forget to note the pre-equalisation used by a recording on the tape label!
Page 21 of 107
Vinyl
There are rare occasions when a piece of music is only available on vinyl. Don’t even think about
using vinyl “live” - if the quality of the record in question is good enough, re-record the music onto
a more friendly format. If the record is scratched and the sound quality is poor, tactfully suggest to
the director that it may be wise to find another piece of music - assuming that the pops and crackles
aren’t meant to be atmospheric!
The ADC owns two record decks, which can be used to transcribe sound off a record onto another
media. If you’re recording onto MiniDisc or open reel tape, it is worth starting the recording before
starting the vinyl and then editing the recording to get a cue point. If you’re recording onto cassette
(or are in a tiny regional theatre which only has a gramophone!) you will have to cue the record by
hand.
Ensure that the turntable that you are using is as manual as possible; semi-automatic turntables
(which lift the stylus at the end of a record) are just about tolerable, but fully automatic decks
(which also lower the stylus at the start of a record) are completely unusable for this sort of work.
Use the following sequence to cue a record:

• ensure that the output from the turntable is muted on the mixer and select headphone monitoring.
• check the speed, check the side, and check the track - on most records, the tracks can be visually
counted as the periods of silence in between them show up as “flat” ring on the record.
• put the stylus into the silence at the start of the track
• gently rotate the deck by hand until the desired cue point is heard. Normally, this will be the first
sound of the track.
• rotate the record anticlockwise by about half a turn. This gives the record time to speed up; the
actual amount will have to be varied to suit the deck - the ADC decks need about half a turn,
standard “disco” decks (e.g. the classic Technics SLP-1210 Mark 2) need about a quarter turn,
and the “standard” BBC studio deck requires about a tenth of a turn.
• unmute the output from the turntable and check that the output level is correctly set
• start the deck on cue; there will be a slight delay before playback due to the amount by which the
record was reversed. If timing is critical, take this into account!
This requires practice, but can be surprisingly accurate, even with mediocre equipment. If you have
problems, find an experienced DJ and get them to demonstrate.
The common speeds for records are 331/3 revolutions per minute (rpm) and 45 rpm, although 78
rpm and occasionally 16 rpm used to be common.
Recordings on vinyl are “pre-equalised” to the RIAA (Recording Industry Association of America)
standard to reduce noise by cutting low frequencies and boosting high frequencies. This means that
record deck outputs should be connected to RIAA preamplifiers to restore the correct frequency
response; however, the levels are in the same order as microphone levels, so microphone and phono
inputs can be interchanged at the expense of sound quality.
Page 22 of 107
Sound Material
Sound material falls into several categories: These include:
• pre-recorded music
• voice-overs
• sound effects.
The ADC owns a library of sound effect CDs. These are listed in Appendix G (see page 90).
Cambridge Central Library in Lion’s Yard is a very good source of material. Members of the library
(it’s free and instant to join - just make sure you have some ID!) can borrow a wide range of CDs
and tapes covering all types of music and sound effects for a small charge; at the time of writing,
this is about 80 pence per CD. The Union Society also operates a music library, which you have to
be a member to use. Your friends and neighbours are also a useful source of material. Raid their
music collections for that really obscure piece of music, and ask for their advice about the choice of
music.
Some sound effects can be downloaded off the Internet (see page 58 for more details) and then
recorded onto a conventional medium from a suitably equipped computer or a synthesiser, but this
can sometimes be a long-winded method of getting mediocre results.
It’s quite fun to record really obscure or specialised sound effects yourself, but make sure that the
quality is suitable or these tend to sound “naff.” Generally, MiniDisc is the standard recording
media in theatres. Check for insidious background noises such as traffic and aeroplanes - these
might not be obvious when you play your recording back at the time, but will sound very out of
place in a Shakespearean play! Ensure that pre-recorded effects have been made as easy as possible
for the sound op to cue during the show by audible naming and counting into each effect; see page
17 for more details. Live recording is a fantastic way to learn about sound, and in particular about
the capabilities of the various microphones at your disposal. Note that you may have to find your
own equipment for live recording, as the ADC equipment is not normally available for this sort of
work unless not in use in the theatre.
Several sound effects are best run live by the stage manager. These include doorbells and telephone
rings - the ADC owns all the necessary equipment for this. Also, a “thunder sheet” hangs above the
stage manager’s position - it’s traditional and it’s fun to use! If your production uses a band, and
especially if there is a synthesiser involved, don’t forget that this can also be a useful source of
varied effects.
Some directors will give you a script marked up with all sound effects. Other directors will ask you
to suggest a list of sound effects for approval. The best sound plots stem from a combination of
these techniques. By all means give the director a range of material, but ask yourself what you are
seeking to achieve with each effect; gratuitous sound distracts the audience from the action. Don’t
forget that background effects, such as waves lapping on the shore, or twittering birds, can often set
a scene more effectively that hundreds of pounds worth of set - the producer will love you for
suggesting these effects! Very often, the director won’t know what he wants, and the description of
the music or effect which he gives to the sound designer will be rather vague.
Read the scene in which the material is used. Show it to people, and hopefully, someone will have
an idea!
Page 23 of 107
Finally, ensure that the sound effects are appropriate to the show. Musicals and comedies require
less subtle effects than serious plays, where (unless they are to illustrate a point in the script, such as
a clock chiming) the audience should only be peripherally aware of them. If a sound effect distracts
the audience, the atmosphere which the playwright has been trying to build since the start of the
show may well be lost - some shows may well be better without any sound design at all.
PPL and PRS

Be aware of the regulations regarding the Phonographic Performance Licence (PPL) and the
Performing Right Society (PRS.)
A Phonographic Performance Licence is held by the theatre; the hire fee paid by a show includes a
contribution towards the cost of this. The revenue from this is distributed amongst the various
record companies.
A Performing Right Society Licence is also held by the theatre; the revenue from this is paid as
royalties to composers and musicians and charges will be collected by the theatre according to the
following regulations:
• “Incidental” or “curtain” music is defined as being “music heard by the theatre audience as an
accompaniment to the play but which is not performed by or intended to be audible to any of the
characters in the play.” This includes background music played before or after a show. This is
charged at a flat rate per week.
• “Interpolated music” is defined as being “music not specifically written for a theatrical
production (and excluding overture, entr’acte, exit, incidental or curtain music) and which is
performed by or intended to be audible to a character or characters in that theatrical production.”
This is charged according to the running time of the music.
Note that the PRS royalty is based on a proportion of box office receipts before the ADC hire
charge is taken into account.
These arrangements and charges are specific to the ADC Theatre; other venues may have different
arrangements for PRS and PPL - be sure to check! Where known, PRS and PPL arrangements for
each venue have been described in Appendix A (page 59).
Page 24 of 107
Sound Processing
Sound processors are devices which modify an existing sound for creative purposes. As a general
rule, only those which involve some form of time delay, such as echo, reverb or chorus can also be
called “effects.” The reason for this difference is that effects are generally applied to part of the mix
and then fed back into the mixing desk, whereas the entirety of a signal is usually passed through a
processor.
A signal which has been passed through a processor is referred to as a “wet” signal, and one which
has not been processed is referred to as a “dry” signal.
The ADC Theatre currently owns a good number of sound processors; a reverb unit, a multi-effects
unit, a stereo compressor/ limiter/ expander, a four channel compressor, a digital graphic equaliser/
signal analyser and a standard graphic equaliser.
Figure 6 - The ADC processing & effects units
Page 25 of 107
The Reverb Unit

The basic reverb unit is a digital Yamaha REV100 unit. This offers good reverb and echo type
effects, at the expense of not offering pitch shifts or other exotic effects. This means that the unit is
very good at adding a natural-sounding “presence” and power to a singing voice. A small amount of
the right type of reverb can increase the impact of an actor’s voice immensely; however, too much
reverb will sound obviously tacky.
The unit offers about a hundred different “basic” effects, all of which can be minutely adjusted with
the controls on the front, which cover initial delay, reverb time, and high-frequency damping
parameters. Input level and dry/wet balance controls (which regulates the amount of processed
signal mixed with the unprocessed signal) are also provided - note that many effects require some of
the original signal to be present to get the desired result! MIDI control is also possible; this allows
complex changes to be made instantly, and even allows access to features not available on the front
panel. The theatre currently has no means of supplying these control messages; however, see the
section on MIDI on page 43. More information on the effects available from this unit is given in
Appendix E
Reverb Unit Instructions.
It is recommended that the reverb unit input signal is kept as high as possible (so the level of the
return fader can be lower to produce the same amount of signal) to avoid the small amount of hiss
that the unit can produce.
The usual way to patch the reverb unit into the sound desk is to feed it from a post-fade auxiliary,
returning to a spare channel or stereo return. Using a post-fade auxiliary ensures that the amount of
effect remains proportional to the input level fader. Effects can be connected to insert points, but
then the proportion of the effect in the signal is governed solely by the effects unit wet/dry control.
The Multi-effects Unit

The other effects unit is a Behringer Modulizer Pro DSP1224. This is a newer, more flexible and
less noisy machine that has many more effects beyond the simple reverb.
Basic operation:
Change basic effect Press EFFECT (lights up); use jog wheel.
Change effect See table below, then:
parameters Press VARIATION (lights up); use jog wheel.
Press EDIT A (lights up) then ENGINE L (or R); use jog wheel.
Press EDIT B (lights up) then ENGINE L (or R); use jog wheel.
Change high/low post Press EQ HI (or LOW) (lights up); use jog wheel.
EQ
Change wet/dry ratio Press EQ HI and EQ LO together (both light up); use jog wheel.
Load preset Ensure no buttons (other than store) are lit green by pressing any that
are; use jog wheel to select preset; wait 1 second.
Save preset Press store; use jog wheel to select location; press store.
Reset factory presets Power off; hold down STORE and EFFECT while powering up for 2
seconds.
Page 26 of 107
• Presets are effects saved
with user defined
parameters.
• Effects 1–9, 12–16 and

20–24 are stereo effects;
parameters apply to
both channels; ENGINE
L (and R) are only labels
for various common
parameters.
• Effects 10, 11 and 17–

19 are dual mono
effects; L and R
parameters can be
edited separately.
• Effects 25–32 are split

mono effects; different
effects are applied to L
and R.
Page 27 of 107
The Graphic Equalisers

Graphic equalisers work by splitting the sound spectrum into narrow, adjacent frequency band and
giving each its own cut/boost slider. The spacing of the bands is at intervals of a third of an octave;
significant as the ear is made up of 24 bands of cells (each called a “ ark”) which are sensitive to
different frequency bands spaced by a third of an octave.
Uses for this equipment include:

• fighting feedback, by patching it into the path of the offending microphone, and then removing
the frequencies that are causing feedback.
• removing “hiss” from a bad recording by notching out high frequencies to give a compromise
between noise reduction and dulling the sound.
• modifying sounds from microphones, and especially from the band. Percussion can be made
more or less heavy, and brass can be made brighter or sharper.
• distorting signals; for example, notching out all but 300 Hz to 3.4 kHz to make a signal sound
like it is being heard over a telephone.
Each channel of the equaliser is fitted with an in/out switch which allows the equalisation to be
taken out of the audio chain; this is useful for comparison of the raw and processed signal.
The ADC’s original graphic equaliser is a stereo 30 band Alesis M-EQ230 unit. It consists of two
separate channels that can either be used for both channels of a stereo signal, or for two different
mono signals. The controls are very simple, with a cut/boost fader for each band and an in/out
switch for each channel.
Recently, the ADC Theatre has bought a Behringer Ultracurve digital graphic equaliser/ signal
analyser. This features a 31-band stereo graphic EQ with a real-time analyser and an Auto-Q
function for automatic room measurement and correction (using a reference microphone) plus three
bands of parametric equalisation. Further highlights include a peak limiter, an adjustable delay of up
to 2.5 seconds, a noise gate and a Feedback Destroyer.
Page 28 of 107
Behringer Ultra-Curve Pro DSP8024P Basics:

• EQ activates graphic EQ (and/or parametric + feedback elimination) mode.
• RTA activates real time analyser mode (if RTA LOCK is off).
• SET UP activates EQ Set Up (in EQ mode); RTA Set Up (in RTA mode)
• Holding down SET UP for 2 seconds enters Global Set Up and Midi Set Up (use SET UP to toggle
between the two).
• IN/OUT inserts/removes the Ultra-Curve from the signal path.
• 4 soft-keys (A, B, C, D) to left of display perform function indicated on LCD.
• 4 Cursor keys at right of display scroll through and edit graphic EQ settings and navigate round Set Up
screens.
Edit graphic EQ Press EQ; EDIT; use cursor keys to select bands and boost/cut;
use L (or R) to toggle between L and R.
Zero EQ EQ; EDIT; TOOLS; ZERO.
Copy L to R (or vice versa) EQ; EDIT; TOOLS; L > R (or R > L).
Invert EQ EQ; EDIT; TOOLS; INVERT.
Use predefined shelving curvesEQ; EDIT; TOOLS; SHELV; use soft-key D to select curve; cursor keys to
edit; OK.
Load EQ EQ; EDIT; MEMORY; LOAD; cursor up/down to select location; OK.
Save EQ EQ; EDIT; MEMORY; SAVE; cursor up/down to select location; OK.
Compare EQ with last loaded EQ; EDIT; A/B.
Turn delay on/off EQ; DLY ON (or DLY OFF).
Change delay EQ; SET UP; use cursor keys to select; use soft-keys to edit; SET UP.
Edit parametric EQs EQ; FB D; use cursor keys to select; soft-keys to edit; set MODE to PAR;
set parameters; press EQ.
Turn on feedback destroyer EQ; FB D; use cursor keys to select; soft-keys to edit; set as many MODEs
on L and/or R (as appropriate) as possible to AUT; press EQ.
Turn RTA LOCK on/off Hold down SET UP for 2 seconds; press SET UP until you reach GLOBAL
SETUP; use cursor keys to select RTA LOCK; use soft-keys to edit; SET
UP. (Stops you from accidentally selecting RTA and pumping pink noise to
the main mix!)
Use RTA RTA; use soft key C to select source; use soft key D to set time constant.
Select RTA output signal RTA; SET UP; use cursor keys to select RTA OUTPUT; use soft-keys to
edit; SET UP.
Use RTA output as EQ curve RTA; MEMORY; RTA > EQ.
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Dynamic Processing
The ADC currently has a stereo compressor/ limiter/ expander and a four channel compressor,
which form an important set of tools to the sound engineer.
Dynamic processing is where the processor automatically adjusts the signal level. The most
common examples are as follows:
• compression, where loud signals above the threshold are turned down, so “compressing” the
dynamic range of the signal. This is useful for recording signals with a large dynamic range (e.g.
classical music) onto a media which has a lower dynamic range (e.g. cassette tape), or where a
reasonably constant level of signal is required (e.g. from microphones or instruments). See
Appendix I A guide to compression on page 95 for more information.
• limiting, which is an extreme form of compression. The average signal level is restricted
(“limited”) from increasing at all above an adjustable threshold; this is often used before power
amplification, recording or transmission of signals to avoid distortion.
• expansion, where quiet signals below the threshold are turned down, so “expanding” the
dynamic range of the signal. This is the opposite of compression, and can be used to decompress
a signal after it has been transmitted.
• gating, which is an extreme form of expansion. Signals below an adjustable threshold are turned
off or “gated” to reduce noise in silent passages.
A typical recording setup might first use a gate to remove noise during silences, then feed the signal
through a compressor to reduce the dynamic range and finally use a limiter / clipper combination
(often called a peak limiter) to prevent the tape from being overloaded by high levels with
consequent distortion. This is shown in Figure 7.
Recording
Mixer
Device
Gate Compressor Peak Limiter
Figure 7 - Dynamic processing in a typical recording setup
In each of these devices, the level of the input signal must be detected, so that the unit can make the
necessary changes to signal level. The part of the signal which is used for this detection is called the
“sidechain” - occasionally, an external signal is used in place of the main signal for special effects.
This arrangement is shown in Figure 8 and is discussed further in Appendix I A guide to
compression.
Input Signal Level Control Output Signal
Sidechain Input Level Detector

Internal / External
Sidechain Selector
Figure 8 - Main signal and sidechain signal paths in a dynamic processor
Page 30 of 107
The ADC Sound Desk (Mixer)

The ADC has Soundcraft SeriesTWO sound desk. It has 24 mono input channels and two stereo
channels and four stereo returns which can be mixed into eight submasters and thence to the main
mix, mono and matrix outputs. This arrangement either effectively gives five master outputs, or
allows the submasters to be mixed into the master outputs. In addition, there are eight auxiliary
mixes; this means that you can generate up to twenty one independent output channels!
Figure 9 - The ADC sound desk
There are three main parts to any mixer:

• input channels, where the signals are fed into the mixer, amplified and equalised and fed to the
busses
• signal busses, the internal parts of the mixer which collect signals from the input channels and
transport them throughout the mixer for use by other parts of the mixer, such as output channels
and the headphone facilities. Typical busses include the main mix, the pre-fade mix, sub-group
mixes and auxiliary mixes.
• output channels, which take the contents of the busses, control the levels and send them out of
the mixer whether to main outputs or monitor outputs such as headphones.
Figure 10 illustrates how these parts connect together:
Input channels Output channels

Pre-amplification
and equalisation Line drivers
Input level Output level

control control
Signal busses
Figure 10 - Block diagram of a simple mixer
Page 31 of 107
Input Channels
The ADC mixer has 24 mono input channels, two stereo channels, four stereo returns, a ‘replay’
stereo line input and a talkback microphone input:
1-24 Mono Two Stereo Four Stereo Return
line in socket L jack in socket L jack in socket

mic in socket R jack in socket R jack in socket
direct out socket
phantom on/off switch
insert socket
bargraph meter and bargraph meter and
peak warning LED peak warning LED
phase reverse switch
input -20dB pad
gain control gain control gain control
high pass filter on/off
high pass filter
frequency control
high frequency EQ high frequency EQ tilt (tone) control
high mid EQ (swept) high mid EQ
low mid EQ (swept) low mid EQ
low frequency EQ low frequency EQ
EQ on/off switch EQ on/off switch
send to Aux. 1 send to Aux. 1 send to Aux. 1 or 3
send to Aux. 5 send to Aux. 5
connect to subs 1&2 connect to subs 1&2 connect to subs 1&2
connect to mono mix connect to mono mix
connect to master mix connect to master mix connect to master mix
Pan Balance
direct output source
selection switch
solo (PFL or SIP) solo (PFL or SIP) PFL
mute mute mute
fader fader fader
Figure 11 - Input channel facilities on the ADC sound desk
Page 32 of 107
The mic input is via a low impedance balanced XLR connector, while line level signals can either
be connected via the XLR input or to the high impedance balanced ¼” line jack socket.
The rear panel connections to the input channels are connected to the main patchbay in the sound
box, for patching of mic and line level sources, and insert send and returns. However, the mixer
uses a switched jack socket to determine whether to use the XLR or jack input so it is necessary to
unplug the line input on the rear of the mixer in order to use the mic input.
The insert sockets allow the input signal to be put through an effects loop (i.e. hijacked, externally
processed by an effects unit and then returned to the mixer) on any channel. To use them, use a red
stereo patch lead to connect the insert socket to one of the four insert points on the left of the mixer.
The send and return points are then available at the patchbay. The advantage of using the insert
point rather than putting the signal through the effect unit before sending it into the mixer is that by
the time it has reached the insert point, the signal has already been through the mixer’s input
circuitry and gain control, so that whatever form the signal started off in (microphone, phono, line
level etc.) it is now a “tamed” line level signal.
There is a direct output on an impedance balanced ¼” jack socket on the rear of the mixer if
needed. A typical use would be to feed a microphone input to a second mixing desk which could be
used as a dedicated recording or monitor console.
Stereo channel inputs are by a pair of balanced ¼” jack sockets for each channel.
+48V phantom power is available on all mono XLR inputs, and is enabled by depressing the rear
panel switch for the corresponding channel.
Both mono and stereo input channels have dedicated 12-segment LED bargraph meters that read
the pre-fade, post-EQ input channel signal. The uppermost LED indicates peak level (PK) on either
the main input or insert-send signals, warning of clipping by lighting for a short period when either
of these reaches 3dB below clipping.
The -20dB pad switch and the gain control (SENS) allow adjustment of the amplification applied
to the signal entering the desk, accommodating signals from -60dBu to +26dBu. It is good practice
to use the gain control to set the working range so that with the fader set to 0 dB, the sound level is
about right, giving a good signal to noise ratio, a long fader length to perform smooth fade-outs and
some extra gain ‘in-hand’ if needed. The loudest signals should not quite illuminate the peak
indicator. When running a show, the op should not need to touch the gain control.
The pad button reduces the gain of an input by 20dB. This is useful (and usually necessary for line
level inputs) to allow the gain control to be used over a sensible range.
A phase reversal switch ( ) is also provided at the input, reversing the phase of the signal.
The switchable variable frequency high pass filter is very useful when a microphone is
connected, as it cuts out “floor noise” and rumble without affecting the desired portion of the signal
too noticeably. It also helps to prevent low frequency feedback, and even removes some mains hum.
The frequency threshold can be altered from 40 to 400Hz.
All input channels have a four band equaliser (EQ) control offering ±15dB of gain control. The
24 mono input channels (1-24) have two fixed frequency and two parametric EQ knobs, and the
stereo channels only have fixed frequency EQ knobs. The desk also has four stereo return input
channels, useful for inputs from effects processors and the like; these have very basic EQ facilities
limited to high/low frequency tone control (tilt).
The EQ available on each of the mixer channels is useful so long as you realise what its limitations
are. It is very useful for brightening up effects, mics, and so on, but it isn’t really selective enough
for fighting feedback - for that, you really need the smaller bands of the graphic equaliser. All mono
Page 33 of 107
channels are fitted with a four band “parametric” EQ: with this, not only can the amount of boost
and cut applied to parts of the signal be varied, but the frequency around which this occurs can also
be controlled. A good way to use this is to first boost the EQ and use the sweep control to locate the
‘worst’ sound then cut this frequency. If you’re not using the EQ, then do switch it out of the
channel using the EQ in/out button. It’s one less thing to go wrong!
The auxiliary sends feed mixes which don’t go through the normal output faders, and can be used
to feed effects units, foldback speakers and the like. Eight auxiliary mixes can be generated on the
desk, and these are taken from different points in the audio “chain”. All are switchable pre/post
fader in pairs. The PRE=PREQ button on each channel sets any pre-fade auxiliaries to also be
sourced pre-EQ. Pre-fade sends are taken post-insert. On the stereo returns, the aux. busses are
accessed from 4 send pots which are switched in pairs to toggle between which pairs of busses to
send to (e.g. 1&2 or 3&4), and all aux. sends are post-fade.
Each channel has a set of routing buttons which choose to which output busses the signal is fed;
there is a main group (mix) which feeds the input channel directly to the master output fader, a mono
output fader and eight sub-groups (which are selected in pairs) which feed the input channel to the
sub-master output faders. The sub-master outputs can either be used as outputs in their own right or
can be combined into the main outputs using the mono- and mix-send switches above the faders, so
that part of a mix (often logical groups of microphones e.g. band; overstage; radiomics) can be
controlled separately.
The pan control on a mono input channel allows the signal to be “placed” within a stereo mix by
assigning different proportions of the signal to the left and right channels (or odd and even sub-
groups respectively). This is why the groups are selected together by the “group” buttons. On stereo
input channels, this knob is marked balance.
Each channel is provided with a mute button (and LED activity indicator) which will silence the
output from that channel to all busses (including auxes) when depressed. If you’re not using a
channel, silence it! It keeps the amount of hiss in the system down, and in the case of mics, prevents
background noise (and indecent comments by the band) from being amplified. The mute control can
be operated locally or controlled by a mute group or MIDI snapshot from the master section.
The function of the solo switch is determined in the master section. The normal use is as mono “pre-
fade listen” (PFL), adding the signal on that channel to the “pre-fade bus” which can then be heard
over the headphones. This enables a channel to be heard without routing it to an output, and is often
used for cueing pre-recorded sounds; it is also a useful tool for listening to a single microphone in
the main mix to identify the source of any unusual sounds!
Alternatively it can be configured to trigger a “solo-in-place” (SIP), whereby the selected channel
alone is heard over the main mix output and all other channels are muted. The signal heard will be
at the post-fader level, and so heard ‘in situ’ along with any associated effects. Obviously SIP
should not be engaged during a performance as the entire audience will hear the output!
The DIR PRE switch causes the direct channel output to be sourced both pre-fader and pre-mute -
i.e. the output acts as a simple mic splitter.
Page 34 of 107
Other Inputs
The SeriesTWO also offers inputs for a stereo replay source such as a CD or cassette tape player
which can be routed to the main mix bus, and a talkback microphone (and oscillator) which can be
routed to any of auxes 1-4, mix&mono, matrix 1 or matrix 2.
Master Section
This area houses the four stereo returns (which have already been discussed), the monitoring
control, controls for the 11x2 output matrix, the final level controls for all outputs and the mute
scene control.
A bank of 8 aux. master knobs control the final output levels of each bus. An AFL switch for each
bus allows the to be monitored on the headphones.
Each group master fader has a number of controls above it. The mix and mono switches route the
signal to the corresponding master output, with the pan control determining the stereo position of
the signal in the mix bus. Usually the groups are used in stereo pairs and so even numbered groups
would be panned right and odd numbered groups left. A pfl switch is provided for monitoring of
each group. Each group (and also the mix and mono output) also has an insert point, allowing
effects or compression etc. to be applied across a whole group of mics.
The SeriesTWO offers an 11x2 output matrix. These are two outputs (matrix 1 and 2) that can
‘pick up’ signal from any combination of the 11 busses (the groups, mix left and right and the mono
output) and at any level, set by the corresponding knob. This allows two different outputs to be
assembled from the signal busses, perhaps for a good quality show recording which necessitates a
different mix to the main output.
All group, mix and mono outputs have dedicated 12-segment LED bargraph meters. The mono
meter switches to display PFL level when required to do so. The main L and R meters can be
switched to read monitor output instead, using the monitor to L-R meters switch.
The monitoring panel is used to select the default monitoring source (replay input, mix output or
mono output), which is overridden by any solo-ed source (PFL or SIP as described earlier) to allow
monitoring of just this source. The PFL/AFL LED lights when any pfl, afl or solo button is pressed
- an indication that the usual monitor signal has been replaced by another signal.
Mute System Operation

All input channel mutes can be operated locally by depressing the mute button, or from the mute
group switches or the mute scene control panel. When a mute is engaged, the corresponding status
LED for the channel will light.
The eight mute groups are created by enabling the individual channel mutes required then pressing
store together with the required mute master button. For example all the band microphones could be
selected, allowing one master switch to mute all those channels at the end of a song, without having
to press each individual channel’s mute button. These work independently of and normally override
the mute scene snapshot system.
Page 35 of 107
128 mute scene snapshots can be created. These could be used one per scene or song to mute (or
unmute) the required microphones for that scene/song. They are stored by enabling the individual
channel mutes required then selecting the memory to use with the up/down arrows, then holding the
store button and pressing the scene button. They can be recalled using the up/down arrows to select
a snapshot the pressing recall; or sequentially using the next switch. When scrolling through the
scenes, the LED display will flash the scene number unless that scene is the one currently in use.
Individual input channels may be set to be ‘mute safe’ by pressing its mute button while holding the
safe set button down - effectively removing them from control by the mute scene controller even if
programmed into scenes. Simply pressing the safe set/ view button causes the mute lights of any
‘safe channels’ to light - as default the stereo inputs and stereo returns are set as mute safe.
The scene control can be MIDI controlled if required, and when recalling of scenes causes a MIDI
program change signal to be sent.
To clear the desk’s memories, hold down next, scene and store during power-up.
[Much of this information is taken from the SeriesTWO user guide, a copy of which should
be found in the sound file]
Once the sound desk has been set up for a show, don’t forget to label the input and output channels;
even if you are intending to operate the desk every night, circumstances beyond your control may
mean that someone has to stand in at the last minute. Also, where several shows are running in rep.
(such as the main show and the late show at the ADC) several sound designers might be working
around the same system; it is polite to make the workings of your design obvious to anyone else
who may have to work with it. The simplest way of labelling the channels is to run a width of LX
tape across the desk under the faders and to write on that with a ball-point pen.
It is also a sensible precaution to make a rough note of patch, desk and outboard equipment settings
lest anything get altered by other users of the sound equipment.
When mixing (and especially mixing live music) use your ears and keep asking yourself the
following questions:
• Is it feeding back?
• Is it distorting or clipping?
• Can I hear every instrument / voice / sound source that I’m meant to be able to? In the right
quantity?
• Does it sound “natural”?
• Is the director (or musical director) pointing a loaded crossbow at me?
Finally, don’t fiddle with a mix for the sake of it - once something sounds right, leave it! Remember
the golden rule of mixing:
“If it sounds right to the audience, the mix is right.”
Page 36 of 107
ADC Sound Output

You’ve collected all the possible sources of sound that you (or the director) could desire. So what
do you do with them?
Auditorium Speakers
The most obvious form of sound output are speakers for the audience. These may be put in many
locations around the auditorium; a rig with two speakers in front of the perches is suitable for
simple work, and adding two speakers on the “rear of house” shelves makes the rig incredibly
flexible; you can now “fly” effects all round the auditorium and encompass the audience in a sea of
sound.
The Bose 802 speakers which the theatre owns are very versatile and quite powerful. They are good
for the ADC as they have a wide dispersion angle, useful where the audience is so close to the
speaker positions.
They give very high quality output if the signal being fed to their power amplifier is first put through
a Bose EQ box. This “predistorts” the signal to compensate for the frequency response of the
speakers, so that the sound output from the speakers follows the original signal very precisely. The
speakers can be used without this box, but the sound is noticeably coloured and the chances of
damaging the speaker cones with excessive levels is increased. If stacked in pairs, the Bose 802s
undergo ‘bass-coupling’ and the bass response is increased.
Stage and Band Foldback

Sound is more directional than most people think, and for musicals, opera and dance it is often
necessary to add “foldback” loudspeakers for the cast and band so that they can hear each other;
inadequate foldback often results in cast and band performing songs at different speeds and
consequently drifting away from each other - once this process has started, it is very difficult to
reverse!
Because foldback loudspeakers are close to the performers, they are often close to the microphones;
they are often responsible for feedback, so levels must be ruthlessly limited.
Remember that some band instruments don’t make any noise themselves, such as synthesisers with
no built-in speakers, or electric guitars that are directly connected instead of being fed through an
amplifier. These will need to be put into the band foldback mix in quite large quantities so that the
musicians can hear themselves play!
The stage foldback mic should include all effects that the cast use for cues, and the most important
band instruments - often the lead keyboard and drums are all that is required to give the cast enough
idea of what is being played.
Page 37 of 107
Sound to Light
Sometimes, a lighting effect will be synchronised to sound; there are three common ways of doing
this:
• the “sound controlled chase” consists of a sequence of lighting states which advances when the
sound level reaches a certain point. This is how most “disco” effects are generated.
• the “sound meter” controls the brightness of lighting to be proportional to the sound level.
• the “tone organ” flashes different lights at different frequencies.
When these effects are required, a sound feed is patched to the lighting desk via the main sound
patch. The feed must be carefully chosen to match the nature of the effect. For example, if a sound
controlled chase is being controlled by the beat of some music, this can be enhanced by providing a
lot of bass boost to the signal, or if an effect varies the intensity of light proportionally with a
microphone input, it will be more useful to provide a “clean feed” from that microphone alone than
it would be to provide the entire mix. Note that this microphone might not even be used for “sound”
purposes! For this example, instead of tying up a mix bus for this output, the signal could be taken
out of the microphone channel’s insert point.
Induction Loop
The theatre is fitted with an induction loop for the benefit of members of the audience wearing
hearing aids. When a hearing aid is switched to the “T” setting, instead of picking up the ambient
sound (such as the person behind them coughing) and amplifying it, it picks up the sound
transmitted on this loop.
The induction loop driver unit has two inputs; one is permanently connected to the show relay
microphone (which is flown from the auditorium roof to pick up sound from the stage) and the other
is a line level input available on the patchbay. By patching a convenient mix to this input, sound
effects and dialogue can be sharpened up for hearing aid wearers. There is a gain control on the
front of the unit to select the levels of the microphone input and the “direct” input from sound
sources. Don’t be tempted to turn the show relay microphone input right down, even if a show is
fully miked; research has shown that hearing aid users feel a sense of isolation if they cannot hear a
limited amount of noise from the audience around them.
To check levels on this unit, or to check that it is working, a test receiver for the loop is available
from the Technical Manger.
Page 38 of 107
The Perfect Cue Sheet

A cue sheet is a compete list of what an operator must do during a show, when they should do it,
and how they should do it. In the case of sound, all effects and any sudden mic operations should be
listed on the cue sheet. The following information is required:
• what is to be done - in the case of an effect or music cue, which track on which piece of media
is to be played; in the case of a mic cue, which mic is to be altered, and in what manner.
• what it will be called - the Stage Manager will be calling all the cues over the cans system. They
will refer to the cue by a specific number.
• when it is to be done - the exact timing should be given by the Stage Manager (but beware -
even SMs make mistakes sometimes!); however, it is useful to have some warning of when the
operator will be required to do something.
• what levels are to be used - every setting on every knob to be altered should be recorded.
As an example, here is the part of the cue sheet for the opening scene of “The Portrait of Dr.
Pattenden”:
Cue When Level media Description

1 start of Act -10 dB MD sound of bustling students
track 1
2 after “Come, let us 0 dB CD organ music. Fade as lights
enter the Chapel” track 14 come up.
3 after “what’s down set wet/dry N/A add reverb to vocal mic
this deep hole then?” to 20% on
reverb unit
4 after “nothing set wet/dry N/A kill reverb on vocal mic
much!” to 0% on
reverb unit
5 Dr. P. looking at his 0 dB MD clock striking 4.
sundial. “I wonder track 2
what time it really
is?”
6 Higgs picks up note fade to N/A kill stage mics as lights fade
and reads “Junior - dB on at end of act.
members are channels
reminded that 14-16
perambulation on the
structural
integuments is not...”
The test of a good cue sheet is that it should be all that a sound op needs to be able to run a show.
Page 39 of 107
Recording Shows
Audio
The quickest way of making a recording of a show is to use the show relay microphone that hangs
above the audience. This has a socket in the patchbay, but needs to be amplified before being sent to
the recorder - this is where the auxiliary channels on the desk become useful! However, the show
relay systems tend to buzz if connected to the sound desk like this – probably some earth looping
problem. If a better quality, or a stereo recording is required, some microphones will have to be set
up and mixed to make the recording. It may be easier to use the microphones that are already in
place for the show, and then create a different balance of sound on an auxiliary mix to compensate
for not being able to hear direct sound. If there are no usefully placed microphones, a PZM on each
side of the Proscenium (“Pros”) arch can give a good stereo field and pick up over most of the stage
area.
It is easier to make recordings onto cassette than onto MiniDisc for the following reasons:
• a MiniDisc is only 74 minutes long (148 minutes in mono) - most shows are longer than this
• being a digital medium, MiniDiscs distort severely when subject to excessive levels. Cassette
tape, being an analogue medium, will undergo distortion, but not so noticeably. When mixing a
live source directly, unless the budget will stretch to a compressor, there will always be
something which makes the levels shoot up unpredictably, be it an enthusiastic crescendo from
the band or an actor who suddenly decides to shout - plan for it!
Note that the ADC cassette deck has a feature where two recordings can be made at the same time,
so backup tapes can easily be made.
Video
Since the ADC already has a video camera pointing at the stage, making video recordings of a show
is not difficult. A SCART socket on the back of the SM’s desk is fed with a clean video signal taken
from the camera and a stereo audio feed taken from the jack patch on the amplifier rack. You will
need to find a video recorder for yourself; various hire shops will hire you one for a weekend. A
sound feed will need to be created as with an audio recording, sent down two spare tie-lines to the
sound shelf, and then patched to the SM’s desk at the patch on the amplifier rack. The video
recorder should be plugged into a sound power socket.
The Law
The law of copyright is quite clear - you are not allowed to make either audio or video recordings of
shows without the written permission of the copyright holder. However insistent the director is,
don’t risk a heavy fine!
You are not allowed to make any recordings of shows at the ADC without asking the Manager first.
If the show has not been copyrighted, then permission is easy to obtain.
Page 40 of 107
Mixing the Recording

Don’t leave it to the last night before trying to mix the recording. A few nights’ practice beforehand
will result in a much better recording. It is also helpful to have a second person mixing - so that one
can concentrate on running the show, and the other can concentrate on the recording. Good quality
headphones are acceptable for this type of work, but watch out for the lack of bass response inherent
in most headphones - the authors have a really good show recording that was spoilt by an excessive
amount of bass guitar in the mix.
It is usual to compress recordings for domestic use; this allows the listener to enjoy all the sound at
a comfortable level without permanently reaching for the volume control. It also boosts “quiet”
passages to the point that they can be heard against background noise, such as in a car. One simple
way of doing this is to record using Dolby “S” noise reduction (see page 17) and play back using
Dolby “C” noise reduction. The main difference between these formats is that Dolby “S” uses
compression; by playing back in Dolby “C”, this can be heard in the reproduced sound.
When distributing copies of a recording, remember that many people only have Dolby “B” facilities,
so ensure that the distribution version of the tape can be played in this format!
Page 41 of 107
The Band
A Happy Band...
... is a helpful band. Sound crew have to work very closely with musicians during stage shows
because your reputations depend upon each other. Inspiring confidence and relaxing the band go a
long way towards producing really good performances.
Make the band feel wanted by having the basic facilities set up on the pit or platform before they
arrive (ask the Musical Director how he would like it to be set out beforehand!) Don’t forget to
install foldback speakers, adequate lighting, cans, cue lights, sound power and as many video
monitors as are available - this last item is particularly important if the band is in a covered pit,
since they won’t be able to see anything for the entire show.
Miking the Band

Bands often need to be miked, either to assist the musicians in achieving a suitable balance between
the instruments or to overcome the limitations of putting the band in a (covered) pit.
Fully miking a band can often obtain spectacular results; however, it does involve a lot of time and
money to supply and set up at least one microphone per musician and then mix that many mics.
Usually fewer microphones than this are necessary to bring all the instruments into balance, sharpen
up the sound, and project it into the auditorium.
The AKG C1000s & Beyer M201s are very useful in these situations as they have a narrow range.
This means that the clarinet (for example) can be picked out from the rest of the instruments. A
normal sized band for a musical is only six or seven musicians - so only a little help is needed.
General miking to improve the quality and clarity of the sound is sometimes needed if the band are
in the pit.
It is also important to make sure that all the electronic instruments (such as keyboards and guitars)
are plugged into ‘sound power’ sockets and are connected to the sound system through a direct
injection box. Some guitar amps have DIs built in, but most don’t. If you can find an output that is
not affected by the musician changing the volume of the amp, then mixing is easier - even the best
sound op cannot deal with sudden changes of volume from an instrument (usually due to the
musician feeling underappreciated and turning himself up) without spoiling the mix slightly! Even
when using outputs which aren’t affected by the local volume control, it is important to emphasise
that the volume must not be turned up excessively or the sound will start to spill over into other
microphone channels, and in extreme cases can be heard in the auditorium! One over-loud musician
will also start the complaints from the rest of the band that THEY can’t hear themselves and the
endless pleas for more foldback (see page 37) will begin...
If the musicians can be persuaded to bring their own amplifiers, then these can be used in place of
foldback speakers in the pit, freeing up other speakers for use elsewhere.
Page 42 of 107
Mixing the Band

The key philosophy of mixing is to ensure that the music is heard by the audience as the composer
and musical director wish it to be heard. This means ensuring clarity without drowning the dialogue
or singing on stage. Mixing is best done whilst not wearing cans, but this makes cues difficult to
hear, so use the cue lights. Keep a cool head, and it will sound fine.
The hardest part of balancing the sound is keeping the band quiet enough for the cast to be heard
clearly. In all musicals the priority is on the words being sung rather than the notes being played.
Most musicians can play quietly but have to be coaxed into doing so - a quiet drummer is a very
rare beast!
Finally, make sure you send plenty of the lead instrument (usually the keyboard) into the stage
foldback mix so that the cast can keep in time with the band - but watch out for feedback!
MIDI
MIDI (Musical Instrument Digital Interface) is a digital standard for linking instruments and other
audio equipment together so that they can control each other by sending information which can be
broken into three main groups:
• note on/off messages, which say which notes have been played and how loud they are
• control messages (sometimes called “patch changes”), which control other aspects of the
instrument such as which sound to make
• data messages (sometimes called “bulk dumps”) which are often contain equipment specific
information, such as how to produce a particular sound.
So why is that useful, and why has it been brought up in a sound guide?
Imagine that you have an expensive synthesiser which has two sounds that you wish to access at the
same time. You have two options: either “split” the keyboard, so that part plays one sound, and part
plays the other, which is limiting, or buy a second synthesiser, which is expensive. With MIDI, a
second keyboard could be linked to the existing synthesiser so that each keyboard can play one
sound.
Imagine that there is a point in a song where the keyboard player cannot reach a button for a sudden
modification of the sound, or that the modification is too complex to be done instantly. With MIDI,
the required control message could be sent from an external source (e.g. by the sound operator!)
whilst the musician concentrates on playing. An external data source can also be used between
songs to totally reprogram an instrument.
The reasons that MIDI has been described here are as follows:
• the sound operator may be called upon to send MIDI data to the pit on cue. This is quite common
for West End musicals.
• MIDI is fast becoming a standard which will control items in the sound box: the ADC effects
unit is MIDI compatible, and is much more versatile when controlled from MIDI. The mute
section on the mixer is also MIDI-controllable.
• For awkward bits of sound design, a MIDI solution might be appropriate.
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• When it all goes wrong, the sound operator is likely to be the first port of call for help...
Synthesisers and electric pianos are the most usual instruments to be compatible with MIDI but
other instruments have been adapted: “guitar to MIDI converters” are common, and MIDI oboes are
not unknown! Computers, effects units, and some mixers and even lighting desks can also be linked
into MIDI, as well as MIDI-specific devices such as “sequencers” which record and play back MIDI
information and “expanders” which can best be thought of as the bit of a synthesiser which makes
the noise but without a keyboard attached.
Physically, MIDI is sent down 5 pin DIN leads, although only pins 2, 4 and 5 are used - this means
that with suitable adapters, MIDI can be sent down 3 pin XLR cables and even through the XLR
patch. MIDI in and MIDI out are presented on separate connectors, which makes routing MIDI data
easier, and MIDI thru ports are often provided to replicate the data sent into MIDI in so that several
instruments can be linked to one output; however due to the delays caused by internal circuitry, it is
more reliable to use a MIDI splitter box if many instruments are to be “chained” together. The
difference between these two schemes is illustrated in Figure 12 and Figure 13.
Out In Thru In Thru
MIDI Device 1 MIDI Device 2 MIDI Device 3
Figure 12 - MIDI device 1 controlling MIDI devices 2 and 3 via a thru port
MIDI Splitter
Out In In
MIDI Device 1 MIDI Device 2 MIDI Device 3
Figure 13 - MIDI device 1 controlling MIDI devices 2 and 3 via a splitter box
Further discussion is beyond the scope of this guide; however, it is worth becoming familiar with
MIDI, as it can often provide a neat solution to otherwise awkward problems. There is a book,
Theatre Sound by John Leonard, in the Club bookcase which gives a good coverage of MIDI in
theatres.
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ADC Sound Wiring

Fixed Wiring
The ADC has a huge amount of fixed wiring to enable microphones and speakers to be connected to
the sound box and amp rack from most places in the theatre with the minimum of “gash” cabling.
Connection points are provided in the following places, illustrated in Figure 14:
Location Microphones Speaker

Stage Box A1-15
Pit B1-8 US = upstage (rear)
OP ladder C1 MS = midstage
PS ladder D1 DS = downstage (front)
OP perch E1-2
PS perch F1-2 PS = prompt side
OP juliette G1-2 (also called stage left)
PS juliette H1-2 OP = opposite prompt
DS OP I1-4 (also called stage right)
DS PS J1-4
MS OP K1-2 FoH = front of house
MS PS L1-2 RoH = rear of house
US (left) M1-2
Counterweight gallery N1-2
Hemp gallery O1-2
Dome P1-2
Bar Zero Q1-4
FoH OP
FoH PS
RoH OP
RoH PS
M
Upstage
A
K Opposite Prompt
Prompt Side L
N
to Green Room
Downstage
I J O
G B H
E Forestage / Pit F
Q
C Auditorium D
Figure 14 - Plan of ADC Stage showing connection points (not to scale)
The stage box is a metal box fitted with 15 XLR sockets which attaches to a multi-pin connector
(MS PS on the wall below the amp rack) using a 20 metre multicore cable; it is very useful for
providing enough channels to mike a band which has not been put in the pit.
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Cabling
The ADC Theatre has colour coded cables. Use the right one!:
Colour Purpose Storage location

Grey or black Mic cables and adapter leads Sound shelf cupboard
Purple or red XLR patch in the sound box Sound box patch lead rack
Red Jack to jack leads Sound shelf cupboard
Grey or black 1/4 inch B-gauge jack patch in the Sound box patch lead rack and amp rack
sound box and on the amp rack patch lead rack
Blue Intercom (cans) cable Sound shelf cupboard
Green D54 (i.e. lighting control) cable Sound shelf cupboard
Black Speaker (Speakon) cable and patch Sound shelf cupboard (long leads) and
leads amp rack patch lead rack (patch leads)
White or Video Distribution Sound shelf cupboard
brown
Figure 15 – The Sound Shelf Cupboard
Currently, the theatre has a decent selection of adapters (such as phono to jack converters); however
if you can’t find what you need and if you give the technical manager some notice (at least a week),
then he will probably be able to make you an adapter lead. An hour before the first performance is
not the time to find that you need a 5 amp plug to phono plug adapter - if there ever is such a time!
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The ADC has various lengths of “cable rubbers”. Use them to protect any cables which you lay
across the stage, especially if they are where actors might trip over them - the cable fits inside the
rubber via a slot on the bottom, and then the rubber is gaffer taped to the stage (see Figure 16). If the
cable rubber is out of the audience’s view, it is helpful to lay white gaffer tape over the top to
improve visibility. Don’t gaffer tape cables directly to the stage - it provides minimal
protection for the cable and the tape is almost impossible to remove from the cable afterwards!
White Gaffer Tape for visibility
Cable Rubber
Cable
Gaffer Tape
Floor
Figure 16 - How to fit a cable rubber
There are also some pieces of black lino for laying over large numbers of cable, which can be
gaffered down similarly.
On larger or more complex sound rigs, labelling cables can help to speed up faultfinding and
modifications and is absolutely essential when repatching is likely to take place during a run; a
convenient labelling method is to write on lengths of LX tape using a ball-point pen and attach these
to the connectors.
Try not to run sound cables too near to any other type of cable, and particularly parallel with them.
In certain situations, you can pick up amazing amounts of hum from lighting cables. Running mic
cables too close to dimmers (especially the moveable “Act 6” dimmer packs) can also cause all
kinds of interference. You may notice that Parcans do audibly hum because of the way that their
filaments are made - they also generate much more interference than any other type of lantern. If the
problem is bad, try using Starquad cable (marked with the letters “SQ” along its length); this cable
is designed to reject even more interference than normal microphone cable and quite often it will
solve your problem. Another useful technique for removing hum on microphone channels is to use
the variable threshold-frequency high pass filter on the mixing desk.
Appendix B
Sound Connectors gives details of the various types of audio connector to be found, how to choose
them and how to wire them, and Appendix C
Sound Cables describes the most popular types of audio cable.
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Patchbays
You’ve got microphones. You’ve got all your “signal sources” such as CD players and the
MiniDisc. You’ve got the mixer, the amplifiers and stacks of inviting-looking shiny black boxes. So
how do you connect them together?
In very simple setups, you simply plug the output of your sources into the input of the mixer and the
output of the mixer into the input of the amplifier. There are several problems with this approach:
• all the different pieces of equipment use different types of connectors
• all the different pieces of equipment are in different places
• all this plugging and unplugging wears out the connectors on the equipment
• a mass of untidy cabling soon accumulates
• the connectors are on the back of the equipment so you can’t see what you are doing!
As the system grows larger, these problems become worse. The solution is to use a patchbay, which
is a panel covered in identical sockets. Each piece of equipment is permanently wired back to the
patchbay so that each socket on the equipment is replicated on this panel. This has solved all the
previous problems - all the connectors are the same type in the same place, the connectors on the
equipment itself are never touched, and the panel is easily visible. To connect equipment,
patchcords are used to link the sockets.
The sockets themselves are usually “B” gauge jack plugs, which are like headphone connectors, but
with a more knobbly end - they were developed by the Post Office for ultimate reliability on manual
telephone switchboards (another type of patchbay!) and are often referred to as P.O. 316 connectors.
Some budget patchbays are built to use “A” gauge jacks, which are normal headphone connectors.
Don’t plug “A” gauge jack plugs into a “B” gauge patchbay as it will damage
the sockets. XLR plugs and sockets and tiny “bantam” jack connectors are also used. Speaker
patches are used to link high-current amplifier outputs to fixed speaker wiring; these normally use
Speakon connectors.
Obviously, some pieces of equipment are normally connected to each other; the mixer output is
normally connected to the amplifier input, the CD and MiniDisc players are normally connected to
some of the mixer inputs, and a pair of the mixer groups outputs are often used to feed the recording
input of a MiniDisc player. To cut down on the amount of patchcords, a technique called normalling
is used to connect certain pairs of sockets together by default. This involves the sockets on the panel
being carefully arranged so that a row of output sockets is above a row of input sockets, with the
order of sockets in the rows arranged so that the pairs of outputs and inputs so-created form the
basis of a “normal” setup, so that only changes to this “normal” setup require the use of patchcords.
There are two types of normalling:
• “Fully normalled” pairs are connected together when both sockets are empty, and become
separate inputs and outputs when a plug is inserted into either socket.
• “Semi-normalled” or “half-normalled” pairs are much more common - inserting a plug into the
bottom (input) socket cuts out the default input as before, but inserting a plug into the top
(output) socket simply splits the feed, so the signal is fed to the default input as well as down the
patch lead. This is sometimes called “sniff and break” as the top socket allows the existing signal
to be “sniffed” whilst the bottom socket allows the signal path to be broken. (Double-normalling
is a variation on this where either socket splits the feed, but is rarely seen)
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Normalling is usually only available on patchbays fitted with one of the types of jack socket, as few
other types of socket are supplied with “break” contacts, which are used to link the sockets
internally.
The ADC sound system includes:

• an XLR patchbay in the sound box linking all the microphone wall boxes to the mixer
microphone inputs
• a “B” gauge patchbay in the sound box, linking all other sources to and from the mixer, including
eight “tie-lines” leading to the amp racks and four tie-lines to the XLR patch. Half-normalling is
used here.
• a “B” gauge patchbay in the amplifier rack, linking the tie-lines with the equaliser boxes and the
amplifiers. Also has connection to the SM’s desk SCART socket for recording shows on video.
Full normalling is used here.
• a “B” gauge patchbay in the amplifier rack for patching and re-routing the cans system. This is
normalled, and should not need to be used. Full normalling is used here.
• a Speakon patchbay in the amp rack, linking the amplifier outputs to the speaker connectors on
the wallboxes.
The ADC patchbays are also provided with “parallel” connectors to split signals – though half-
normalling in the sound box usually avoids their use. Diagrams and other details of the patchbay
layouts are given in Appendix F (page 86).
If several large shows are running in repertory, it may be necessary to re-use some of the audio
resources (e.g. mixer channels and effects units) in different ways. This is achieved by repatching
between shows (or in extreme examples during the interval or even the show...) To minimise the
chances of this going wrong, make sure that all leads involved in the repatch are fully labelled,
ensure that a single person has the responsibility for carrying the repatch out every night, and always
liase with other shows if you are repatching their setup - and out of courtesy, always leave their
patch as you found it, however illogical their routing might seem.
Unusual Patchleads
There are two variations on the normal patchlead.
• The “phase inversion” patchlead connects reverses the “hot” and “cold” connections between its
ends, allowing the phase of any balanced source to be inverted. This is useful when “phasing” is
occurring between two microphones due to their separation, or to cause phasing between
foldback speakers (which minimises low frequency spill into stage and band mics) or just as a
cure for non-standard wiring in hired-in equipment. A final use is for generating surround effects:
invert the phase of the rear pair of speakers and add a few milliseconds of delay, and the sound
field becomes much more realistic. Use with unbalanced outputs is not recommended as the
result will depend on the patchbay wiring, and may cause a short circuit. Phase inversion
patchleads are identified by the use of yellow cable and are often simply referred to as “the
yellow patchlead.”
• The “earth lift” patchlead has a break in the screen connection at one end. This can be useful in
removing earth loops (see page 50)
At present, the ADC Theatre does not have any of these patchleads, though the mono input channels
of the mixer provide a phase-reversal switch.
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Hum and Interference

Hum
Hum is the sound designer’s worst nightmare. You may have a set-up that works perfectly in all
respects - apart from the low droning sound that persists in the background.
Where does it come from? Why?
Electricity is supplied at a frequency of 50 Hz. This is an audible frequency. So are its harmonics
(multiples of 50 Hz.) This means that if anything electrical interferes with the sound system, you
will hear it as a low hum. To avoid this, the golden rule is to avoid generating earth loops. These
occur when two pieces of sound equipment are connected so that a current can flow from one to the
other through the screen of the signal cable and then back again through the earth connection in the
mains lead. For an example, see Figure 17.
Signal Cable
Guitar Amp Mixer
Mains HUM Mains

Lead LOOP Lead
Mains Wiring
Figure 17 - How a hum loop is generated
This is a problem because the loop starts acting like a transformer winding; current is “induced” in
the wire by the currents in adjacent mains cables. This causes hum.
OK. How do I deal with it?
Don’t panic. To get really bad hum, you need very large loops, and as all the ADC sound power is
fed separately and “star-earthed” (all the earths are fed back to a single point), the maximum size of
the loop is quite limited. If you do get hum, try using a direct injection box. These contain little
transformers which allow the audio signal to get from one place to another without actually making
a direct connection. They also come with an “Earth Lift” switch, which physically breaks the earth
connection between two pieces of equipment; these are worth trying if you are having real
problems, but can sometimes cause more problems that they solve. Earth lift can also be achieved
by using special patchleads: see page 49 for more details. Don’t ever remove the earth connection of
the mains power lead!
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Interference
One prime source of interference is the mains supply. Switching heavy loads (and especially
inductive loads) such as motors, fluorescent lighting and transformers causes “transients” which are
heard as crackles. In the theatre, the most important source of interference is the lighting system,
which not only draws heavy currents but switches them a hundred times a second to achieve
dimming - the worst case is when the entire rig is being run at medium intensity, as this requires
very aggressive switching of the supply.
The ADC was rewired in 1994, which improved the situation immensely. Sound power has been
separated from all other power supplies, and is filtered at each of the special sockets for powering
sound equipment to eliminates crackle from other equipment. You will need a special adapter to
plug anything into a sound power socket; these can be found in the Sound Shelf, and consist of a
four way plugblock with a round, blue, 16 amp plug at the other end. The sockets can be found in
the orchestra pit, upstage, in the green room, and behind the Stage Manager’s desk. Sound power
sockets in the sound box are of the normal 13 amp type. Outside the theatre, plug-in filter units can
be very effective, and are a “must” for touring shows.
Note that mains-borne interference is even more of a problem with modern digital or
microprocessor-based equipment; whereas older equipment would simply have produced a crackle
and continued working, more modern equipment has the propensity to “crash” (i.e. lock up until it
can be reset), behave erratically, or simply stop doing whatever it was doing. This makes effective
filtering essential when using this type of equipment.
Another source of interference is crosstalk or co-channel interference, where a signal breaks through
from one conductor to another physically close conductor. This sometimes occurs on tie-lines,
where the same conductors run next to each other for many metres, or within equipment, where
individual screening has been replaced by the overall screening of a metal box. To prevent this,
avoid running high level signals next to low level signals (e.g. a line level signal in a multicore
cable with mic level signals) use screened cables wherever possible (to minimise pickup of external
signals) and use balanced lines for long runs as they can “reject” any interference which is picked
up. Note that in extreme cases, running a mic cable next to a speaker cable can cause feedback, and
putting a very high level into one mixer channel can cause co-channel interference with adjacent
channels - try using the pad in this case.
A further source of interference is radio frequency interference (RFI) which is caused by powerful
transmissions close to equipment. These are often caused by taxi radios, but mobile phones can also
be a cause, so keep them away from sound equipment. Remember that digital phones transmit their
status to the cellular network every ten minutes whether they are being used or not, so to be safe,
switch them off totally when they could cause problems. To cure problems from RFI, follow the
same measures as with co-channel interference.
Intermittent crackling can often be traced to defective connectors. A common fallacy is that when a
connection fails, silence will be re result. In fact, a failing connection can be responsible for
crackling, fuzz, low signal levels, and even radio reception.
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If a sound system is humming smitten by interference, working through the following steps should
help:
• Check that all pieces of sound equipment are plugged into the sound power supply. For example,
make sure that the electric guitarist hasn’t plugged his amp into the normal supply in the
orchestra pit.
• Check that there is nothing plugged into sound power that shouldn’t be. For example, have the
musicians taken a fan down into the pit? (However, music stand lights are acceptable as they
generate no interference.)
• See whether closing all the faders on the mixer helps. If so, is it something audibly buzzing or
generating interference close to a microphone? (Air conditioning, lights, transformers, etc.)
• Unplug each of the sources from the patchbay in turn (remembering how it was all connected!) If
this cuts the hum and closing the fader didn’t, you have an earth problem. Use a direct injection
box.
• Replace every (accessible) lead in the sound system in turn. Try using different tie-lines.
Anything can happen if you have a dodgy connection - dodgy mic cables make wonderful radio
aerials, and Radio Four uses very powerful transmitters which all too often seem to be on the
right frequency...
• As a last resort, check every piece of equipment by replacing it (if a replacement is available) or
bypassing it. On rare occasions, humming can be caused by faulty power supply units providing a
badly smoothed DC supply, especially within power amplifiers.
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Balanced Connections
The simplest way to connect two pieces of equipment is to run a wire between them that follows
what the sound level is doing (the signal wire which forms the core of the cable) and another wire
that acts as a reference point (the earth wire which forms the screen of the cable.) This is called an
unbalanced connection, and is what happens when you wire your hi-fi together. Whilst perfectly
adequate for short cable runs, this method tends to pick up as lot of interference on longer cable
runs, and for very low level signals such as microphones.
In the 1930’s the BBC invented the balanced line. This was a beautifully simple technique to cancel
out interference by using a double-cored cable. Somewhat simplified: half of the original signal is
sent down one core. The other half is “flipped over” so that the signal goes negative with respect to
the earth wire instead of going positive, and vice versa, and this signal is send down the second core
wire (see Figure 18 and Figure 19.)
Flipped signal
Original signal
Figure 18 - An audio signal Figure 19 - An audio signal and its inverse
As the signal travels to its destination, both cores experience exactly the same conditions, so they
pick up exactly the same interference (see Figure 20.) At the destination, the second core is again
“flipped over” so that the signal is now the same as in the first core. The difference is that the
interference is now the other way up, and so by adding the two halves of the signal together, the
interference disappears (see Figure 21.)
Interference
equal in each
direction so it
cancels itself
Interference in
both halves
Figure 20 - The signals pick up interference... Figure 21 - ...and then it is cancelled out
Balanced connections are made using three pin XLR connections in the main body of the theatre,
and “B-gauge” (knobbly-ended) jack plugs in the patchbay. All cables from the theatre into the
sound box are balanced to minimise interference, but connections from the tape decks and CD
players are unbalanced as they are so short. (However, the Denon MiniDisc connections are
balanced.)
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What if you have a balanced output and unbalanced input, or vice versa?
Unbalanced inputs such as electric guitars can be balanced by plugging them into a direct injection
box. The transformer in this box will do all the splitting and inverting necessary, and output a
balanced signal. It will also standardise the signal level to something that the mixer can cope with
more easily. (Balanced lines typically run at lower levels than unbalanced lines.) The cheap and
nasty way is to wire the unbalanced signal to one of the balanced signal connections and to earth the
other one (see Figure 22.) But beware - you will lose all the advantages of balanced line operation
by doing this!
Signal Hot
Unbalanced Cold Balanced
Output Input
Ground Ground
Figure 22 - Connecting an unbalanced output to a balanced input
Likewise, balanced outputs can be wired to unbalanced inputs by wiring one of the cores to the
unbalanced signal connection, and the other to ground, and not connecting the screen at the output
end. If this is done (as shown in Figure 23), some but not all of the interference “rejection” will still
work - this is called quasi-balancing.
Hot Signal
Balanced Cold Unbalanced
Output Input
Ground Ground
(no connection)
Figure 23 - Connecting a balanced output to an unbalanced input
For high quality work, unbalanced inputs can be balanced using special adapters with integral
transformers which act like direct injection boxes in reverse. However, the ADC Theatre currently
does not own any of these.
A useful by-product of the use of balanced lines is phantom power. Capacitor (condenser/ electret)
microphones need a power supply for their built-in amplifier. This can be supplied by fitting
batteries; however, this is cumbersome and unreliable. The elegant solution is to feed the
microphone power from the mixing desk; several schemes exist, the most common being:
• phantom power - this raises the voltage level of two signal wires relative to the screen.
• T-power - this superimposes a voltage difference between the two signal cores. This is rarely
used, phantom powering being standard today.
To use this powering technique, the mixer and the microphone must use the same powering scheme
at the same voltage (12 and 48 volts are common). The ADC mixer can supply 48 volt phantom
power on all microphone inputs. There are individual switches beside the connections at the back of
each mono input channel.
Only use phantom power on the channels that need it, as other equipment (especially unbalanced
equipment) may occasionally react unpredictably. None of the microphones which the ADC owns
require phantom power; however, most shotgun and rifle mics do.
Page 54 of 107
Sound out and about

What has been described so far is fine within the confines of the theatre. However, on tour, and
particularly outside, there are other issues to consider.
Touring
“Prepare for the worst, but hope for the best.”
Assume nothing - you will find very few other theatres as well equipped as the ADC. Most usually
you will find a few mics (probably at least one SM58 or equivalent), lots of random cables, a
cassette deck, a large mixer, and some nice amps and speakers. This list is most notably missing a
CD player.
So for tours around the UK (and beyond), it is probably wise to take a CD or MD player with you,
and to record all the sound cues onto a CD before you go. Choose a CD player that is robust rather
than packed with features; an autocue facility is useful for fast-paced sound scripts, but see the
comments on cueing on page 19. Recording CDs is quite cheap these days (less than £1 each), so it
makes sense to master several in case you lose one.
Remember to take plenty of different adapters to convert between different types of sound
connector; it is highly likely that the house mixer will have a jack, phono or XLR input somewhere.
Putting all travelling equipment in flight-cases is a good way of ensuring that it survives the tour.
Label all your equipment, and especially the connecting leads, to save arguments with venues over
what you brought and what they own.
An occasional problem is that of a “noisy” mains supply; “spikes” and fluctuations in the supply
voltage can both affect sound quality and damage equipment; it is worth taking a plug-in surge
protector and mains filter unit to guard against this.
Sound in the Open

When rigging sound equipment outside, the first thing to consider is power - how to obtain it, and
how to use it safely. Power can come from a normal mains socket (be that in a nearby building or
external location), from a temporary “wire-in” or from a generator. Wire-ins should be done by
someone who is electrically competent, especially if they involve “working live,” and however
temporary they may be should conform to the same safety standards as a permanent job. Don’t
forget to check with whoever is supplying the power that the supply can handle the load that you
propose to draw! Wherever possible, generators should be put away from the audience to minimise
nuisance from noise and fumes.
Always label your power connection; if you are a long way from it, it may not be immediately
obvious what the plug going out of the window powers, and it may well be mistaken for something
which has been accidentally left on and be unplugged by an over-zealous cleaner...
Wherever the power comes from, make sure that the supply chain is protected by a Residual Current
Circuit Breaker (RCCB) if it is going outside. These devices monitor the outgoing current in the live
wire and the returning current in the neutral wire, and switch off the supply if there is more than a
Page 55 of 107
certain discrepancy between the two (usually 30 mA or 100 mA) as the discrepancy indicates that
the current has found another route to earth - which could be you! Permanently installed RCCBs are
normally provided for external sockets and often for internal sockets; even if you are assured by the
venue that one is in the circuit, take the time to find it yourself, as you’ll need to find it very quickly
if it trips in the middle of a show! Portable RCCBs are available as either in-line adapters (which are
more versatile) or built into plug tops.
Always bring more than enough cable, and ensure that once rigged, all cables are out of the way and
do not present a trip hazard. Remember that outside it is more difficult to exclude the general public
from areas, and it is not acceptable to assume that they will not be staggering around behind a
marquee simply because there is no reason for them to be doing so.
Cable rubbers do not work very well on grass surfaces; rubber matting is more successful, but the
best solution is to fly the cables from high objects such as trees and buildings, ensuring that they are
clearly visible (tie “flags” of brightly coloured material to them) and over head height - or lorry
height if necessary! When flying cables, remember that the cable must be capable of taking its own
weight; in many cases, this will require the use of either special “flying cable” (with a strength
member built in) or an external “cantenary wire”, from which the cable is supported across the span.
It is acceptable to secure the ends by tying the cable off against itself for temporary work, but cable
anchors (a stiff wire fixed at one end and with a helix through which the cable passes at the other)
should be used for permanent installations.
When deciding upon speaker placement, spread them out around the area to be covered; sound
diffuses more in the open, and if the speakers are only in a single place, the sound will be
unbearably loud there, yet still be inaudible a short distance away. More speakers over a wider area
but delivering less power will give a more even volume over the area, and will avoid creating
“sound pollution” - another effect of sound diffusing more in the open is that more consideration
has to be given to areas where sound is not intended to go!
Higher frequencies tend to get lost outside, so it is often necessary to apply some “treble boost” to
the sound output; this is important for clarity on public address applications.
If very long cable runs to distant speakers are considered, or if there is a large number of speakers, a
100 Volt line system may warrant consideration. This system cuts losses in cables in the same way
that power losses in the National Grid are minimised - it involves stepping up the output voltage at
source, and then stepping it back down again locally using step-up and step-down transformers; this
is shown in Figure 24. This reduces the current in the wires, and so reduces the loss due to the
resistance of the wires. A 100 Volt line amplifier usually has a pair of screw terminals on the back,
to which either speakers with built-in transformers or separate transformer units to drive external
speakers are connected, again usually with screw connections. This system is usually only used with
low powered speakers and not for high-quality reproduction, but is very useful for public address
and background music (e.g. the ADC telephone paging and show relay systems).
Low voltage, high current - hence

high losses due to resistance
Amplifier Speaker
Step-up Step-down
transformer transformer
High voltage, low current - hence
low losses due to resistance
Figure 24 - A 100 Volt line system
Page 56 of 107
Further Reading
If you want to learn more about sound, there are several books on the subject which merit attention:
• John Leonard’s book, Theatre Sound is a great book to read. Its major plus point is that it is
recent, published in 2001 by A & C Black, and therefore up-to-date; and is a very comprehensive
treatment of the subject. The ADC Library has a copy of this title, so have a read through some
time.
• Vivian Capel has written many short books on sound (the subjects tend to be biased towards
loudspeaker systems) which start by explaining the theory, continue by giving some examples
and finish by giving you details of how to build one. He also wrote the seminal The Audio and
Hi-Fi Engineer’s Pocket Book, which is unbelievably complete for a book of its size. If you
only ever buy one book on sound theory, this has to be the one!
• Robert Penfold churns out books on all aspects of electronics, including sound. These are more
theoretical, but still useful if you have the time to play with the ideas that these books describe.
Treat them as a source of inspiration, not a source of answers.
• Sound for the Theatre by Graham Wallace and published by A & C Black and Theatre Arts
Books is an excellent book which covers acoustics, audio basics, system design and
communications, and is ideal for a beginner who wants to become proficient very quickly. It is
rather out of date; although it was rewritten in 1990, much of the material has not been updated
from the original 1981 edition. However, it should be remembered that the ADC has a much
more modern sound system than most theatres boast, and the equipment and techniques
described in this book are still in widespread use. The illustrations are excellent.
• Creative Recording by Paul White is a well written three volume guide to studio sound
techniques, some of which are also relevant to theatre sound. Volume two covers microphone
selection and placement, and is useful reading for anyone involved in the sound design for a
musical. Volume one (Effects and Processors) and volume three (Acoustics, Soundproofing and
Monitoring) are less relevant but still interesting.
• Sound equipment catalogues are a valuable source of information; before they can sell their
wonderful piece of equipment to someone, they have to explain what it does, how it solves a
problem which you never even knew existed, and in some cases, the theory which causes the
problem in the first place! They are also very valuable for looking up the specification of a piece
of equipment for which you only know the model number. The Maplin Catalogue is very down-
market, often wrong, and not to be recommended for educational purposes. The Studiospares
Catalogue is quite complete and features excellent explanations - you can get a copy by phoning
them. The Canford Catalogue (“The Source”) is the “who’s who” of every conceivable sort of
sound equipment. Canford have the contract to supply the BBC and many Independent Local
Radio stations (ILRs) with equipment - these people know a lot about sound! They’ll put you on
their mailing list if you phone them on 0191-415-0205.
• Finally, for upwardly mobile sound demigods, the BBC Specification for the Wiring of
Equipment (ED122) is worth a look. This book describes wiring installation techniques and
equipment construction techniques, but its real value is the ideas that it throws up on how to do
everything perfectly.
Page 57 of 107
The Internet
The growth of the Internet has provided several new information sources for sound engineers. Due
to the unregulated nature of this medium, there is no guarantee that what you read will be accurate;
however, most of the information is useful and thought-provoking.
There is a large amount of information about sound available on the World Wide Web. The
following URLs (Uniform Resource Locators) provide a useful starting point:
• http://www.midifarm.com/ MIDIfarm (MIDI files and software are available here)
• http://www.lcsaudio.com/ Level Control Systems
• http://www.spectralinc.com/ Spectral Inc.
• http://www.shure.com/ Shure Brothers Inc.
• http://www.prs.co.uk/ Performing Right Society
• http://www.theatre-sound.com/ Richmond Sound Design
• http://www.wrn.org/ibs/ The Institute of Broadcast Sound
• http://www.comlab.ox.ac.uk/archive/sound.html Oxford University Sound Archive
Several of these sites offer downloadable sound effects. These are files which can be turned back
into sound using a computer fitted with a sound card. The “Cool Edit Pro.” package for the PC is
highly recommended for this; it can play the vast majority of sound file formats, including those for
other makes of computer. However, some of these effects are of rather poor quality, and a
reasonably good sound card will be needed to turn any file into a realistic sound effect.
USENET news also provides several interesting discussion forums for sound engineers, where
questions can be asked and unusual problems brainstormed. Of particular value is
rec.arts.theatre.stagecraft, which is aimed at all types of theatre technicians, but the following are
also interesting:
• alt.binaries.sound..... Downloadable sound effects
• rec.audio.pro Discussion of professional audio equipment
• rec.audio.tech Discussion of technical audio subjects
• uk.tech.broadcast Discussion of broadcasting in the UK
Page 58 of 107
Appendix A
Sound in Other Cambridge Venues
This list of descriptions is not intended to be exhaustive, but represents venues for which
information was available, and the situations at the time of research.
If you are involved with a venue not listed here, or you find that something has changed or is
incorrect, then please send the authors details of what you find at your disposal - contact details are
given on page 6.
Page 59 of 107
Arts Theatre
Mics: 4 Beyer M201s, 5 Beyer M88s and no.
of Shure SM58s
10 Sennheiser UHF radio mics
3 Crown PCC 160s
3 Audio-Technica AT815 rifle mics
Mic stands: Plenty
Sources: 2 Sony 1U MiniDisc recorders

1 Tascam MiniDisc recorder
1 Denon DA30 DAT machine
1 Tascam CD401 CD player varispeed
1 Tascam 122 Mk.2 cassette player remote control
Mixer: Soundcraft k3 24-8-8-6 24 mono, 8 stereo inputs, 8 sub-groups, 8

way matrix
Yamaha 03D
Effects: Yamaha SPX1000 reverb / effects unit

2 x Yamaha Q2031 30 band stereo EQ
unit
2 TOA programmable speaker/ dynamic Used for the stalls and circle delay
processors speakers
Amplifiers: 2 C-audio RA3001

2 C-audio RA1001
3 C-audio ST600
2 d&b P1200L amps
Speakers: 6 d&b 902 speakers (4 main, 2 sub- flown above and beside stage
bass)
2 EV-SX200
2 EV-SX40
2 JBL Control 1 stalls delay speakers
2 JBL Control 1 circle delay speakers
2 JBL Control 5
2 JBL Control 1
Tie-lines: XLR tie-lines from everywhere to everywhere else! There are also plenty of speaker
lines to the auditorium intended for rigging cinema surround sound. No normalling
exists within the main patchbay, so lots of patchleads are needed.
The Arts Theatre also boasts a sound and light sculpture costing over £50,000 as the St Edward’s
Passage stairs - each step makes a different noise and lights a different LED. Wow.
Page 60 of 107
Christ’s New Court Theatre

Mics: None
Mic stands: None
Sources: Tape Deck

CD player
Mixer: Allen & Heath Icon DP1000P - 8 channel digital mixer with 300W/channel
amplifier combined unit.
Speakers: 2 x 350W speakers
Tie-lines: 6 XLR tie lines from points around building to tails by the sound position.
The box is sound-proof!
Corn Exchange
Lots of very nice equipment, but used for few student productions.
Downing College Theatre
Mics: Built into lectern and no. of tabletop goosenecks

Radio mic.
Sources: CD, double cassette deck.
Mixer: Soundcraft Spirit F1-14/2. 6 mic/line inputs
Amplifiers: Yamaha PM1300
Speakers: Four JBL Control 28s rigged FOH, 2 per side.
Tie-lines: Some XLR tie lines from points around building to tails by the sound position
Nice recent install. Pity that standard mixing position is in the stage right wing!
Page 61 of 107
Emmanuel URC
Mics: Most built into furnishings (e.g. pulpit, lectern)
Radio mic.
Mic stands: Several
Sources: 1 tape deck
Mixer: Six channel mixer / amp combined unit.
Amplifiers: Additional amplifier for induction loop
Speakers: Column speakers on a single 100v line circuit
Tie-lines: 8 XLR tie lines from points around building to tails by the sound position
terminated in 4 way DIN plugs.
The quality of the installation work is very good for a church building, and the only caveat is that
the dimmer unit for the Parcans on the sidebars tends to generate a lot of interference.
Fitzpatrick Theatre, Queens’

Mics: 6x AKG D190
SM58 mic - cabled
2x SM58 mic - radio
2x lavalier mic - radio
Mic stands: Several available
Sources: Minidisc deck

DVD Player
CD Player
Laserdisc Player
Double Tape Deck
Reel-to-Reel
Mixer: Allen & Heath GL3000 24 channel
Speakers: Two hanging JBL, though Queens’ Ents speakers may be available
Other: Graphic equaliser and compressor
Page 62 of 107
Holy Trinity Church

Mics: Several available.
Two radio mics with installed aerials at the sound position
Mic stands: Several available
Sources: 1 tape deck
Mixer: Stagecraft 16?
Amplifiers: Four zone 100v line amp

Additional amplifier for induction loop inside vestry; cable to sound position fitted
with jack plug.
Speakers: 100v line speakers on four zones. Note the switch underneath the speakers covering
the altar area.
Tie-lines: Stagebox with plenty of slack cable under pulpit with tails at sound position fitted
with XLR connectors.
The equipment is of good quality, and whilst the speakers are in good positions for public address,
they tend to also be in the right positions to initiate feedback. The slightly illogical speaker zoning
doesn’t help matters.
The McCrum Theatre, Corpus

Mics: 2 x Shure SM58
Mixer: 12-2 channel mixer.
Tie-lines: XLR tie lines from stage to wings. Jack and video tie lines from stage to control
position.
The college does not like students using this venue - it was built for the conference trade. It has tie
lines for both mics and speakers, two speakers built into the walls permanently connected to the
Speakon patch and two SM58 mics, but no cable or amplifier. An induction loop is also installed.
Good luck.
Page 63 of 107
Mumford Theatre, APU

Mics: A small selection of average units.
Mic stands: 2
Sources: 2 tape decks
Mixer: Allen and Heath GL3 16-4-2 6 auxiliary mixes.
Amplifiers: C-Audio RA2001 (450W / channel) Fed to the speakers flown above stage.
Hill Audio (~200W / channel) For the Bose speakers at the side of stage
Speakers: 2 x CSX 5252 permanently flown above stage

2 x Bose 802 can be rigged as required.
Tie-lines: 8 XLR tie lines from each side of stage to A type jack patch in LX box.
24 tie lines from LX box to rear of auditorium PS.
All the tie lines are normalled at a jack patch under mixer in the LX box. 1-16 lead to the stage, and
17-24 are bi-directional lines between the LX box and RoH PS. 8 mains sockets on the sound power
circuit are provided by the XLR box in the auditorium. The sound power breaker is in the LX box.
There is a single cans ring, which is fed to the front of house sound position through a tie-line.
Plenty of leads are available, many made from Starquad cable.
The position of the speakers above the stage seems to cause many problems with feedback. This,
coupled with the shape of the theatre, make an equaliser highly useful when working here.
Most of the equipment provided uses unbalanced connections.
The theatre is also used as lecture theatre by Anglia Polytechnic University, so any equipment
rigged in the auditorium must come out during the day. The Mumford Theatre has some resident
crew, who are extremely knowledgeable if slightly poor communicators (although they are getting
better...)
Page 64 of 107
Peterhouse Theatre
Mics: 2 x AKG D190E
1 x AKG D900E rifle mic used for show relay to projection room
Mic stands: 1 x Mic stand

1 x Boom arm
2 x Table gooseneck stands
Sources: Tascam 133 cassette deck remote start connector from lectern.
Cue track facility.
Technics M229X cassette deck
Mixer: Studiomaster 8 into 4
Amplifiers: 2 x Amps (4 channels) 2 channels fed to front of house

2 channels for monitors in projection
room
Speakers: 2 x Fostex SP11 fitted to auditorium side walls

2 x AKG LSM50 fitted in projection room.
Tie-lines: 4 XLR lines from stage to projection room.

1 XLR line from relay mic on front of balcony to projection room.
This decent equipment is spoiled by poor installation. There is no patchbay, and the wiring is rather
Heath-Robinson. Having said this, the sound is very acceptable, and the only major gripe is the lack
of a CD player.
For high-quality sound, Peterhouse Films own an additional two speakers which can be clamped to
a lighting barrel (these are useful for covering the balcony and for “in-place” sound effects), an
amplifier to drive them, and a bass driver with built-in amplifier which gives the system more
wellie. These can be hired separately.
The projection room monitor speakers appear to have been chosen for size rather than sound; AKG
designed the LDM50 to sound like a cheap TV loudspeaker so that they could be used in the studio
to monitor “what the average consumer would hear” rather than a true reproduction of the sound!
The College has a PPL licence, and charges all events which might require it the princely sum of
£1.50. However, there is no PRS licence, so arrangements will have to be made directly by the hirer.
Page 65 of 107
Playroom
The system currently consists of a pair of wall mounted Mission speakers (somewhat knackered) in
each of the two sections of the auditorium, which are cabled back to the control position. A cassette
deck exists as does a CD player, but no mixer! An induction loop has recently been installed.
Apparently, if a college member is involved in the show, the Corpus Ents sound system can also be
used; this includes a 1.2kW amplifier, an 8-2 mixing desk and two huge speakers!
Queens Building, Emmanual

This relatively new venue is fitted with a Peavey 12-2 mixer with built-in effects, and a pair of
200W Toa speakers. A radio mic is provided, and show relay is via a “shy” Shure SM57.
Robinson Auditorium
Sources: Cassette deck

CD player
Sony minidisc recorder
Mixer: Soundcraft 24 channel mixer
Mics: A few, and some radiomics
Speakers: 4 x Bose 802 fitted to auditorium side walls
Tie-lines: XLR lines from stage to projection room.
Yet another conference venue. Has a soundproof box and a poor show relay system to the control
box. A horrific cabling tangle exists due to lack of a patchbay.
St. Chad’s Octagon, St Catherine’s

No sound equipment installed. The domed structure causes severe acoustic problems, and amplified
sound seems to find feedback paths very easily!
Page 66 of 107
School of Pythagoras, St. John’s

Mics: none operational
Sources:
Denon DRM-510 Cassette Deck
Mixer: Yamaha MV422 4 mic channels

2 stereo sources
Amplifiers: ???
Speakers: 2 x Bose 402 mounted on side walls
Tie-lines: 8 XLR lines from stage to sound / lighting box (4 male on stage to female in the box
and 4 in the opposite direction.)
At first, this venue looks reasonably well equipped for its size; however, time must be spent finding
out what does not work. Silly oversights abound; for example, the cable store consists of very few
lengths of absurdly long cable.
Beware of the cassette deck - it lacks a useable pause function!
Even with all the windows removed from the lighting and sound box, the action on stage is barely
audible. To assist in running the show, it is useful to place a stage relay microphone above the stage,
and to feed this to some monitor speakers in the box; setting levels for this is tricky, so use the tech
and dress rehearsals!
Page 67 of 107
Appendix B
Sound Connectors
There are many different types of sound connector in use. Each has its own advantages and
disadvantages, and represents a different trade-off between quality and price; therefore different
connectors are appropriate in different situations. Some of the most important criteria are:
• latching - if someone knocks the connection or accidentally pulls on the cable, will the plug fall
out of the socket? Many professional connectors have a locking ring or clip that must be rotated
or depressed to free the plug.
• price - obviously, this is a major consideration, or everyone would buy the very best connectors.
However, remember that you will regret saving a few pounds on a connector when it lets down
your entire sound system in the middle of a show.
• cable retention and strain relief - this is how well the body of the cable is gripped inside the
connector. Cheap connectors will rely on the electrical connections to hold the wire in place,
whereas more expensive connectors will provide some form of clamp to grip the insulation.
Connectors may also include a plastic or metal spring which surrounds the outgoing cable for a
short distance. This prevents the cable from being bent too sharply, and so prevents “wirebreaks”
where the conductors snaps inside the cable due to mechanical stress. These are very frustrating
faults to find, as they are often intermittent and always occur when time is of the essence!
• contact resistance - this covers both the contact area and the contact material. Gold is often used
as it has a low resistivity, but the subject of what makes good contact materials and complex
overplating schemes involves an understanding of metallurgy. Good quality connectors are
generally made out of the right material, whereas cheaper connectors have “metal” pins.
• number and arrangement of pins - the need for the right number, or at least a minimum
number of pins is obvious; the arrangement is slightly more subtle. Most connectors are
“polarised,” which means that they can only be inserted one way around, thus preventing
misconnection. Some connectors guarantee that the earth pin is the first to make and the last to
break; this helps to prevent the “thumps” associated with connecting equipment to a powered-up
amplifier. Other connectors have sockets which can operate “break contacts” which are made
when the socket is empty, and are used for normalling.
• electrical rating - the chosen connector must be able to withstand the voltage and current to
which it will be subjected. Bear this in mind with speaker connectors; speakers can draw currents
in the order of a few amps, and the smaller sound connectors cannot handle this!
• frequency rating and impedance - at what frequencies will the connector be operating? At
radio frequencies (e.g. on a radio microphone aerial or a video signal), the connector’s
impedance must match the cable being used to prevent “reflections” down the cable and
consequent signal degradation.
• convention - if everyone else uses a particular connector for a particular job, there’s probably
(but not always!) a good reason for it. Bear in mind that if you use non-standard connectors, you
will need adapters to connect into other systems, which could involve extra cost and will
introduce additional points where the system could fail.
• ease of wiring - connectors come with solder connectors, screw connectors or push-fit
arrangements. Solder connections are best for a really good electrical and mechanical connection,
but screw and push connectors are useful in the field where time and tools are limited.
Page 68 of 107
• quality - this covers the amount of thought which has gone into the design, the quality of
manufacture and dimensional accuracy, and the quality of materials used. A properly made
connector is very much easier to fit! Also, bear in mind that cheaper connectors are only designed
for occasional connection and disconnection before they wear our, whereas quality connectors
are more suitable for regular use.
The authors use and recommend Neutrik brand connectors; whilst more expensive than some
brands, they are very well designed, a joy to assemble, and last virtually forever whilst giving very
little trouble.
A summary of the more popular types of audio connector is given below:
BNC
These are not normally used as audio connectors, as they are
designed for use at higher frequencies; however, they may be
encountered for video connections or for radio mic aerials.
The plug is held to the socket using a collar with a bayonet
fixing, which also forms the screen connector. The signal
Figure 25 - A BNC connector
connector is a pin recessed into the centre of the connector;
note that this can sometimes become bent with heavy
handling.
DIN
This connector is designed to a German standard (Deutsche Industrie Normal.) It is available in
many versions with varying numbers of pins and pin spacings. The types commonly encountered in
audio applications are shown in Figure 26.
2 pin DIN connectors are used for speaker outputs on cheaper domestic audio equipment. The round
pin is positive and the flat pin is negative.
The 5 pin (180° spacing) version of these connectors used to be the standard for domestic audio
equipment. Like the current “SCART” video connector, the idea was to provide a single plug which
carried all the connections, keeping things simple for the non-technical domestic user.
Unfortunately, this system was inflexible so the standards weren’t followed by all manufacturers,
leading to several different ways of wiring the same connector for the same function. The
connectors themselves leave much to be desired; cable retention is poor, most have no locking
mechanism, and they are very fiddly to connect - the plastic body tends to melt before the pin is up
to soldering temperature!
Page 69 of 107
1 3 1 3 1 3 1
4 2 5 4
2 2 2
2 pin DIN 3 pin DIN 4 pin DIN 5 pin DIN (180°)

2 5 7 6
5 1 5 1
3 1
4 1 4 6
2 3 2
4 5 4
3 3 2
5 pin DIN (240°) 5 pin DIN (360°) 6 pin DIN 7 pin DIN
(shown looking into socket)
Figure 26 - Pin layouts of common DIN connectors
Although they are most commonly used for line level signals, microphones used to be fitted with
DIN plugs. Mono connections were sometimes made with 3 pin connectors, which plugged into the
correct holes on 5 pin sockets - this is why the numbering of the 5 pin plug is out of sequence.
The correct way of wiring these connectors is as follows:
Pin 1 2 3 4 5
Stereo Line Left Input Screen Left Output Right Input Right Input
Stereo Line Left Output Screen Left Input Right Output Right Output
Mono Line Output Screen Input
Headphones Left Muff Screen Right Muff N/C N/C
(180° 5 pin)
Headphones Right Muff Left Muff N/C Screen N/C
(240° 5 pin)
Headphones Screen Left Muff Right Muff Left Muff Right Muff
(Sennheiser Return Return Send Send
360° 5 pin)
“Universal Mic level left Screen Line Level Mic Level Line Level
Input” Left Right Right
Balanced Hot Signal Screen Cold Signal
Mic Input
Unbalanced Signal Screen N/C
Mic Input
Unbalanced Input (High Screen Input (Low
Mic Input Impedance) Impedance)
MIDI N/C Screen N/C +5 Volts MIDI data
If these connectors are encountered, the best approach is often to use a ready-made lead which
breaks the five pins out into four separate connectors; this save guesswork and repeated soldering to
find out which pin is which!
Page 70 of 107
Jack
There are a whole family of jack connectors, used for a huge variety of applications. The main
differences are the size, the gauge (shape) and the number of poles (connections).
Size is measured by the width of the tip, and is usually either 3.5 mm on 6.25 mm (often called
quarter-inch). 3.5 mm plugs are usually only used on cheaper domestic equipment and miniature
equipment (e.g. Walkmans), as the retention is poor and the contact area is small.
The two gauges are “A” type, which is common on domestic equipment, and most commonly seen
as a “headphone connector,” and “B” type, which is more knobbly, and used in patchbays. Note
that it is possible to damage a socket of one gauge by inserting a plug of another gauge
into it - even if it does work! Only quarter-inch, three pole plugs are commonly available in “B”
gauge.
Two and three pole plugs are most common, although four pole “B” gauge plugs used to be used for
telecommunications. Two pole plugs can only be used for mono unbalanced signals, but three pole
plugs can either carry mono balanced signals or stereo unbalanced signals. Three pole plugs are also
used for “insert” connections, where an unbalanced signal is sent on the tip contact and returned on
the ring contact.
Two common jack plugs are shown in Figure 27 and Figure 28.
Figure 27 - “A” gauge three pole Figure 28 - “A” gauge two pole
quarter-inch jack quarter-inch jack
The connections for typical applications are shown below:
Application Tip Ring Sleeve

Mono Headphone Signal Earth
Stereo Headphones Left Signal Right Signal Earth
Unbalanced Mic Signal Earth
Balanced Mic Hot Signal Cold Signal Earth
Balanced Patchlead Hot Signal Cold Signal Earth
Insert Lead Send Signal Return Signal Earth
Page 71 of 107
Lemo
These tiny, slender Swiss plugs are most often encountered connecting microphones leads to radio
mic transmitter packs. They are quite difficult to assemble. In theatres they are most often seen in 4
and 5 pin versions, but Lemo make a huge number of connectors.
Phono
Phono connectors are used extensively in domestic hi-fi. The
name stems from their original use to connect record decks
(which used to be called “phonographs.”)
Figure 29 - A Phono connector They are two pole connectors, used exclusively for unbalanced
systems, and often in stereo pairs. Normally, they are used for
line level signals, but some older systems use them for low power speaker connectors as well. In the
ADC, they are used for connections to the back of the cassette decks and CD players, and on the
smaller mixer.
Whilst cheap and commonplace, they suffer from several problems. Cable strain relief is poor, and
the plug and socket don’t lock together, so if the cable is pulled, either the plug or the cable will
come out. More importantly, the signal connector makes contact before the earth connector - this
can lead to loud buzzing as the plug is pushed in; always power down amplifiers when phono plugs
are being connected to avoid damage to the speakers. Canford and Neutrik sell “re-engineered”
phono plugs which have been modified to mitigate these problems, but they are quite expensive.
The connector fitted to the equipment is always female, so most leads are fitted with male
connectors on both ends. In-line female connectors are occasionally used.
SCART
20 18 16 14 12 10 8 6 4 2
Also called “Peritel” connectors and
“Euroconnectors,” these are multipole
21 connectors which allow for stereo unbalanced
19 17 15 13 11 9 7 5 3 1 sound and a variety of video formats
(shown looking into socket) (including control signals) to pass in both
directions. Whilst useful for connecting two
Figure 30 - Pin layout of a SCART connector
or three devices together, they become
unwieldy for larger setups and separate connections become more practicable. However, the vast
majority of new video recorders, televisions, satellite or cable receivers and other video equipment
have these connectors, and a lead which breaks one of these connectors into other connectors
(usually phono and BNC) carrying separate signals is most useful. Argos sell a very useful kit for
about £10 which splits a SCART plug into several connectors and has a couple of adapters thrown
is as well, but it is worth taking the time to rebuild the SCART to DIN lead with a higher quality
audio-visual cable! Note that cheaper versions of this connector do not take kindly to repeated
connection and disconnection.
Page 72 of 107
Cable retention is poor, especially if only a handful of pins are connected and therefore a slimmer
cable is being used. Connecting all the pins is a fiddly job, but easier than wiring a DIN connector
as the pins are spread out over a wider area. The pin-out is as follows:
Pin Function Pin Function

1 Right audio out 12 Future use / D2B data
2 Right audio in 13 Red / Chrominance ground
3 Left audio out 14 Fast blanking ground
4 Audio ground 15 Red / Chrominance
5 Blue ground 16 Fast blanking
6 Left audio in 17 CVBS/ Luminance out ground
7 Blue signal 18 CVBS/ Luminance in ground
8 Status (16:9) 19 CVBS/ Luminance out
9 Green ground 20 CVBS/ Luminance in
10 Future use / D2B data 21 Screen
11 Green
Speakon
These are fast becoming
the industry standard
speaker connectors; they
feature good cable
retention a positive
locking action, and a
high current carrying
capability (20/50A).
They come in 2, 4 and 8
Figure 31 - A Speakon connector pin configurations, for
bi- and tri-amped
speakers. They are preferred to XLR for speakers as they can carry higher currents, and being
physically different, the possibility that an amplifier output will be connected to a microphone input
and cause damage has been removed. Note also that as the output voltage from amplifiers can
exceed 50 volts, legislation requires that it should be fed out on “touch-proof” connections. Speakon
connectors fulfil this requirement, whereas XLR connectors don’t. Wiring convention is as below:
Devices to be Type 1- 1+ 2- 2+ 3- 3+ 4- 4+
connected
Full range NL2 Return Signal - - - - - -
system
Full range NL4 Return Signal - - - - - -
system
Bi-amped NL4 LF LF HF return HF - - - -
system return signal signal
Bi-amped NL8 LF LF - - - - HF HF
system return signal return signal
Tri-amped NL8 LF LF - - Mid Mid HF HF
system return signal return signal return signal
Quad-amped NL8 LF LF Low mid Low mid High mid High mid HF HF
system return signal return signal return signal return signal
Page 73 of 107
XLR
2 1
(shown looking into socket)

Figure 32 - An XLR connector, also showing pin layout
The XLR connector is probably the most widely used professional sound connector, and is used for
carrying signals at any level from microphone level to speaker level and even low voltage power
supplies. It features excellent cable retention, a positive locking action, is available with many pin
configurations, and boasts large pins which provide low resistance for low level signals, and high
current carrying capacity for speaker level signals. They are virtually indestructible, easy to wire and
they even look pretty. Finally, as there are a variety of pin configurations, connectors for different
uses can be made physically incompatible with each other, leading to a limited degree of idiot-
proofness!
Some of the more common uses of XLR connectors in the theatre environment are as follows:
Pins Uses
3 Mic, line and speaker level sound interconnections.
Ring intercom single circuit ring connection.
D54 lighting control.
4 Ring intercom single muff headsets.
Pyrotechnic control occasionally.
Scroller control.
5 DMX512 lighting control.
Ring intercom dual muff headsets.
6 Ring intercom multiple circuit ring connections.
They are also occasionally used as low voltage power connectors (e.g. to “Birdies”, the miniature,
low voltage version of the Parcan), due to their high current rating.
The correct way of wiring three pin XLR connectors to balanced and unbalanced audio lines is as
follows:
Pin number
1 2 3
Balanced Earth Hot Signal Cold Signal
Unbalanced Earth Signal connect to earth at
unbalanced end.
The shell of the plug forms a separate “chassis” earth connection (tied to mains earth), which should
not be connected to the pin 1 “screen” earth connection (used in audio circuits.) This would give
rise to mains interference and the possibility that if the connector is touching a metal surface (e.g. a
lighting bar or the counterweight gallery) the resulting “extra” earth connection may form a hum
loop.
Page 74 of 107
Sound sources are always fitted with a male connector, and inputs are fitted with a female
connector, so most leads are fitted with a male connector at one end and a female connector at the
other end. As many different signal levels are carried by XLR cables, make sure that you know what
type of wire has been used to make up the cable; microphone cable does not last long when carrying
the signals to a speaker or a birdie!
When used with the ring intercom system, the pins are connected as follows (greyed out boxes
represent pins which don’t exist!):
Pin Number
1 2 3 4 5 6
Single Muff Mic earth / Mic Signal Earphone Earphone
Headset screen Earth / Signal
screen
Dual Muff Mic earth / Mic Signal Earphone Left Right
Headset screen Earth / Earphone Earphone
screen Signal Signal
Single Earth +24V DC Audio Bus
Circuit
Ring
Connection
Multiple Earth +24V DC Audio Audio Audio Audio
Circuit Circuit 1 Circuit 2 Circuit 3 Circuit 4
Ring
Connection
Page 75 of 107
Appendix C
Sound Cables
There are many different types of cable used in conjunction with sound equipment. Each has its own
advantages and disadvantages, and represents a different trade-off between quality and price;
therefore different cables are appropriate in different situations. Some of the most important criteria
are:
• current rating - can the cable cope with the maximum current which it will be expected to
handle? Putting more current through a given area of conductor makes it generate more heat, so
making the cross-sectional area of the conductor bigger makes it generate less heat. If the
insulation becomes too warm it might melt and fail or even cause a fire. Therefore, each cable is
given a maximum current rating depending on the cross-sectional area of conductor and the
resistance of that conductor. In audio applications, this factor is normally only a problem with
mains and speaker cables.
• screening - this is the metal layer under the insulation which prevents external electrical noise
from interfering with the signal inside the cable, and which also provides a current return path for
unbalanced connections. There are several different forms of screening:
• lapped (or helical) screens, where the screen wires are helically (i.e. spirally) twisted around
the signal conductor. These are easy to wire and provide acceptable screening.
• braided screens, where two helical screens (in opposite directions) are braided around the
cable. These provide fine screening, but are time consuming to wire as the screen has to be
un-picked before the central conductor is accessible.
• foil screens, where the screen is produced by a wrapping of aluminium foil. This provides
magnificent screening, but can easily be damaged by flexing, so these cables are most often
used in fixed installations. A “drain wire” is often provided; this is a bare metal wire in
contact with the foil along the length is the cable. When terminating the cable, the foil is
simply cut off and the drain wire terminated in its place.
• conductive plastic screens, where the metal is replaced by a inner conductive coating on the
plastic insulation. As with foil screened cables, a drain wire is usually provided giving fast
termination. The screening is only marginally acceptable with mic level signals.
• number and configuration of cores - can the cable carry enough signals for the task in hand?
Are they arranged so that they will not interfere with each other - this could involve twisting
pairs together or individually screening pairs.
• insulation breakdown voltage - this decides the maximum voltage which the cable can carry;
above a certain voltage, the insulation will start to conduct more electricity than would be
desirable, finally resulting in insulation breakdown. In audio applications, this factor is only
usually a problem for mains cables and 100 volt line speaker system cables.
• insulation material properties - the insulation material has to be suited to the environment in
which the cable is to be used. It must be robust enough to survive the treatment which it will
receive - will it be installed in conduit and then left undisturbed for 20 years or will it be dragged
across the stage and trodden upon every week (rubberised sheath; thickness)? It must be suitable
for the temperature range in which it will be used - will it be run so close to a hot lantern that it
melts, or will it be used outdoors and crack in cold weather? Finally, bear in mind that cable
installed indoors can be of the low smoke zero halogen (LS0H or LSZH) variety so that in the
event of a fire, the fumes given off by the insulation are less toxic.
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• colour - is the cable insulation the correct colour for identification or decorative purposes? Note
that some types of cable (e.g. Canford HST) are available in a wide variety of colours, including
UV-fluorescent and luminous versions!
• strength member - if a cable is to be flown under its own strength (e.g. from building to
building) it should be capable of taking its own weight. Some cables therefore incorporate a steel
or kevlar “strength member” so that this is achieved.
• flexibility - is the cable flexible enough for its intended use - cables installed in trunking will
almost never be flexed after installation, whereas cable on items such as microphone booms and
headphones will be flexing almost continuously. Where a cable will be flexing constantly, tinsel
cable is often used; this uses thin metal foils wrapped around a thread for conductors, which are
very flexible but have to be terminated by crimping as they are awkward to solder and too fragile
for screw connections. An example is in patchcords.
A summary of the more popular types of audio cable are given below:
Helical Screened Twin (HST)

This is a general purpose flexible cable often used for microphone cables. As the name suggests, it
consists of a twisted pair which has a screen helically wound around it. Many versions are available
for special applications.
Musiflex
This cable consists of a twisted pair and drain wire inside a conductive plastic screen; the absence of
a metal screen makes it much easier and faster to terminate than a conventional balanced screened
cable. However, the plastic screen is inferior to its metal counterpart, so long runs of these leads are
best used with line level signals where interference is less likely to be noticeable.
Musiflex is used in the ADC theatre for the green D54 leads, blue ring intercom leads and purple
XLR patch leads. Note that both of these applications use signals at a relatively high level. A single
core version of this cable called “Phonoflex” also exists.
Oxygen Free Copper

This is a form of high purity copper, used in the manufacture of audio cables. When it was first
introduced, the “loony golden ears” brigade started making all sorts of claims about how wonderful
cables made with OFC were; in reality, they are slightly better (and easier to solder, if nothing else!)
and now the hype has died down and prices have fallen, they are worth buying if the budget will
stretch to it.
Starquad
This is an unusual microphone cable as it consists of four cores twisted together with a very short
“lay length” or length between twists. The cable is connected to a balanced line with the red and
white cores carrying the “hot” signal and the blue and green cores carrying the “cold” signal. The
reason for the use of two cores (which are on opposite sides of the cable) for each connection and
Page 77 of 107
for the very short twist is to ensure that the two sides of the balanced line really do experience
exactly the same interference, and so to improve the noise cancelling effect. It works phenomenally
well! The ADC Theatre owns a limited amount of starquad cable, kept on a separate spike in the
sound shelf cupboard.
Many varieties exist, including a tinsel variety (e.g. for boom microphones), a flying variety (i.e.
with a strength member) and varieties with different jackets for different environments.
Multicore Cables
Beware of crosstalk or co-channel interference when using multicore cables; this occurs when the
signal from one core starts to make its presence felt on adjacent cores. This can be minimised by
using individually screened cores (or pairs, if the cable is carrying balanced connections) rather than
many cores with one overall screen. Keeping all the cores in a multicore cable running at similar
levels (e.g. not mixing mic and line level signals) also helps to prevent cross-talk..
Speaker Cables
Ensure that speaker cables can handle the current which they are carrying. One complication with
this is “skin effect” where higher frequencies only travel in the outer “skin” of each strand of cable,
and the higher the frequency, the thinner the skin; to reduce the effect of this, most speaker cables
are composed of a large number (79 and 120 are common) of very thin strands, so that the “skin” of
the strand covers the bulk of the cable’s cross-sectional area.
Speaker cables are available either packaged side by side in a “figure of eight” arrangement, which
is convenient if the cores are to be terminated to separate 4mm plugs, or in a concentric arrangement
(like a screened cable) it the cable is to be terminated to a Speakon, XLR or jack plug. The ADC
Theatre is wired in 2.5mm2 conduit cable, with speaker leads in 2.5mm2 high power speaker cable.
Due to their high signal levels, speaker cables do not require a screen.
Mains Cable/flex
Ensure that the cable used is suitable; the current carrying capacity should be sufficient, and the
insulation should be able to withstand the environment in which it is being used. Inspect mains
cables frequently, and replace if any damage is evident.
Page 78 of 107
Appendix D
MiniDisc Instructions
Instructions for the Denon DN-990R MiniDisc player:
Playback
Selecting the play mode:
• Set play mode switch to SINGLE to play one track, cue the next and then pause.
• Set play mode switch to CONT for continuous play from one track to the next.
Selecting the track:

• Rotate the SELECT knob to increase or decrease the current track number.
• Depress the SELECT knob whilst rotating to increase or decrease track number in multiples of 10.
• When a track has been selected, the track number is displayed the track is cued. STDBY/CUE
flashes while cueing is in progress and lights steadily when cueing is complete.
Starting playback:
• Press the PLAY/PAUSE button whilst in the play, pause or standby mode to begin playback
immediately.
Stopping playback:
• Press the PLAY/PAUSE button during playback to enter pause mode. PLAY/PAUSE will flash
yellow.
• Alternatively, press the STDBY/CUE button during playback. The player will cue back to the
point at which playback started, or if a track has subsequently been selected with the SELECT
knob, the player will enter standby mode at that track.
Search operations (fast forward / rewind):

• Press one of the SEARCH buttons in the play, pause or standby mode to enter the search mode.
• Each press of a SEARCH button steps through the disc in frames of 1/75 second. If the button is
held depressed, the frames will step continuously, and with gradually increasing speed.
• When the search operation is completed, press STDBY/CUE to enter standby mode at that point.
• Press PLAY/PAUSE to begin playback.
End monitor function:

• Press the END MON button whilst in standby mode to play the last few seconds of the track to
check how it ends.
Pre-setting the next track to be played during playback:

• Rotate the SELECT knob to select the next track to be played. The track number will flash.
• In SINGLE mode, the beginning of the next track is set in standby mode when playback of the
current track finishes.
• In CONT mode, the next track is played when playback of the current track finishes.
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Switching the display mode:
• Press DISP/CAPS/NUM button to cycle through Elapsed Time, Track Title and Remaining Time
Displays
• Hold DISP/CAPS/NUM button depressed to show disc title.
Recording
Note that the recording time for a track must be at least 2 seconds.
Starting recording:
• Press the REC button whilst in the standby mode. The player allocates the new track the first
unused track number, finds space on the disc for the recording, and enters recording pause mode.
REC flashes in red whilst cueing takes place, and lights steadily when cueing is complete.
• Press the PLAY/PAUSE button while in the recording pause mode to start recording. (REC and
PLAY/PAUSE will both light.)
Stopping recording:
• Press PLAY/PAUSE button to enter recording pause mode. The track number will increment and
recording resume when the PLAY/PAUSE button is pressed again.
• Press STDBY/CUE to stop recording, write “table of contents information” (e.g. where on the disc
the recording can be found, how long it is, track title) to the disc and return to standby mode. The
player will then cue to the point where recording started.
Incrementing track numbers:

• Press the REC button during recording to increment the track number.
• The track number will automatically increment if a digital link to a CD or DAT player is used
and the external player sends details of a new track. (See full manual for more details.)
Editing
Erasing tracks:
This function erases an entire track and decreases all following track numbers to fill the gap. For
example, in track 5 is erased, track 6 will become track 5 and so on.
• Set the player to standby mode on the track to be erased.
• Press MODE/CUE to enter editing mode.
• Rotate the SELECT knob until the display shows Track Erase? and then depress the knob.
• The display will show Track OK? Press REC to confirm or MODE/CUE to cancel.
Erasing all tracks:

This function erases an entire disc - beware!
• Press MODE/CUE to enter editing mode
• Rotate the SELECT knob until the display shows All Erase? and then depress the knob.
• The display will show All Erase OK? Press REC to confirm or MODE/CUE to cancel.
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Dividing a track into two:
This function splits a track into two and increases the track number of all following tracks. For
example, if track 3 is divided into track 3 and 4 the original track 4 will become track 5 and so on.
Tracks containing recording cue signals or which have been end-trimmed cannot be divided - clear
cue signals and end-trims before dividing.
• Use the SEARCH buttons to find the approximate point at which the track is to be divided and set
the player to the standby mode.
• Press MODE/CUE to enter editing mode
• Rotate the SELECT knob until the display shows Divide? and then depress the knob.
• The display will show Position OK? The last 3 seconds before the divide point is played
repeatedly
• Use the SEARCH buttons to finely adjust the divide point.
• Press REC to confirm the divide or MODE/CUE to cancel.
Combining two adjacent tracks:

This function joins two adjacent tracks and decreases all following track numbers to fill the gap.
For example, in track 5 is erased, track 6 will become track 5 and so on. Due to limitations of the
MiniDisc format it may not be possible to combine short tracks or tracks with end-trimming - play
them in continuous mode instead.
• Arrange the disc so that the two tracks to be combined are next to each other (using the Move
function if necessary), then select the second track and set the player to the standby mode.
• Rotate the SELECT knob until the display shows Combine? and then depress the knob.
• The display will show Track OK? Press REC to confirm or MODE/CUE to cancel.
Moving a track:
This function moves tracks to any point in the playback order by changing its track number. Other
track numbers are incremented and decremented to accommodate the move. For example, if track 6
is moved to track 3, track 3 will become track 4, track 4 will become track 5 and track 5 will
become track 6.
• Select the track to be moved and set the player to the standby mode.
• Rotate the SELECT knob until the display shows Move? and then depress the knob.
• The display will show Move To 000Tr?
• Rotate SELECT to input the number to which the track is to be moved.
• Press REC to confirm or MODE/CUE to cancel.
Entering a Disc Title:

This function allows the disc to be given a title of up to 255 characters.
• Rotate the SELECT knob until the display shows Disc Name? and then depress the knob.
• Rotate the SELECT knob to select characters. A capital A appears first.
• Press the SELECT knob to enter the selected character.
• Press the DISP/CAPS/NUM button to switch between capital letters, small letters, numbers and
symbols.
• Use the SEARCH buttons to move the cursor. Press SELECT to insert a character. Press END
MON/CLEAR to delete a character.
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Entering a Track Title:
This function allows a track to be given a title of up to 255 characters.
• Set the player to standby mode on the required track.
• Rotate the SELECT knob until the display shows Track Name? and then depress the knob.
• Rotate the SELECT knob to select characters. A capital A appears first.
• Press the SELECT knob to enter the selected character.
• Press the DISP/CAPS/NUM button to switch between capital letters, small letters, numbers and
symbols.
• Use the SEARCH buttons to move the cursor. Press SELECT to insert a character. Press END
MON/CLEAR to delete a character.
End trimming:
This function non-destructively trims the ends of tracks, so that the edit can be undone. It does not
form part of the official MiniDisc specification, and only works on DENON players in SINGLE
mode.
• Set the player to standby at the track whose end is to be trimmed.
• Set end monitor mode by pressing the END MON button.
• Press MODE/CUE when the approximate desired end point is reached. The last 3 seconds before
the trim point will be played repeatedly.
• Use the SEARCH buttons to finely adjust the trim point.
• Press REC to confirm the trim point or MODE/CUE to cancel.
Page 82 of 107
Appendix E
Reverb Unit Instructions
The Yamaha REV100 Reverb Unit has 99 presets which can be modified via front panel rotary
controls for initial delay, reverb time, and high-frequency damping parameters. Input level and
dry/wet balance controls are also provided. Each of the parameter controls also has an LED that
lights when the control setting matches the original preset parameter value.
The full list of available presets is shown below. The “type” of each reverb shows how the effect is
generated by the unit - all the effects listed are modifications of a handful of effect types!
No. Preset Name Type Description

1 Vocal Reverb 1 Vocal REVERB
2 Vocal Reverb 2 Hall
3 Vocal Reverb 3 Vocal This category includes the essential
4 Room Ambience 1 Plate reverb effects such as realistic hall,
5 Room Ambience 2 Plate room, and plate reverb simulations.
6 Room Ambience 3 Plate
7 Wood Booth 1 Vocal
8 Wood Booth 2 Vocal
9 Acoustic Piano Plate + Vocal
10 Club Piano Hall
11 Booming Kick 1 Hall
12 Booming Kick 2 Room
13 Loud Snare Room
14 Acoustic Steel Guitar 1 Plate
15 Acoustic Steel Guitar 2 Plate
16 String Plate Plate
17 Acoustic Gut Guitar 1 Vocal
18 Acoustic Gut Guitar 2 Vocal
19 Brass Room 1 Room
20 Brass Room 2 Room
21 Large Hall 1 Hall STEREO REVERB
22 Large Hall 2 Hall
23 Stage 1 Hall True stereo reverb effects that let you
24 Stage 2 Hall add ambience to stereo signals without
25 Chamber 1 Vocal sacrificing the stereo image.
26 Chamber 2 Hall
27 Church 1 Room
28 Church 2 Hall
29 Old Tunnel Hall
30 New Tunnel Vocal
31 Large Room 1 Room
32 Large Room 2 Room
33 Slide Reverb Room
34 Huge Room 1 Room
35 Huge Room 2 Room
Page 83 of 107

36 Bathroom Plate STEREO REVERB (Continued)
37 String Ensemble Plate
38 Rude Reverb 1 Vocal
39 Rude Reverb 2 Vocal
40 Concert Grand Piano Vocal
41 Small Ambience 1 Hall GATE REVERB
42 Small Ambience 2 Hall
43 Tight Room 1 Room A range of "gated" reverb effects that
44 Tight Room 2 Hall add warmth and ambience while
45 Gate Reverb 1 Plate maintaining a tight sound.
46 Gate Reverb 2 Vocal
47 Gate Reverb 3 Hall
48 Gate Reverb 4 Hall
49 Stone Room Room
50 Big Curve Vocal
51 Analog Delay 1 Delay DELAY
52 Ping Pong Delay Delay
53 Eight Note Triplet Delay A selection of mono and stereo delay
54 Karaoke Delay programs including straightforward one-
55 Short Delay Doubler Delay shot repeats and complex bounce effects.
56 Stereo Long Delay Delay
57 Stereo Medium Delay Delay
58 Stereo Short Delay Delay
59 Mono Long Delay Delay
60 Mono Short Delay Delay
61 Electric Piano Delay + Hall DELAY/REVERB
62 String Pad Delay Hall
63 Synth Delay Vocal Combinations of delay and reverb that
64 Vocal 1 Delay Vocal can add a little more life to your sound
65 Vocal 2 Delay Hall than delay or reverb alone.
66 Vocal 3 Delay + Room
67 Bright Vocal Delay Plate
68 Chorus Delay + Plate
69 Drum Kit 1 Delay + Room
70 Drum Kit 2 Delay Plate
71 Soft Flange 1 Hall + Flange REVERB/MODULATION
72 Soft Flange 2 Hall + Flange
73 Ambience Flange 1 Room Flange Reverb combined with a range of
74 Ambience Flange 2 Room Flange modulation effects including flange,
75 Short Reverb Flange Room Flange symphonic, chorus, and tremolo.
76 Organ Cabinet 1 Plate Flange
77 Organ Cabinet 2 Room Symph.
78 Symphonic Reverb 1 Hall + Symph.
79 Symphonic Reverb 2 Vocal + Symph.
80 Flange Room 1 Vocal Flange
Page 84 of 107

81 Flange Room 2 Room + Flange REVERB/MODULATION (continued)
82 Rolling Flange 1 Plate + Flange
83 Rolling Flange 2 Plate + Flange
84 Big Flange Vocal Flange
85 Chorus Reverb 1 Hall + Chorus
86 Chorus Reverb 2 Plate + Chorus
87 Chorus Reverb 3 Hall + Chorus
88 Chorus Reverb 4 Vocal + Chorus
89 Tremolo Reverb 1 Hall + Tremolo
90 Tremolo Reverb 2 Room Tremolo
91 Tremolo Reverb 3 Plate + Tremolo
92 Tremolo Reverb 4 Vocal + Tremolo
93 Tremolo Reverb 5 Vocal + Tremolo
96 Ambient Slow Pan 1 Hall + Tremolo
97 Ambient Slow Pan 2 Room + Tremolo
98 Sequence Pan 1 Room + Tremolo
99 Sequence Pan 2 Room + Tremolo
Technical specification:
Signal processing: 16 bit

Sample rate: 44.1kHz
Frequency response: 20Hz - 20kHz
Input/output levels: -10dBu nominally into 20k
stereo unbalanced THD: less than 0.1% @ 1kHz
Dynamic Range: 80dB typical
Page 85 of 107
Appendix F
ADC Patchbay Layouts
There are five patchbays at the ADC. These are as follows:
Name Connectors Location Normalling

Microphone Patchbay XLR Sound Box No
Main Patchbay “B” Jack Sound Box Half
Ring Intercom Patchbay “B” Jack Amp Rack to rings
Amp Rack Patchbay “B” Jack Amp Rack Full
Speaker Patchbay Speakon Amp Rack No
Tielines exist between the microphone and main patchbays, and between the main and amp rack
patchbays.
Note that due to limitations of space, inputs and outputs have been mixed in the same row on a few
of the patchbays.
On the amp rack patchbay and the speaker patchbay, sets of three or four paralleled sockets exist to
split signals; as these patchbays are either fully normalled or not normalled, there is no other way to
do this conveniently.
Sockets on the ring intercom patchbay are normalled to their allocated ring. To reassign a station to
another ring, a patchlead would be inserted from that station’s socket to one of the ring sockets.
Note that the sockets marked “Ring C” are paralleled together and powered, but not permanently
connected to any stations.
Between the microphone patchbay and the main patchbay is a cable tester for XLR and jack leads.
To test a lead, plug one end into the top row of sockets and the other end into the bottom row of
sockets and press the button; three green LEDs should light; If any red LEDs light, the cable has a
core to core short, and if fewer than three green LEDs light, there is a cable break.
The patchbays are shown in Figure 33 and Figure 34.
Page 86 of 107
Figure 33 - The microphone and main patchbays, also showing the cable tester
Figure 34 - The ring intercom, amp rack and speaker patchbays
Page 87 of 107
Microphone Patchbay
No. 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16
Main Patch Links (male) Stage Box Pit
1 2 3 4 A1 A2 A3 A4 A5 A6 A7 A8 B1 B2 B3 B4
1 2 3 4 A9 A10 A11 A12 A13 A14 A15 B5 B6 B7 B8
Main Patch Links (female) Stage Box Pit
No. 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16
OP Lad. OP Perch OP Juliette OP Downstage OP Midstage OP Upstage OP Hemps
C1 E1 E2 G1 G2 I1 I2 I3 I4 K1 K2 M1 M2 O1 O2
D1 F1 F2 H1 H2 J1 J2 J3 J4 L1 L2 N1 N2 P1 P2 Show
PS Lad. PS Perches PS Juliette PS Downstage PS Midstage Counterweights Dome Relay
No. 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16
Mixer Mic Inputs Bar Zero
1 2 3 4 5 6 7 8 9 10 11 12 Q1 Q2 Q3 Q4
13 14 15 16 17 18 19 20 21 22
Mixer Mic Inputs
Main Patchbay
Page 88 of 107
Ring Intercom Patchbay

No. 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20
SM MS PS MS OP US Hemps Cwts Pit and Perches Sound LX Box LD Ring A Ring B Ring C
DS PS Scene Dir Patch Box / Lime (Aud)
Dock (Aud) Dome
Normalled to Ring A Normalled to Ring B Not normalled to any stations
Amp Rack Patchbay

No. 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20
Amp Rack Tie Lines SM Desk Link Bose EQ returns
Out 1 2 3 4 5 6 7 8 L R 1 2 3 4 5 6 7 8 9 10
Norm | | | | | | | | | | | | | | | | | |
In 1 2 3 4 5 6 7 8 9 10 1 2 3 4 5 6 7 8 9 10
Bose EQ sends Power Amp Sends
Speaker Patchbay
No. 1 2 3 4 5 6 7 8 9 10
Out Amp 1 Amp 2 Amp 3 Amp 4 Amp 5 Amp 6 Amp 7 Amp 8
In RoH OP FoH OP OP Ladder OP Perch OP Juliette DS OP MS OP Pit
In RoH PS FoH PS PS Ladder PS Perch PS Juliette DS PS MS PS US PS
Common Common Common
Page 89 of 107
Appendix G
ADC Sound Effects Library
The ADC possesses a library of sound effects CDs which can be used by any shows using the
theatre. A detailed list is here. Outside shows can use the CDs, but there may be a charge, and the
CDs must not normally be removed from the theatre, so all editing should be done in the theatre.
Please check all the CDs are there when you are given them, as you will be held responsible for any
missing on return.
CD Title No. Manufacturer's Code

Spectacular Sound Effects (Vol. 1) 1 EMI CZ350
Spectacular Sound Effects (Vol. 2) 2 EMI CZ351
101 Sound FX 3 EFX002
Workshop of Sound 4 EFX004
Essential Comedy Sound Effects (Vol. 1) 5 BBC CD843
Sound FX - Frights of the Night 6 EFX001
Essential Sound Effects 7 BBC CD792
Essential People Sound Effects 8 BBC CD863
Essential Crowd Sound Effects 9 BBC CD862
Essential Sounds of the Countryside 10 BBC CD861
Essential Sounds of the City 11 BBC CD860
Essential Foreign Sound Effects 12 BBC CD870
Essential Weather Sound Effects 13 BBC CD868
Essential Seasonal Birdsong (Woodland and Garden Birds) 14 BBC CD846
Essential Home Video Sound Effects 15 BBC CD853
30 Years at the Radiophonic Workshop 16 BBC CD871
Essential Science Fiction Sound Effects (Vol. 1) 17 BBC CD847
Essential Science Fiction Sound Effects (Vol. 2) 18 BBC CD855
Essential Death and Horror Sound Effects (Vol. 1) 19 BBC CD822
Essential Death and Horror Sound Effects (Vol. 2) 20 BBC CD823
Essential Combat and Disaster Sound Effects 23 BBC CD839
Totally Gross Sound FX from Hell 24 CDGR500
Essential Sound Effects of England 25 BBC CD867
Essential Animal Sound Effects 26 BBC CD869
Essential Transport Sound Effects (Vol. 1) Land 27 BBC CD865
Essential Transport Sound Effects (Vol. 2) Air and Water 28 BBC CD866
Essential Sound Effects of Babies and Children 29 BBC CD864
30
Page 90 of 107
Appendix H
Sound levels, frequencies and other figures
Whilst basic sound design can be done using common sense and trial and error, it is useful to have a
grasp of the units in which sound is measured. This allows the full relevance of the markings on
controls and meters to be understood, and assists communication with other sound engineers and
musicians. It also allows effects such as pitch shifts and equalisation to be quickly and precisely
engineered.
Although this section touches the surface of some very complex concepts, it has been written to be
comprehensible to as many people as possible, and examples have been included to give non-
mathematicians an idea of what the maths means. Don’t worry if you do not follow all of this
section - it isn’t essential, and just a flavour of this subject will be of great benefit!
Sound Levels
The ear is an amazing device. The power contained in the loudest sound which the ear can stand
(the threshold of pain) is 1012 (1,000,000,000,000) times greater than the power contained in the
quietest sound which the ear can detect (the threshold of hearing.) This gives an awkwardly large
scale of figures with which to describe how loud a sound is!
However, the ear provides a solution in that as the sound level increases, the ear’s sensitivity
decreases; this results in the perceived volume of a sound being logarithmically proportional to its
power. It is therefore convenient to use a logarithmic unit to measure sound levels. The problem
with this is that logarithms are ratios and not absolute values, so a reference value has to be
specified before absolute measurements can be taken.
Perceived
Volume
Sound Level
Figure 35 - The ear's logarithmic response
The logarithmic unit used for sound is the decibel (i.e. a tenth of a Bel); this is commonly
abbreviated to dB, and is often suffixed by an indication of the reference being used, although it is
sometimes left unclear.
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The simplest use for a decibel is as a pure ratio. For example, a mixer or an amplifier takes a signal
in at one level and sends it out at a different level. The decibel can be used to express the ratio
between the levels of these two signals. Note how the faders on a mixer are labelled:
• - dB represents infinite loss - however high the input level is, there will be no output.
• other negative values indicate loss - a lower output level than input level.
• 0 dB represents the same input and output levels - the signal level is unchanged.
• positive values indicate gain - a higher output level than input level.
When controlling the voltage of a signal (e.g. with a mixer fader) note that an increase of 3dB
doubles the perceived volume of sound.
Sound pressure levels (SPLs) are often used to express how loud a sound is; they are measured in
decibels using a reference of 0dB representing 20 µP (micropascals), which is the quietest sound
that the human ear can detect; this is called the “Threshold of Hearing.”
When measuring the SPL from a single source (i.e. not background noise from many sources) it is
important to specify from how far away the measurement is being taken - this is normally
1 metre. As a sound spreads out, it becomes quieter; the measured SPL of a sound typically drops by
6 dB every time the distance between the source and the measurement is doubled. Note that a drop
of 6dB equals a halving of perceived volume.
The following table gives an idea of typical values for sound pressure levels:
dBSPL Example
200 Heavy artillery
130 Jet aircraft taking off
120 Threshold of Pain
110 Pneumatic drill at 1 metre
100 Typical disco sound level
90 Symphony orchestra at its loudest
80 Vacuum cleaner at 1 metre
70 Speaking voice at 1 metre
60 General office noise
Orchestral woodwind solos
50 Whisper at 1 metre
Orchestra string section playing at its quietest
40
30 Quiet room
20
10 Quiet countryside
0 Threshold of Hearing
A further complication is that the ear does not respond equally to all frequencies. Therefore, when
figures are used to indicate how loud a human will perceive a sound to be, a “weighting curve” will
be applied to modify the figures to take account of this perception. Although there are many of
these, the most common for audio work is the “A” curve, and figures which have been modified in
this manner are usually shown with units of dBA.
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For measuring the voltage of audio signals, a reference of 0dB is used to represent 1V. Levels
measured in this way are sometimes designated dBV. For reference, a microphone will produce
voltages in the order of 1mV (-60dBV), a record deck will produce voltages in the order of 3mV
( 50dBV), and a line level signal is in the order of 150mV (-16dBV.) Loudspeaker levels are
measured in volts.
When measuring the power of audio signals, a reference of 0dB is used to represent 1mW being
driven into a load of 600 , which equates to 0.775V. Levels measured in this way are often
designated dBm. Likewise, levels are sometimes measured using a reference of 0dB to represent
1µW being driven into a load of 600 , which equates to 24.5mV. Levels measured in this way are
often designated dBU.
A summary of the references is shown below:
Unit 0 dB reference
dBSPL or dBA (weighted) 20 µP
dBV 1 mV
dBm 1 mW
dBU 1 µW
Professional equipment is normally specified with an operating level (i.e. 0dBSPL) of 4dBU (i.e. 1.22
V into 600 ), whilst semi-professional equipment is normally specified to run at an operating level
of -10dBV (i.e. 0.316 V.)
Metering Sound Levels

There are two main type of sound level meter: volume unit meters and peak programme meters. The
“best” type of metering to use in a particular situation is often down to personal preference; get
experience of using both types and learn the advantages and disadvantages of each.
VU (Volume Unit) meters are the most common form of sound metering; They simply track the
level of the audio signal and display it . Whilst they are intuitive to use, they sometimes “average”
the signal due to the finite speed at which the meter can move, and so the true level of peaks in the
signal is not shown.
Peak Programme Meters (PPMs) are designed to rise quickly and fall slowly, so indicating peak
levels in an audio signal. Apart from showing the “worst case” signal level scenario, they move
more slowly and so are less tiring to watch than VU meters. The scale is marked from 1 to 7, and
different types of material should peak at different levels. The policy on peak levels is usually
decided locally, but compressed material is usually allowed to peak at lower levels the
uncompressed material, as it is perceived to sound louder. PPMs are often used in broadcasting, but
can also be useful for monitoring recording levels. As these meters require accurate drive circuitry,
they are considerable more expensive than VU meters.
Note that as the various standards organisations cannot agree on how fast the meter should rise and
fall, there are seven common calibration standards for PPMs, of which the most common are DIN,
BBC and Nordic.
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Hearing Damage
The range of human hearing is typically 20Hz to 20 kHz for a healthy person under 30. This upper
limit decreases with age and noise exposure.
Although 120 dBSPL is referred to as the “Threshold of Pain” and often taken to be the point at
which hearing damage starts, it can actually occur at much lower levels, and the longer the exposure
to high SPLs, the greater the risk of permanent hearing damage. There is a legal limit for the length
of time that exposure to certain SPLs may occur; this ranges from 8 hours for 90dBA to 3 minutes
45 seconds for 111dBA.
A loud pair of headphones can approach levels of 100 dBA at close range, giving great scope for
hearing impairment. Therefore, use sensible levels when monitoring over headphones.
Octaves and Frequencies

Engineers work in frequencies. Musicians work in octaves. Sound Engineers are caught in the
middle. As might be expected from this generalisation, frequencies are more logical, but octaves are
more intuitive.
Sound is made up from pulses of air pressure, and the rapidity at which the pulses follow each other
determines the pitch of the sound. The frequency of a note (i.e. a pure tone) is simply the number of
pulses arriving in a fixed time; frequency is usually measured in Hertz (or cycles per second), so
when you hear a note at 500Hz, 500 pulses of air are entering your ear every second. It is therefore
possible to precisely define a note by quoting its frequency.
As with other musical “intervals”, an octave is a ratio. It does not define a note, but defines a
relationship between two notes. If two notes are an octave apart, the frequency of the higher note is
twice that of the lower note. Likewise, a ratio of 1.5 exists for a “perfect fifth” and a ratio of 1.4 for
a “perfect fourth.” It is worth noting that most good musicians can tell the interval between two
notes, but only the lucky few with “perfect pitch” can tell the actual pitch (and therefore frequency)
of a note. Sound engineers with perfect pitch have a distinct advantage when using a graphic
equaliser to notch out feedback!
As music is expressed in terms of these ratios, it is therefore possible to raise or lower a frequency,
and so long as all the other frequencies are altered in the same ratio, the music will still sound
musical. This is how “pitch” adjustments on record decks ,tape decks and “varispeed” CD players
work. To take an extreme example, if a tape is played at double speed, as the individual pulses
which make up notes will pass the tape head at twice the speed, the reproduced sound will consist of
notes of twice the frequency, and hence playback will be an octave higher than recording.
With this flexibility, it is sometimes difficult to pin musicians down to which frequencies they are
using! Orchestras usually tune up to treble A (traditionally given by the oboe) which corresponds to
440Hz in “concert pitch” which is now generally used. Note than scientists take middle C as 512 Hz
as it halves nicely; this contrasts to 523.2 Hz in concert pitch, which is slightly sharper.
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Appendix I A guide to compression
Dynamic compression is one of the most commonly used processes when it comes to mixing,
probably second only to equalisation. However, there are many different kinds of compressor and
even more ways of using them, each producing different end results with a variety of side effects.
Compressors all work by pulling loud things (any signal above the threshold) down in level. The
amount of reduction is determined by the ratio control in combination with the threshold, and is
shown on the gain-reduction meter. For example, if a ratio of 5:1 is set then an input exceeding the
threshold by 5dB will be output with a level of only 1dB above the threshold; once the signal falls
back below the threshold level, the gain returns to normal. Low thresholds mean the compression
starts at a lower level, so there will be a lot of gain reduction. High ratios squash signals above the
threshold harder, also introducing a lot of gain reduction. It is, therefore, normal to balance
threshold and ratio so that low thresholds tend to be associated with gentle ratios, while high
thresholds have steeper ratios.
Overall Compression
The simplest way to use a compressor on the mix is to apply it across the overall stereo mix.
Ideally, this should always be done by patching it across the desk's main stereo bus insert points - if
you simply plug a compressor between the desk's output and the amplifier's input you can no longer
perform fade-outs using the desk's master faders as this will affect how hard the compressor is being
driven and possibly therefore the sound or balance of the mix during the fade-out.
Whenever you are using any kind of dynamic control device on a stereo source, it is absolutely vital
to always use the stereo link mode so that both channels are processed in the same way. Whichever
channel contains the loudest signal peak at a given moment, the resulting gain reduction is applied
equally to both channels, maintaining the stereo image.
There are two basic jobs that you might wish the compressor to do; the first of which is to control
signal peaks without affecting the dynamic range of the rest of the piece. The usual approach here is
to set the threshold to a value just above the average music level and to use a ratio of between 2.5:1
and 8:1 to reduce the gain of the peaks. It is sometimes easier to set up the threshold with a very
high ratio set so that the gain reduction meters will kick in very obviously when a signal peak
exceeds threshold; and then revert the ratio to a more suitable level, giving a maximum gain
reduction of between 6dB and 10dB.
The second basic way to use a compressor is to reduce the dynamic range of the entire signal - not
just the peaks - so the whole thing sounds that little bit louder and more energetic and impressive. In
this case a low ratio of between 1.1:1 and 2:1 would be used, with the threshold set to around 20dB
or 30dB below the peak level.
However, putting a compressor across the output of the desk will actually make it all quieter...
which is not what we want at all, so we have to use the gain control to bring the overall level back
up again. The effect of this combined processing is to make the quieter elements of the mix louder,
whilst the loud stuff stays more or less where it was
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While the concept is simple, the application is a little more involved. The first point to note is that
the gain make-up brings up all quiet signals - tape and mixer hiss, hums, ambient noises, and
reverberation - it does not discriminate between wanted and unwanted sounds, so it has to be used
carefully. The second point is that the amount of compression applied at any moment in time is
determined by the loudest signal in the mix. This might seem obvious, but it is very important,
because whichever signals are dominant are going to modulate the level of the entire mix.
The Controls
The key to making compression work lies in setting the compressor sympathetically. Typically, a
little gentle, subtle compression is all that is needed, with a very low ratio (perhaps 1.5:1 or 2:1)
combined with a low threshold. If you are after a harder, more obviously compressed effect you will
need to use higher ratios such as 3:1 or maybe even 5:1, but with a much higher threshold.
Setting the threshold and ratio is only half the story though, and the controls which make a large
difference, are the release and attack times. The attack time determines how long the compressor
takes to reduce the gain once the input signal has passed the threshold, while the release time
determines how long the gain takes to return to normal after the input signal has fallen back below
the threshold.
Most compressors have an ‘auto’ mode which works just fine, giving fast releases for large but
brief transient peaks, and a slower time constant for smaller peaks. However, if your machine does
not have an automatic setting you will have to adjust the attack and release times by hand.
Multi-band Compression
One of the problems with overall compression is that if any one sound is dominant, that sound will
drive the compressor. Sometimes this is fine and just what is wanted, but more often than not it is a
problem looking for a solution. A common fix is to insert an equaliser in the side-chain of the
compressor. If the kick drum or bass guitar are dominating the compressor, using the equaliser to
turn down the lower frequencies present in the side-chain will reduce the compressor's sensitivity to
them (effectively giving a higher threshold for low frequencies compared to everything else). Now
the mid-range signals will tend to dominate, which is probably a lot more useful.
A more elaborate but far more effective solution is to employ a multi-band compressor. The multi-
band compressor splits the input signal into three or more frequency bands, each being processed by
a separate compressor before being recombined at the output. The advantage is that bass signals can
be controlled and squashed as necessary, as can the mid-range signals, but loud peaks in any one
section will not affect the levels of the others.
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Glossary of Sound and Theatre Terms
µP micropascal; a millionth of a Pascal.

µW microwatt; a millionth of a Watt.
A see Amp.
A-B microphone A pair of spaced microphones used for a stereo pickup.
pair
AC Alternating Current.
A/D Converter Circuit for converting analogue waveforms into a series of equally spaced
numerical values represented by binary numbers (digital). The more 'bits' a
converter has, the greater the resolution of the sampling process.
ADC Amateur Dramatic Club.
AF Audio Frequency
AFL After Fade listen; a system used within mixing consoles to allow specific
signals to be monitored at the level set by their fader of level control knob. Aux.
sends are generally monitored AFL rather than PFL.
Amp(ere) A unit which measures the “flow” of electricity.
Analogue Circuitry that uses a continually changing voltage or current to represent a
signal. Compare with "Digital"
Anechoic Echo free; an anechoic room is a room whose walls, ceiling, and floor are
covered with a sound-absorbing material.
Attenuation Reduction in a signal level; the opposite of gain. See page 92.
Auxiliary mix A mix which is subordinate to the main mix. Facilities provided for this mix are
more sparse (e.g. rotary instead of linear faders) and it is often used to feed
effects units or foldback speakers. See page 34.
Balance A control which varies the balance between the level of the channel of a stereo
signal, usually by attenuating one channel (e.g. if the control is turned to the
left, the right channel is attenuated.) This changes the perceived placing of the
stereo signal within the stereo image. Compare with Pan.
Balanced A sound connection where the signal is split into two halves, one of which is
inverted, to reduce interference. See page 53.
Band • A portion of the frequency spectrum. The term is often used to refer to a
frequency range on an equaliser.
• The people who provide the music!
Band Riser A raised area, usually towards the back of the stage, where the band are
situated. These are usually built as required rather than being an integral part of
a theatre.
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Bandwidth • The difference between the lowest and the highest frequencies which may be
reproduced by a recording medium.
• The difference between the lowest and the highest frequencies which are
affected by a filter.
For analogue filters which gradually takes effect over a frequency range, the
point measured is the point at which the amplitude of the signal has been halved
(the “-3dB point.”)
Bar Zero The lighting bar immediately on the auditorium side of the Proscenium Arch.
Bias A high frequency a.c. signal added to the signal used to record on magnetic
media to ensure that the most sensitive region of the recording material’s
magnetic properties are used and to drive the erase head. Bias is generated by a
bias oscillator
Bit a binary digit (i.e. a “0” or a “1”.) These are the basic blocks of information
passed around digital systems.
BNC Bayonet connector
BOP Bastard opposite prompt. An OP position on the left of the stage as seen from
the auditorium.
BPS Bastard prompt side. An PS position on the right of the stage as seen from the
auditorium.
Bulk Eraser A device for erasing magnetic recording media, either by running it past an
erase head at high speed or by immersing the entire recording media into a
chamber with a magnetic field.
Bus A device (often a single conductor) which distributes a signal by collecting a set
of inputs and feeding a set of outputs. In a mixer, signals are fed onto a bus
from the input modules and then fed to output modules from the bus. See page
31.
Cans The ring intercom system used for backstage communications during a show.
Capacitor • A type of microphone. See page 10.
• A device which stores electrical charge.
Cardioid Meaning ‘heart-shaped’; describes a microphone polar pattern. See page 10.
Carrier A signal which is modified (modulated) in some way to carry information, as
opposed to sending the information out as a signal in its own right. Carrier
signals are usually chosen for their propagation characteristics (i.e. how far they
travel and how much noise they pick up.) The frequencies by which radio
stations are often identified are those of the carrier frequencies upon which the
broadcast signals are superimposed.
Cartridge Tape An obsolescent recording media widely used in radio stations for its fast cueing
(“Cart”) and simplicity of operation.
CD Compact Disc. See page 19.
Centre Stage The centre of the stage widthways.
Chorus An effect where a signal is delayed and added to the original sound to add
“fullness”.
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Chromium A magnetic coating used on recording tape. See page 17.
Dioxide
Clipper A dynamic processor. See page Error! Bookmark not defined..
CMRR Common Mode Rejection Ratio
Co-channel Interference between adjacent channels, where one signal becomes audible
interference upon another signal.. See page 51.
Cold Signal The out of phase component of a balanced audio signal. See page 53.
Compressor A dynamic processor. See page 30.
Condenser A type of microphone. See page 10.
Counterweights • A system for lifting heavy loads in the fly tower by balancing the load with
an equal load (formed by weights in a cradle) acting in the opposite
direction.
• The gallery from which the counterweights are operated. At the ADC, this
runs along the OP side of stage.
CrO2 See Chromium Dioxide.
Crosstalk Interference between adjacent channels. See page 51.
Cue • (as a noun) a single, identifiable action required of the sound operator, or
sound material relating to this action.
• (as a verb) to instruct the sound operator to perform a predefined action, or
to prepare sound material for this occurrence.
e.g: “The sound operator cued (prepared) the cassette to its cue-point (the point
from which the predefined action starts) ready for the Stage Manager to cue him
(tell him to go) with his cue (a predefined action.)” See page 39.
CVBS Composite Video Broadcast Standard.
Cwts See Counterweights.
D2B A digital video broadcasting standard.
D54 A multiplexing protocol for lighting control, invented by Strand.
DAT Digital Audio Tape. See page 19.
dB A decibel. See page 91.
DC Direct Current.
Decibel See page 91
De-essing Removing the sibilant components of a vocal signal. This is done using a
compressor with an equaliser highlighting the sibilant frequencies (6 - 10 kHz)
inserted into the sidechain.
DI Direct Injection (Box). See page 54.
DIN Deutsche Industrie Normal. A set of German standards. Often used to refer to
multipole plugs conforming to these standards. See page 69.
DisCart A recording system designed to operate like cartridge tape, but improving
sound quality and editing facilities by recording digitally onto 3.5 inch disks.
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Dolby A form of noise reduction. See page 19.
Dome A lighting and microphone position formed by a masked opening in the ceiling
of the auditorium.
Downstage The front of the stage.
Drain wire A bare metal wire in contact with the screen along the length of a cable, used
for terminating the screen. See page 76.
Dry Signal A signal which has not been processed. See page 25.
DS See Downstage.
Ducking Temporarily lowering the level of one signal to enable a second signal (usually
a voice-over) to be heard more clearly over it. This is done using a compressor
on the signal to be ducked but feeding the signal to be heard over the top into
the sidechain.
Dynamic A type of microphone. See page 10.
Dynamics Changes in the “volume” or level of a signal. Dynamic processors alter the
volume of a signal to modify the dynamic range (the range over which the
volume varies.) See page 30.
Effect A device which is used to modify a proportion of an audio signal which is then
fed back into the main mix c.f. Processor.) See page 25.
EIA Electronic Industries Association
Electret A type of microphone. See page 10.
Entr’acte A specially scored piece of music used to cover a pause between two acts.
Ents Entertainments.
EQ Equalisation. See page 28.
Equaliser A device which allows narrow bands of frequency in an audio signal to be
boosted or cut as desired, usually to correct for distortion, to remove feedback,
or to enhance the sound.
Euroconnector Another name for a SCART connector. See page 72.
Expander A dynamic processor. See page 30.
Feedback A whine caused by sound “chasing itself” around a system. See page 13.
Female connector A socket.
Ferric A magnetic coating used on recording tape. See page 17.
FerroChrome A magnetic coating used on recording tape. See page 17.
Figure-of-Eight A microphone polar pattern. See page 10.
Flange An effect where the phase of a signal is slightly modified and then added to the
original signal, leading to interplay between the two signals.
Float A microphone rigged on the front of stage to pick up stage dialogue.
Fly Tower The tower above the stage, into which scenery and lighting may be raised
(“flown.”)
FoH See Front of House.
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Foldback A set of speakers (and the system which drives them) so that performers can
hear themselves and other relevant parts of a performance. See page 37.
Forestage The part of the stage in front of the Proscenium Arch. This covers the orchestra
pit, and can be flat, stepped (by inserting different segments), or removed
entirely.
Frequency The number of times that something happens in a given period of time. Often
used to count the pulses of air in a sound wave, thus quantifying the pitch of a
musical note. See page 94.
Front of House • The “public” area of the theatre, including the public corridors, toilets, box
office foyer, bar, and the auditorium up to the Proscenium Arch.
• The auditorium speaker positions closest to the stage.
FX Effects.
Gaffer tape A strong cloth-backed tape which sticks anything to anything. Usually available
in black, white and silver. Sometimes referred to as duct tape.
Gain Amplification; the opposite of attenuation. See page 92.
Gash A bar specially rigged in an unusual position, and the wiring required to provide
lanterns, microphones and speakers on this bar.
Gate A dynamic processor. See page 30.
GHz see GigaHertz.
GigaHertz 1000 MHz.
Grid The arrangement of flown bars above stage.
Group A sub-mix on a large mixer. See page 34.
Head A magnetic or optical device for reading from or writing to storage media.
Headroom The difference between the operating level and the maximum level of a
recording media. See page 16.
Hemps • A system for lifting loads in the fly tower using a set of ropes fed over a
system of pulleys.
• The gallery from which the hemps are operated. At the ADC, this runs along
the PS side of stage.
HF High Frequency
Hot Signal The in-phase component of a balanced audio signal. See page 53.
Howlround Feedback. See page 13.
High pass filter A filter which attenuates frequencies below its cutoff frequency.
(HPF):
HST Helical Screened Twin. A type of microphone cable. See page 77.
Hypercardioid A microphone polar pattern. See page 10.
Hz Hertz. A measure of frequency. See page 94.
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IEC Internationale Electrotechnique Commission. An organisation which has set
international standards covering connectors and pre-equalisation standards (see
page 21) amongst other things.
Impedance Can be visualised as the 'AC resistance' of a circuit which contains both
resistive and reactive (inductive) components - e.g. speakers and microphones.
in rep In repertoire. Several shows being performed alternately in the same venue.
Induction Loop A wire loop (and the system which drives it), usually around an auditorium,
which emits an unmodulated audio frequency electromagnetic signal fed from
microphones and direct signal sources as appropriate. This can picked up by
coils in hearing aids (switched to the “T” position) allowing hearing aid users to
receive a clearer signal than the microphone built into the aid would allow. See
page 55.
ips Inches per second. Used to measure the speed of recording medium (usually
reel to reel tape.)
kHz kiloHertz - 1,000 Hertz.
kiloOhm (k ) 1,000 Ohms.
Ladder • A device to help people reach high things!
• An arrangement of lighting bars which resembles a device to help people
reach high things (often referring to two lighting ladders permanently fixed
to the side walls of the auditorium close to the stage.)
Lavalier A microphone worn on clothing (e.g. in a tieclip) or on the body. Often used as
radio microphones.
Leader tape Plastic tape with no information storage capability which is inserted into a
magnetic tape reel. See page 21.
Lemniscate A microphone polar pattern. See page 10.
LF Low Frequency
Limiter A dynamic processor. See page 30.
LX Lighting.
LX tape PVC tape coated with an adequate, but not over-powerful adhesive. Useful for
securing and coiling cables, labelling and insulating.
mA milliamp; a thousandth of an Amp.
Male connector A plug.
Mask To conceal something from the audience. (e.g. if a stage microphone cannot be
seen from the auditorium, it is said to be masked.)
MD • MiniDisc. See page 20.
• Musical Director. The person in charge of the band in a musical.
Metal A magnetic coating used on recording tape. See page 17.
MHz MegaHertz - 1,000,000 Hertz.
MIDI Musical Instrument Digital Interface. See page 43.
Midstage The centre of the stage depthways.
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MiniDisc See page 20.
Mix A collection of audio input signals, in the correct proportion to form a useful
output.
Monitor • A speaker or set of speakers used to listen to a signal for reference purposes
rather than to use it for its intended purpose.
• A video display.
Mono Monophonic. An audio system providing a single channel of sound, and hence
no directional information.
MS See Midstage.
M/S Mid/Side. A method of stereo recording, as opposed to A/B (left/right)
recording.
Multiplexing Sending several signals down a single communication channel by encoding
them in some manner, which can sometimes lead to a loss in quality. Often
used for lighting control (e.g. D54.)
MUX See multiplexing.
mV millivolt; a thousandth of a Volt.
mW milliwatt; a thousandth of a Watt.
NAB National Association of Broadcasters. An organisation which has set standards
for tape pre-equalisation (see page 21) amongst other things.
Near field Some people prefer the term 'close field', to describe a loudspeaker system
designed to be used close to the listener. The advantage is that the listener hears
more of the direct sound from the speakers and less of the reflected sound from
the room.
Notching • (as a verb) filtering out a narrow band of frequencies (e.g. to remove
feedback.)
• (as an adjective) a filter specially designed to filter out a narrow band of
frequencies with a high Q factor and an adjustable centre frequency.
Octave A musical interval which corresponds to a doubling of frequency. See page 94.
OFC Oxygen Free Copper. See page 77.
Ohm ( ) A unit which measures electrical resistance.
Omni (- A microphone polar pattern. See page 10.
directional)
OP See Opposite Prompt.
Op. Operator.
Opposite Prompt The side of the stage opposite that in which the Stage Manager (prompt) sits.
Usually on the right of the stage as seen from the auditorium. (The ADC has
BOP.)
Overture A specially scored piece of music played before the start of a performance.
Pa see Pascal.
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PA Public Address
Pad A device for reducing the level of a signal. This could either be an electrical pad
(e.g. an adapter containing an electrical network to reduce a voltage level, or a
control on a mixing desk or microphone to switch in such a circuit) or a
mechanical pad (e.g. is attached to a microphone to restrict the sound pressure
levels reaching the diaphragm).
Pan A control for placing a mono signal within a stereo image. Compare with
Balance.
Parametric EQ Equalisation where the centre frequency of the EQ band can be adjusted. The Q
factor (width of the band) is usually fixed, but can be adjusted on the more
expensive units.
Pascal A unit of pressure.
Patch • (as a verb) to allocate the correspondence between inputs and outputs,
usually using a patchbay.
• (as a noun) a correspondence between inputs and outputs, usually on a
patchbay or in a synthesiser.
See page 48.
Peak The maximum level of a signal or the maximum permissible level to which an
input may be driven. A signal exceeding this level is said to be peaking.
Peak Limiter A dynamic processor. See page 30.
Perch A lighting and microphone position formed by a masked opening on the side of
the Proscenium Arch.
Peritel Another name for a SCART connector. See page 72.
PFL Pre-fade listen. See page 34.
Phantom Power A method of sending power down a microphone cable. See page 54.
Phase The timing difference between two electrical waveforms expressed in degrees
where 360 degrees corresponds to a delay of exactly one cycle.
Phonoflex A single core flexible cable with a conductive plastic screen. See page 77.
Pit The orchestra pit. The area underneath the forestage. The forestage can be
removed to create an open pit, or left in place to create a closed pit.
PO Post Office. A set of specifications for connectors. See page 48.
Polar Pattern The correlation between where a sound source is and how much of it a
microphone picks up. See page 10.
Popshield A fine mesh supported by a fixed ring , which is placed between a vocalist’s
mouth and a microphone. The mesh diffuses any large blasts of air (e.g. from
“p” and “b” sounds) and so reduces “popping”.
PPL Phonographic Performance Licence. See page 24.
PPM Peak Programme Meter. See page 93.
Page 104 of 107

Pre-equalisation To allow for a known (usually unavoidable) distortion which a signal is about
to undergo by modifying it with the inverse distortion, so that the second
distortion cancels out the inverse distortion, leaving a distortion-free signal.
Often used with speakers, where the known frequency characteristics of the
speaker are compensated for by pre-equalisation, thus giving the speaker a
much more linear response.
Processor A device which is used to modify an audio signal in its entirety (c.f. Effect.) See
page 25.
Prompt Side The side of the stage where the Stage Manager (prompt) sits. Usually on the left
of the stage as seen from the auditorium. (The ADC has BPS.)
Pros(cenium) The arch which separates the forestage (in the auditorium) from the main part
arch of the stage (under the fly tower.)
PRS Performing Rights Society. See page 24.
PS see Prompt Side.
PZM Pressure Zone Microphone. See page 11.
Q factor A measure of how wide the frequency range is over which an analogue filter
takes effect. A high Q factor gives a filter which affects a narrow range of
frequencies, useful for removing feedback without unduly affecting the wanted
portion of the sound. A low Q factor gives a broad filter useful for “colouring”
the sound.
Quad • The area behind the Juliettes, used for storage and access at the ADC.
• Quadraphonic. An audio system providing four channels of sound, hence
giving the listener spatial information in two dimensions (usually “left /
right” and “front / rear”.) Very few recordings are in quad, but quad speaker
setups are common in the theatre.
quasi-balancing A wiring technique to connect a balanced output to an unbalanced input and
retain some of the benefits of balanced line operation. See page 54.
RCCB Residual Current Circuit Breaker. See page 55.
Rear of House The auditorium speaker positions furthest from the stage
Release The time taken for a level or gain to return to normal.
Repatch To alter the patching between shows or during a show. See page 49.
RFI Radio Frequency Interference. See page 51.
RIAA Recording Industry Association of America. An organisation which has set
standards for phonographic pre-equalisation (see page 22) amongst other things.
Rifle A microphone polar pattern. See page 10.
RoH See Rear of House.
RS • Recommended Standard (e.g. RS232, a common protocol for serial data
transmission)
• Radio Spares, a very large supplier of electronic components and equipment.
Their catalogue is very complete, so many manufacturers quote spare parts
with an RS catalogue number. The downside of this comprehensive selection
is that prices are often rather high.
Page 105 of 107

Sampling The rate (Hz) at which an analogue signal is measured (“sampled”) to turn it
Frequency into a digital signal. The highest audio frequency which can be reproduced is
half the sampling frequency.
SCART A multipole audio and video connector. See page 72.
Scene Dock The area under the stage where scenery is built and stored.
Screen A conductive layer to provided to exclude interference, and often to provide a
return path for a signal. See page 76.
Shotgun A microphone polar pattern. See page 10.
Show Relay A system by which the action on stage can be monitored in other areas. This
can refer to audio and video systems.
Sibilant A whistling or hissing sound. Often refers to the hissing effect when “s”, “sh”
and “z” sounds are picked up by a microphone at close range.
Sidechain The signal in a compressor, expander, limiter or gate which is monitored to
determine the level of attenuation to be applied to the main signal. By default,
the sidechain signal is the same as the main signal, but for specialist work such
as de-essing or “ducking”, a modified (e.g. equalised) or totally separate signal
may be used. See page 30.
SM Stage Manager.
sniff and break Semi-normalling. See page 48.
S/PDIF Acronym for "Sony/Philips Digital InterFace". [Also sometimes referred to by
its common "standards" title of IEC958 (type-2).]
The S/PDIF digital data format is very similar to the professional AES-EBU
standard although it uses different electrical characteristics. The system
normally carries 16 or 20-bit data, although it can accommodate 24-bits of
audio data per channel. Extra information can also be carried along side the
audio such as track start flags, source identification information, and timing
data.
The electrical interface is unbalanced and normally employs phono connectors.
The source impedance of 75 Ohms and high signal frequencies (0.1 to 6MHz)
require good quality 75-Ohm co-axial (RF) cable to operate reliably. Also, as
the source amplitude of the data signal is only 0.5V peak-to-peak this restricts
the transmission distance to short cable runs of up to about 10 metres.
An optical version of the interface is also available known as "TOSLink"
which transmits the same data signals as the electrical IEC958. This is achieved
with an LED transmitter and an opto-sensor as the receiver.
SPL Sound Pressure Level. See page 92.
SQ See Starquad.
Stage Box A box fitted with many audio connections on the end of a multicore cable. See
page 45.
Starquad A high quality microphone cable. See page 77.
Stereo Stereophonic. An audio system providing two channels of sound, hence giving
the listener spatial information in one dimension (usually “left / right”.) Most
modern recordings are in stereo.
Page 106 of 107

Surround Surround sound is a technique originating in cinema which generates additional
“rear” audio channels from a stereo signal. Some recordings are specially
encoded to enhance this process.
Suspension Microphone suspension.
TD Technical Director.
THD Threshold distortion. A fixed amount of distortion which occurs when an audio
signal crosses through zero volts, which is more noticeable on smaller signals.
ToC Table of Contents. The part of a MiniDisc which stores information about what
has been recorded on the disc including text labels, and where.
T-power A method of sending power down a microphone cable. See page 54.
Transient A unique pulse, often caused by mains interference. See page 51.
Tremolo A fast undulation in the amplitude of a sound.
UHF Ultra High Frequency: the part of the radio spectrum between 0.3 and 3 GHz.
Some radio mics operate in this band; see page 14.
Unbalanced A sound connection consisting of a single signal conductor, where the signal is
returned along the screen conductor. Compare with balanced.
Upstage The rear of the stage.
URL Uniform Resource Locator. The “address” of information on the Internet. See
page 58.
US See Upstage.
V see Volt.
VF Voice Frequency
VHF Very High Frequency: the part of the radio spectrum between 30 and 300 MHz.
Many radio mics operate in this band; see page 14.
Vibrato A fast undulation in the pitch of a sound.
Vinyl Any form of phonograph recording designed to be played on a turntable.
Volt A unit which measures the “pressure” of electricity.
VU Volume Unit. See page 93.
W see Watt.
Watt A unit which measures power.
Wet Signal A signal which has been processed by an effects unit. See page 25.
White noise A random signal with an energy distribution that produces the same amount of
noise power per Hz.
Windshield A close-fitting foam cover which protects the head of a microphone from wind,
thereby reducing whistling noises. Larger rifle mics are sometimes fitted with
windshields constructed from a fleece material, which gives protection against
stringer winds. See page 12.
XLR A balanced audio connector. See page 74.
Page 107 of 107

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The Big Book of Sound:

ADC Sound Design

© Andrew Love, Simon Parry, Matt Johnson and Rob Loxley

Cover Illustration: The Sound Box, ADC Theatre, Cambridge

Name Telephone E-mail

ADC General Information

Access to ADC Sound Facilities

Sound System Power

Sound Box Stage Level Amp Rack

Supply to Sound Box equipment

Incoming Mains Supply

Figure 1 - Block diagram of sound power system

It’s all obvious really - just use some common sense...

ADC Sound Equipment

Mics: 1 x Shure SM48

Mic stands: 4 x Mic stands

Sources: Denon MiniDisc DN-990R wired remote control in sound box.

Mixer: Soundcraft SeriesTWO 24 channel desk

Effects: Alesis M-EQ230 graphic equaliser

Amplifiers: 1 x C-Audio RA3001 (625W / channel)

Speakers: 6 x Bose 802 (240W) 6 channels Bose EQ available

Similar details for other venues in Cambridge are given in Appendix A

Microphones and where to put them

Omnidirectional Cardioid Hypercardioid Lemniscate/

Figure 2 - Microphone Polar Patterns

Microphones can also be divided into two categories according to construction:

Figure 3 - A Shure SM58 Figure 4 - An AKG D190

Windshields and Popshields

“Type” Coating Bias Advantages / Disadvantages

Digital Audio Tape

The following points are worth noting:

Open Reel Tape

Leader Colour Use

Use the following sequence to cue a record:

PPL and PRS

Figure 6 - The ADC processing & effects units

The Reverb Unit

The Multi-effects Unit

• Effects 1–9, 12–16 and

• Effects 10, 11 and 17–

• Effects 25–32 are split

The Graphic Equalisers

Uses for this equipment include:

Behringer Ultra-Curve Pro DSP8024P Basics:

Gate Compressor Peak Limiter

Figure 7 - Dynamic processing in a typical recording setup

Input Signal Level Control Output Signal

Sidechain Input Level Detector

Figure 8 - Main signal and sidechain signal paths in a dynamic processor

The ADC Sound Desk (Mixer)

Figure 9 - The ADC sound desk

There are three main parts to any mixer:

Figure 10 illustrates how these parts connect together:

Input channels Output channels

Input level Output level

1-24 Mono Two Stereo Four Stereo Return

line in socket L jack in socket L jack in socket

Figure 11 - Input channel facilities on the ADC sound desk

Mute System Operation

“If it sounds right to the audience, the mix is right.”

ADC Sound Output

Stage and Band Foldback

The Perfect Cue Sheet