SCHOOL OF AUDIO ENGINEERING RT202C-5 LIVE SOUND

DIPLOMA IN AUDIO ENGINEERING

RA 202(C)
ADVANCED STUDIO
STUDIES

RT202C-5 LIVE SOUND

NOTES

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Live Sound Reinforcement | Public Address Systems

THE PURPOSE OF A SOUND SYSTEM

The purposes of having a sound reinforcement system to begin with are manifold. In almost any indoor
auditorium with over two-hundred seats, sound amplification is usually needed-- for a variety of reasons.
A main reason is that the auditorium is simply too large for one to hear someone speaking on stage.
Maybe the acoustical characteristics of even a small auditorium make it impossible to hear someone on
stage. Maybe there wasn't enough money to design a decent auditorium so a hodge-podge sound system
was added as an afterthought.
Sound amplification came about when people finally realized that they couldn't, in fact, hear what the
guy at the front of the room was saying. First came the antiquated amplified lectern, many of which are
still in use today, with bad sound quality from a cheap-quality speaker hooked up to a cheap-quality
amplifier hooked up to a cheap-quality microphone with a cheap-quality orator who frequently did not
know how to use the microphone. Therefore sound reinforcement can be defined as the amplification of
an acoustic and/or electric performance (musical, vocal or theatrical). And the purpose can be summed
up as providing maximum coverage (audibility) in a given location.
Perhaps at the same time came the need, in such multipurpose auditoria, to hear music, such as that
which was recorded onto vinyl LP's or analog tape. Maybe bands played on stage and found their own
amplification equipment insufficient to fill the space completely. Someone finally designed a
multipurpose sound system that was integrated into the hall itself. However, in the construction /
renovation of such auditoria, sound was given a relatively low priority and quality suffered. The first
systems were pitifully bad in-house versions of the amplified lectern. Many had in-wall amplifiers
backstage with maybe six microphone inputs that would possibly be suitable for public address, but not
for high volume or clear sound reproductions. Sound systems can be classified into installed or tour
sound examples of installed sound can be churches, theatres, clubs, schools, public utility services, etc.,
any place where the equipment is installed or fixed on a long-term basis and live shows, concerts in
arenas or auditoria can be seen a tour sound- wherein the equipment is installed on temporary basis.
Some basic designs of common sound systems encountered are: *monaural sound-- usually the norm;
stereo would make the system more expensive (and more complicated). *the design would be relatively
simple, simple enough that any fool (well, almost any) could operate it, but still have some room for
improvement. These sound systems were by no means top-of-the-line, but they often reproduced orators
sufficiently. Then there are bigger systems, such as those used currently on Broadway for musicals or
large concert reinforcement systems for popular rock bands. We will focus on these sorts of systems. We
will also touch upon some other types of systems.

BAND/DANCE CLUB SOUND REINFORCEMENT

The vague principle behind such systems is that you want to induce a great state of musical euphoria in
the audience so that they'll remember how well the band or Dj played or spun. You do not want to draw
attention to the sound system, but it must be loud and clear. Mostly clear. This does not mean "deafen all
the audience members." It means "induce a great state of musical euphoria”. In nightclubs, one works
closely with the lighting designer to design an integrated system that will send the fearless club-goer into
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oblivion. But in terms of the sound system, clarity is a key. Bass is also a key. Clarity is achieved by
using good equipment located in good positions. The addition of several equalizers may further optimize
the sound. Bass is achieved by using many subwoofers located in different areas. Juliana's Tokyo, a
nightclub formerly in... you guessed it... Tokyo, had a dance floor that vibrates to the music. That's just
cool.
Bad sound systems seem to be those that tend to emphasize the midrange frequencies. Very high
frequencies and very low frequencies seem to be the most essential to dance music and that things in the
middle aren't very important. [a dance-music-history note: House music (and all its different forms) was
pioneered by many DJs, one of whom was Frankie Knuckles. "Knuckles was a sound sculptor, and he
liked to pump up the bass end of disco records while dropping the midrange in the mix. He understood
the DJ's power over the emotions of a crowd, and many rave DJs, particularly those preferring fast,
hardcore techno, still work with Knuckles' idea of the dance floor as thrill ride..." (Rolling Stone, April
7, 1994, p. 19)].
Band reinforcement should basically go under the principle that everything should be heard, and
everything should sound good together. One must have a good ear for listening to what is coming out--
if it sounds bad to you; it probably sounds bad to the audience. For instance, sometimes too much
distortion on guitars sounds bad. If it's loud enough, it'll distort anyway. A major important lesson to
teach band members is to keep their stage volume low. Doing this will ensure that the band will be clear
to each other (not distorted), and it can even help ensure that the band will not get hearing damage. The
whole point of having a sound engineer at a club is to amplify the band. It's baffling when a band comes
in with an entire six-foot-high guitar-amp stack and the bassist comes in with five bass cabinets. Why?
The purpose of the sound guy in a club is to put the band through the house system, not to just amplify
vocals or something. It's weird. They don't need a six-foot high guitar-amp stack... Why? The clearer the
band is on stage, the more the sound engineer can manipulate the levels and get a precise, clear sound
out to the audience. When the sound engineer needs to start competing with the band for levels, sound
quality will inevitably deteriorate.
THEATRE SOUND REINFORCEMENT
The object of a sound system for the theatre is to make everything on stage be heard clearly and
intelligibly, but not letting the audience know that a sound system is being used. The best theatre sound
systems are the ones you don't notice.

SIGNAL FLOW
A basic signal flow might help at this point to understand the various components and the manner in
which they are patched.
Mics/direct injection (D.I) -> mic lines -> junction/stage box -> mutlicore/snake -> console/desk -> effects
units (FX) -> graphic EQ -> peak limiter -> crossover -> power amplifiers -> loudspeakers

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Sound Reinforcement
Now that you've learned about the specific components of the sound system, we'll now put them all
together and make, yes, you guessed it, a sound system.

SYSTEM DESIGN SUGGESTIONS

The type of production and type of venue will dictate what kind of system will need to be utilized. Early
consultation with the director and constant communication with him/her will ensure that designer and
director are thinking along the same lines [i.e. natural sound? amplified, rock-show sound? no sound?].
It is also wise to notify the people in charge of the money early in the game that the sound budget will
probably have to be pretty large. No matter how sturdily the sets are built, and no matter how well the
actors are lit, if the audience can't hear the performers, then they've wasted their money. Equipment is
expensive-- often more expensive than lighting equipment or set construction materials; the technology
is also ever-changing, which also adds to the cost. Factors to consider in the overall system design are:
microphones-- type and use, console selection-- size and type, speaker selection-- type and placement,
and outboard effects-- type and use.
In a large venue, for instance, there will probably be a need for several speaker clusters located in
different parts of the venue, each with a separate program. A center cluster above the stage, for instance,
may need to be used only for vocal material, while the house left and house right clusters may need to be
used only for band material (and maybe some additional vocals). If there is a balcony, under-balcony fill
speakers should be used, with a mix of program materials. Surround-sound and special effects speakers
will need to be dealt with, too. This ultimately leads to the type of mixer one will need to use-- a simple
two-channel mixer won't do in such a situation. Hopefully, an eight-subgroup and eight-matrix console
will suffice. In this way, eight separate outputs can be constructed.
Recommended strategies for the system design are a center vocal cluster, which are high-performance,
high-efficiency, processor-controlled speakers dedicated to reproducing vocal program material. Two
side clusters of speakers located closer to the stage floor can be used for vocal fills (to bring the vocal
image down) and for band reinforcement. Under-balcony delays are a necessity if there is a balcony.
Additional speakers, such as surround-sound speakers and location special-effects speakers, will need to
be considered.
Effects processors, equalization, and compression will all need to be added into the design. Delay lines
may also need to be used, for the under-balcony fills and even maybe for the center cluster.

Microphone Selection
The type of production and the type of venue will dictate what needs to be miked, and when. For
theatrical musicals, most venues of over 200 seats will require some sort of reinforcement. Consult with
the director early to find out his or her ideas regarding microphone use. It is probable that many cast
members will require the use of wireless lavalier microphones. Sometimes, the director will also
advocate the use of handheld wireless microphones for other members of the cast to pass around during
large chorus scenes. Hey, it can happen. Then there's stage dialogue, backstage dialogue, and band
miking.

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EQUIPMENT SELECTION AND USE

Choice of main console and outboard effects will of course depend on the budget, but will also depend
on the director and the type of show. Add the number of mics that are to be used in the show, and allow
for some extras to be thrown in.
Equalization on main output channels is a must. Reverb processors mixed through an aux send on the
console will add body to wimpy voices and, in some cases, will bring back the mood of the show.
Compressors can be used on the subgroups through the inserts, but they are not necessary as long as the
board operator is trained well and knows what he/she's doing. Back in the days before compressors,
people used their ears and quick fingers on the faders. If used, compression should be applied to the
vocal subgroups. If a bass guitar is DI-ed into the board, a dedicated compressor should be applied to
this channel to enable worry-free pick up of the transient picks.
Remember also that as with most sound equipment, what you get is what you pay for. A cheap equalizer
will add more overall system noise and may distort quickly at high volumes. Avoid these. The industry
standard equalizer is the Klark-Teknik DN-360, a dual-channel 1/3 octave equalization unit that is also
quite expensive, but also very clean. Start here, and work down. Reverb units act in much the same way,
but luckily they are less expensive. The Alesis MicroVerb series is quite noisy, but the MidiVerb and
QuadraVerb units are quite dependable, and very quiet. For real sound shaping, check out the Sony
DPM series of sound processors. Digital delay processors will need to be used for any under-balcony fill
speakers to align the sound coming from the speakers with the sound coming from the front of the stage.
The Klark-Teknik DN-716 processor is a wonderful unit for this, although it's getting ousted by the BSS
TCS-804 (a lovely delay unit that measures in seconds, feet, meters, or inches); other units such as those
manufactured by TC Electronics, are similarly well-suited. Try different types of processing, and fiddle
with their parameters.
SPEAKER SELECTION AND USE
All the top-of-the-line equipment in the world won't sound good if one does not select proper equipment
to actually listen to it on. Thus, speaker selection is very important. Well, actually, all equipment
selection is very important, but speaker selection is very important.
Speaker selection should depend on the type of production and also the type of program material one is
planning to use them for. In the past, the standard for music program reinforcement was the Altec
"Voice of the Theatre" A-7 cabinet. One can probably still find them in old movie theatres and even in
stage theatres. Old '70's rock concerts used A-7s. These are very nice cabinets. They are also behind the
times.
"Processor-Controlled" systems have taken the lead in critical reinforcement situations. Processor-
Controlled systems involve speakers and a specially-designed-for-a-specific-speaker processor unit,
which has all sorts of neat equalization and phase circuitry inside of it that will make its corresponding
speakers perform the best (they also include overload-protection circuitry to make sure you don't
damage the drivers). Apogee, Meyer, Bose, and Electro-Voice all manufacture processor-controlled
cabinets. Apogee, Meyer, and E/V to an extent, have taken the lead in theatrical reinforcement cabinets.
The two industry-standard vocal processor-cabinets are the Apogee AE-5 and the Meyer UPA-1C, and
Electro-Voice's Delta-Max 1122, to a smaller degree. They all look very similar. They are wedge-
shaped, which allows clusters of them to be flown or otherwise placed. These are very clean-sounding
cabinets with excellent high- and mid-frequency response. Their low-frequency response is a bit smaller
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since they employ twelve- to fifteen-inch woofers; to compensate they are designed to be used in
conjunction with specially-tuned proprietary subwoofer systems. The AE-5s and UPAs are excellent for
vocal reinforcement. Apogee, Meyer, and Bose all manufacture smaller processor-controlled systems--
these can be used for under-balcony fills, surround, or for reinforcement/playback in smaller venues.
These same speakers can be used, in conjunction with their corresponding subwoofer units, for music
reinforcement. Using processor systems is ideal, but not necessarily economically viable for small-
budget amateur shows. Usually, full-range non-processor-controlled cabinets will suit music
reinforcement fine. Cabinets made by JBL, EAW, or E/V should work very well, although they may
sound a little less "clean" than a processor cabinet. Look for a very wide frequency response (especially
in the low-end if you are doing a rock musical), and also look for high efficiency ratings.
For surround-sound or under-balcony fills, check out small monitors made by Yamaha, JBL, or Bose.
Many small studio monitors will work well as reinforcement speakers, just be careful not to overload
them with excessive program material. Another feature to look for is mounting options or rigging points.
Speaker cabinets with no rigging points will not take well to having holes drilled in their cabinets. This
is very very dangerous. As a last resort, build strong frames out of metal to hold them, and attach rigging
points to these. Remember the safety ratio of 5:1-- if a speaker weighs ten pounds, the rigging materials
should be rated for at least fifty pounds.
For strong low-frequency response, for dance-clubs or loud music reinforcement/playback, check out
subwoofers. For a theatrical effect, place subwoofers under seating platforms or even in the plenum
below. For more information, check out THX Systems or Dolby Surround Systems or Sony Digital
Surround Systems used in movie theatres. Some dance-clubs install subwoofers underneath the dance
floor. Some even have systems that vibrate the dance floor in time to the music.
Proper location of speakers is also key in the whole design. A vocal cluster should not be located twenty
feet upstage and slightly to the right. Make your decisions quickly on speaker placement and let the
lighting and sets people know them early in the game. [You don't want to have to let the sets people
move your speakers two linesets upstage an hour before the house opens and find that they have broken
the Speakon connector on one of the main cables. It can happen.] The center vocal cluster, if there is
one, should be located, as the name implies, in the center. The idea is to cover the audience as equally as
possible-- moving the vocal cluster to the left or right without compensating equally on the opposite side
will make for some interesting reflections in the house and will most likely not cover the house equally.
The house left and house right stacks should be hung, or stacked, equally in the vertical plane.
Otherwise, holes in frequency response may occur in certain parts of the house. Check out the dispersion
angle characteristics of each speaker and align them according to that... or simply listen to them in
different parts of the house and align them that way. If a surround effect is your goal, place the speakers
according to that goal. Try to balance the house between left and right-- don't have lopsided design for
reinforcement.

THE HUMAN VOICE

Human begins generate sound by means of their vocal chords which intercept air exhaled from the lungs
and vibrate to produce a note. The sound must then pass through the pharynx, the mouth and the nose
and it is the way in which these cavities are used to shape the harmonics that determines the tone of the
sound we hear and enables us to differentiate one voice from another.

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In the case of performers it is important that they understand these facts and the necessity of voice-
training. Adequate breath control helps sustain the ends of words, where the vital consonants are often
situated. If these are lost then clarity suffers. Singers' voices often strain to produce adequate loudness at
the extreme high and low frequencies; guttural or strident sounds result. Sadly today fewer and fewer
performers are capable of clear and strong voice projection, perhaps as a result of time spent in film and
television work. As a consequence, more and more reliability is placed on amplifying stage productions.
In musicals where amplification is expected, the sound system cannot do all the work; the performer
must understand correct microphone technique-- too little voice and the microphone is likely to pick up
unwanted sounds or generate feedback in its straining to catch any level; distortion produced by working
too loudly into the microphone can only be contained by expensive equipment.
When selected loudspeakers and operating equalization, it is important to understand the range of
frequencies produced by the human voice. The usual fundamental frequency for males is about 125Hz,
and about 210Hz for females. This would be a normal voice level, but the trained voices such as those of
actors and actresses have higher values of about 140Hz and 230Hz, respectively.
The frequency range of trained voices is unsurprisingly wider than that of untrained voices and males
usually have a wider range than females-- although female voices are purer, having fewer harmonics.
The pitch of the voice is usually raised in emotional moments and when working in chorus.
Fundamentals usually have a frequency range of 125-250Hz. Vowels, usually 350-2000Hz, and
consonants 1500-4000Hz. Fundamentals of singers are: bass-- 85-340Hz, baritone-- 90-380Hz, tenor--
125-460Hz, alto-- 130-680Hz, contralto-- 180-600Hz, soprano-- 225-1100Hz.

Consoles
A primary concern in sound reinforcement is maximizing a system's acoustic gain. To do so, we must:

• a] keep the distance between the mic and loudspeaker as large as is practical;
• b] keep the distance between the mic and the source as small as is practical; and
• c] use directional mics and loudspeakers, placed so that their interaction is minimized.

First remember to mute or bring down the fader on any unused mics-- every time you double the number
of open mics, the total system gain must be reduced by 3 dB. Mic placement plays a large role in quality
sound reinforcement. The proper use of a microphone should be imparted upon the performers. For
vocal soloists, the mic should be as close as is possible to the speaker's mouth. They of course shouldn't
swallow the mic, but there should be less than two inches between lips and microphone. Large groups
should get as many microphones as is practical and the mics should be placed as close to each performer
as is practical. If the number of performers is a multiple of the number of microphones you have, split
up the mics evenly; i.e. if you have eight performers and four mics, place two people on each mic.
Microphone gain is also a very important part of the sound system. The more gain you add at the mixing
board, the higher chance of feedback. A good area to be in is a state known as unity gain.

GAIN CONTROL
Okay. Get a microphone, preferably the one you are going to use for the production. Plug it in. Turn the
"Gain" controls all the way down. These are the knobs generally located at the very top of each channel.
All the way to the left is usually "all the way down." Now bring the fader for that channel almost to the
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top of its travel. There is a line on the fader scale that says "0" or sometimes there is a bold line. Bring
the fader to this mark.
Bring the master faders (that go to the house) to the same bold line or "0" Now speak loudly and clearly
(preferably: have someone speak loudly and clearly...) into the mic whilst you adjust the gain control
knob. When it gets to a suitable level (your ears should be the judge here), leave it if possible to ensure
against preamp overload, system noise, etc. This formal procedure ensures that you have taken as much
power from the microphone as possible.
With the gain lower, you would have had to boost the channel and master faders, increasing system
noise. If you had set the gain higher, you risk overloading the preamplifier and distorting, an increase in
the chances for feedback, or both. This is a formal procedure that really should be done for all
microphones.
For line-level inputs, such as tape decks, the gain control should be set to the least possible setting (all
the way to the left) to begin with and should probably be left there.
INPUT PADDING
Suppose, as you went through the above steps to maximize gain, you set the gain all the way down (to
the left) but yet you were still peaking, distorting, etc. Maybe the sound got quieter but still sounded
fuzzy. What do we do now?
On some mixers (not all), there is a switch labeled "Pad" or "Mic Attenuator" or something to that
effect. It's usually above the gain control. Use this switch. If there is no switch, you can buy inline pads,
which are placed between the microphone output and the mixer input.
You switch in the pad last, when you know that the electronic adjustments alone will not lower the
signal from the microphone to a level that the preamp will handle.
Why adjust the pad last? Because a pad is not going to help your signal-to-noise ratio. It discards signal
coming in on the mic cable before it enters the preamplifier. One of the most common errors leading to
poor signal-to-noise performance is keeping the pad or mic attenuator engaged. That means that
additional amplification must be used to make up for the discarded signal, and with amplification comes
more noise.
EQUALIZATION
In sound reinforcement, equalization plays a considerable part. There are two sections of equalization
that may be found in a large sound system. The first stage of equalization is on the console itself-- the
channel eq. This eq should be tailored to each individual microphone. The second stage of equalization
usually occurs after the console and before the amplifiers-- to equalize the sound system to better match
the room's acoustics. There are a couple of ways to do this. The first way is to get a pink-noise
generator, and patch it in to the system. Pink noise is electronically generated noise that has equal
energy per octave. It essentially has the same amount of energy per frequency in relation to what the
human ear can hear. As this noise is playing through the system, a real-time analyzer is used. A
specially-calibrated test microphone is plugged into the real-time analyzer and the analyzer will display
the levels of each frequency (usually every 1/3 octave) from a given location, dictated by where one puts
it. This display will give a graphical representation of what frequencies are accentuated in the house. By

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taking several analyzers, or at least the data from several different locations, the sound engineer can
boost or cut the corresponding frequencies on the equalizers. When the analyzer displays a "flat," or
relatively flat house curve, the system is equalized to compensate for speaker imperfections and house
imperfections, yielding, ostensibly, a flat frequency response.
There is another way to do this, which doesn't require the use of real-time analyzers (which can be very
expensive). This other principle is known as ringing out the system.

RINGING OUT THE SYSTEM

The process is generally done for foldback monitors on stage in large, big, super-loud, professional
sound systems, but it also applies to house systems to some degree.
The principle behind ringing out a sound system lies in the fact that feedback is caused simply by
frequencies that are accentuated by room acoustics and are fed round and round the sound system. So, if
there are many accentuated frequencies, and we de-accentuate them, we'll have a relatively even
frequency response and be less prone to feedback.
What one does is to first turn on the EQ unit and set a perfectly flat response curve on the eq. Next, open
the mics in question (if wireless mics are a potential feedback threat, use them). Turn up the master
faders until the system starts to feed back. Using the EQ, figure out what frequencies are feeding back,
and cut those frequencies. Do this for all offending frequencies. At some point, you will be able to boost
the overall master gain of the mics. [If the EQ has a "In/Out" switch, try switching it and see just how
much feedback you've eliminated, and just how much more system gain you've obtained.]
From here we can now taylor the system to sound like what the sound designer wants it to sound like.
Remember: a perfectly flat frequency response on paper (or analyzer) is nice and all, but it can also
sound like absolute crap. Use your ears. Using a CD of pre-recorded music that the designer is familiar
with, we can boost frequencies that we want accentuated. Try to use a variety of music, but concentrate
on those types which represent the type of program material that the system will be used for. One can
also use the actual recording of the show in question, but this may become a bit tedious after one hears
the show fifteen times in succession during rehearsals and techs. Remember, though, not to overdo it.
Save really specific eq-ing for the individual mics.
[n.b. there is a third way of eq-ing a system, utilizing a PC-based computer system and various test-
microphones located around the house (Meyer's SIM System or Apogee's Correqt System). The
computer will automatically scan all the microphones and find offending frequencies and will control
the equalizers themselves utilizing a data network. If you can afford it, go with it. Behringer Specialized
Studio Equipment, Ltd., also manufactures a totally digital 31-band 2-channel EQ/RTA. It's really cool,
and not terribly expensive. It has a function on it called "Search and Destroy," which automatically will
cut resonant frequencies.
Now is the time when we would equalize each wireless mic or stage mic to the sound of the respective
performer's voice. Characteristics such as mic placement, mic type, performer location, and performer
type will play into this equalization. Try to make the microphone sound natural and well-balanced.

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PANNING AND STEREO IMAGE

In many theatre reinforcement systems, this is a null issue. Vocal clusters are usually mono, band stacks
are also usually mono. But it may be nice sometimes to pan reverb returns to hard left and hard right.
And electronic instruments, such as keyboards, samplers, and tone generators usually have stereo
outputs, so the band stacks theoretically can be wired in stereo. The only warning is to avoid panning
any instrument (that is not stereo) hard left or hard right. This should be pretty obvious.
SUBGROUPS AND MATRICES
The output section on your console will be your long-time friend. Large sound reinforcement consoles,
such as the Soundcraft K3, will include a number of subgroups, which, essentially, are submaster faders
to which individual channels or groups of channels can be assigned to. For example, if the engineer
wished to have group control over, say, all the wireless mics, which hypothetically take up channels 1-
10, by assigning channels 1-10 to groups 1 and 2, he would have a submaster fader for all the wireless
mics. Typically, reinforcement consoles have four or eight subgroups. Some consoles' channel controls
will have individual controls for routing to groups 1-8 and L-R; others will have only five controls for
panning between L-R, 1-2, 3-4, 5-6, and 7-8. Subgroups are also called groups or busses.
Some large sound reinforcement consoles will probably have a matrix section in addition to the
subgroups. While subgroups give control over the input section of the console (assigning channels, etc.),
matrices give control over the output section. In any theatre- or concert-sound reinforcement situation, it
is not enough to have only two outputs-- the left and the right. Often there may be many different
speaker sections that all need different program material. This is where the matrix section comes into
play. If there are eight subgroups on the console, eight knobs will be laid out per matrix, controlling the
level of each subgroup going to a particular matrix. In this fashion, a complex sound system design can
be facilitated. For example, if there are four subgroups-- vocals, band left, band right, and Bob (we'll use
four for simplicity's sake), and four matrices, whose outputs are driving speakers house left, house right,
vocal center, and Tokyo, and we want Bob to come out of the vocal cluster, we'll adjust the group four
level control on the third matrix. If we want band left to come out of Tokyo and a little bit out of house
right, we'll adjust the group two level control on the fourth matrix and a little bit on the second matrix.
Eventually, when all these levels are set, we'll adjust the matrix master level, which will drive the
speakers in the aforementioned locations.

To the uneducated, a sound reinforcement console looks very imposing. There are lots of faders, even
more knobs, and a countless amount of buttons and switches. However, a good number of faders, knobs,
and buttons do exactly the same thing. Every channel on the console will look almost identical to the
next, and once one is mastered, so are the rest. Going from the top (furthest away from the operator) of
the console, there will usually be the following: a phantom power switch (+48), an input pad switch, a
gain control, high-frequency eq, high-mid sweepable eq, mid-low sweepable eq, low eq, an eq on/off
switch, many aux sends, some with pre/post/off switches, a pan control, a pan control on/off switch, a
channel mute (or channel on/off) switch, a PFL switch, the fader, and group assignment switches. Many
of these controls have been explained before or are pretty self-explanatory. The PFL (pre-fader-listen)
switch, by the way, allows the sound engineer to monitor the channel in question by depressing the
switch and listening through a pair of headphones. In this way microphones can be checked, tapes can
be cued, and actors can be spied upon, without having to bring up the fader in question.

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The output section may include eq for each subgroup, an on/off switch for each subgroup, an AFL
(after-fader-listen) switch to listen to the individual group; the matrix section usually is more sparse--
with only a matrix master control and on/off switch per matrix.
The left-right faders are pretty sparse, too; often located above them are the masters for the aux sends
(on/off, too), oscillator control (for level-setting when mixing to tape), and talkback controls. A mic or
mic input on the face of the console allows the sound operator to talk to various portions of the sound
system. A "Slate" button allows him/her to talk to the groups-- handy for recording, when the engineer
will call out measure numbers or sections (akin to the Director's Slate used in film production), a
"Talkback" button allows the operator to talk through the aux sends (sometimes they are assignable)
during sound checks or on-stage paging, and a "Comm" button is designed to be used with a proprietary
intercom system. Each talkback section includes a master level control, too.

Signal Processors
DELAY UNITS AND APPLICATIONS
Delay units can play a very significant and important role in sound reinforcement systems-- especially
for theatrical sound systems. Remember that the goal of the theatrical reinforcement system is to not be
heard. Judicious use of delay will aid in not being heard. Concentrating on vocal clusters, this is the
theory behind using a delay unit: In 1947, a man named Helmut Haas performed some tests into delayed
sound and how we hear it. Haas found that if a human was listening to two sound sources of the same
material, and one was slightly delayed with respect to the other and also slightly louder, the ear would
associate the direction of the sound source to the sound it heard first, not the sound it heard loudest.
However, this only works within a window of 10dB. If the second, delayed speaker is 10dB louder than
the first, non-delayed speaker, the effect starts to wane. This is known as the Haas Effect, and is the
basis for such image-positioning systems as AKG's Deltastereophony. Thus, the goal is to arrange the
system so that the delayed, amplified sound arrives after the direct sound, thereby focusing the
audience's attention on the real source.
By inserting a delay line before the amplifier to a vocal cluster, we can try to achieve what Helmut Haas
proved. By delaying the cluster from 10ms to 30ms, we can shift the audience's perception of the sound
source to the stage itself, not to the vocal cluster. This effect is limited by the sound coming from the
stage; often if actors are miked well and sound natural while facing front and suddenly turn around to
face the back wall (upstage), a sudden shift in perception is noted as the original sound (their voice) is
not being heard clearly. Another application for the delay unit is in venues with balconies. Underneath
the balcony, intelligbility from the center vocal cluster will diminish, since there is a distinct shadow
caused by the balcony overhead. Thus, insertion of small speakers with good high- to mid-frequency
response is necessary to improve intelligbility. However, without the use of a delay line, sound will
come from these small underbalcony speakers first-- which will confuse if not irritate the audience.
Insertion of a delay line will fix this problem-- by delaying the underbalcony fill speakers, sound will
seem to "appear" from the stage first, and the fill speakers will provide for improve intelligibility.

Foldback monitors are the wedge-shaped boxes one sees lying at the feet of singers during rock
concerts. They provide a mix of whatever the singer in question wants/needs to hear in order to keep in
time and in tune. Similarly, band members use them to hear singers/other band members so they can
keep in time and in tune. Use them for your shows. If the orchestra is located in a pit of sorts
(metaphysically or physically), the singers will inevitably have a difficult time hearing them without the
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aid of additional monitors. Sound will come out of the pit (and out of the band stacks in the house),
bounce around for a while, and eventually will return to the performers on stage. This is far from ideal.
Thus, a few foldback monitors (which, if they're located in the wings or off of electrics, need not be
wedge-shaped) will improve the overall quality of your production. Performers need to hear the band. A
simple mix created from band-mic aux sends on your console will provide a sufficient mix for the
performers. EQ and compress as needed. Band members need to hear the performers (whether they
really listen to them or not is another story). A simple mix created from stage-mic aux sends (post-
fader!) will suffice. EQ and compress as needed. If the band is located directly in the house itself, with
no real pit, sometimes the vocal cluster located above the stage will provide sufficient reinforcement for
the band, thereby eliminating the need for additional monitors.
Show relays are exactly what their name implies. A microphone located in the house (or, if you are
ambitious, a separate submix from the console) is wired through paging amplifiers which, in turn, are
wired to small ceiling- or wall-mounted speakers backstage, in different tech booths, and, occasionally,
in the foyer. Ofttimes these will be pretty much set-up, wiring-wise. A simple "turning on" motion will
usually be sufficient.

Amplifiers
CROSSOVER NETWORKS

Loudspeaker crossovers are a necessary evil. A different universe, a different set of physics and maybe
we could have what we want: one loudspeaker that does it all. One speaker that reproduces all audio
frequencies equally well, with no distortion, at loudness levels adequate for whatever venue we play.
Well, we live here, and our system of physics does not allow such extravagance. The hard truth is, no
one loudspeaker can do it all. We need at least two -- more if we can afford them. Woofers and tweeters.
A big woofer for the lows and a little tweeter for the highs. This is known as a 2-way system. (Check the
accompanying diagrams for the following discussions.) But with two speakers, the correct frequencies
must be routed (or crossed over) to each loudspeaker.

Passive
At the simplest level a crossover is a passive network. A passive network is one not needing a power
supply to operate -- if it has a line cord, or runs off batteries, then it is not a passive circuit. The simplest
passive crossover network consists of only two components: a capacitor connecting to the high
frequency driver and an inductor (aka a coil) connecting to the low frequency driver. A capacitor is an
electronic component that passes high frequencies (the passband) and blocks low frequencies (the
stopband); an inductor does just the opposite: it passes low frequencies and blocks high frequencies. But
as the frequency changes, neither component reacts suddenly. They do it gradually; they slowly start to
pass (or stop passing) their respective frequencies. The rate at which this occurs is called the crossover
slope. It is measured in dB per octave, or shortened to dB/octave. The slope increases or decreases so
many dB/octave. At the simplest level, each component gives you a 6 dB/octave slope (a physical fact of
our universe). Again, at the simplest level, adding more components increases the slope in 6 dB
increments, creating slopes of 12 dB/oct, 18 dB/oct, 24 dB/oct, and so on. The number of components,
or 6 dB slope increments, is called the crossover order. Therefore, a 4th-order crossover has (at least)
four components, and produces steep slopes of 24 dB/octave. The steeper the better for most drivers,
since speakers only perform well for a certain band of frequencies; beyond that they misbehave,
sometimes badly. Steep slopes prevent these frequencies from getting to the driver.
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You can combine capacitors and inductors to create a third path that eliminates the highest highs and the
lowest lows, and forms a mid-frequency crossover section. This is naturally called a 3-way system. (See
diagram) The "mid" section forms a bandpass filter, since it only passes a specific frequency band. Note
from the diagram that the high frequency passband and low frequency passband terms are often
shortened to just high-pass and low-pass. A 3-way system allows optimizing each driver for a narrower
band of frequencies, producing a better overall sound.
So why not just use passive boxes?

Problems

The single biggest problem is that one passive cabinet (or a pair) won't play loud enough and clean
enough for large spaces. If the sound system is for your bedroom or garage, passive systems would work
just fine -- maybe even better. But it isn't. Once you try to fill a relatively large space with equally loud
sound you start to understand the problems. And it doesn't take stadiums, just normal size clubs. It is
really difficult to produce the required loudness with passive boxes. Life would be a lot easier if you
could just jack everyone into their own cans amp -- like a bunch of Mojo Series MH 4 Headphone Amps
scattered throughout the audience. Let them do the work; then everyone could hear equally well, and
choose their own listening level.

Monitor speakers on the other hand most likely have passive crossovers. Again, it's a matter of distance
and loudness. Monitors are usually close and not overly loud -- too loud and they will feed back into
your microphone or be heard along with the main mix -- not good. Monitor speakers are similar to hi-fi
speakers, where passive designs dominate ... because of the relatively small listening areas. It is quite
easy to fill small listening rooms with pristine sounds even at ear-splitting levels. But move those same
speakers into your local club and they will sound thin, dull and lifeless. Not only will they not play loud
enough, but they may need the sonic benefits of sound bouncing off close walls to reinforce and fill the
direct sound. In large venues, these walls are way too far away to benefit anyone.

Passive 2-Way Crossover Passive 3-Way Crossover

So why not use a bunch of passive boxes? You can, and some people do. However, for reasons to
follow, it only works for a couple of cabinets. Even so, you won't be able to get the high loudness levels
if the room is large. Passive systems can only be optimized so much.
Once you start needing multiple cabinets, active crossovers become necessary. To get good coverage of
like frequencies, you want to stack like-drivers. This prevents using passive boxes since each one
contains (at least) a high-frequency driver and a low-frequency driver. It's easiest to put together a sound
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system when each cabinet covers only one frequency range. For instance, for a nice sounding 3-way
system, you would have low-frequency boxes (the big ones), then medium-sized mid-frequency boxes
and finally the smaller high-frequency boxes. These would be stacked — or hung, or both — in some
sort of an array. A loudspeaker array is the optimum stacking shape for each set of cabinets to give the
best combined coverage and overall sound. You've no doubt seen many different array shapes. There are
tall towers, high walls, and all sorts of polyhedrons and arcs. The only efficient way to do this is with
active crossovers.

Some smaller systems combine active and passive boxes. Even within a single cabinet it is common to
find an active crossover used to separate the low- and mid-frequency drives, while a built-in passive
network is used for the high-frequency driver. This is particularly common for super tweeters operating
over the last audio octave. At the other end, an active crossover often is used to add a subwoofer to a
passive 2-way system. All combinations are used, but each time a passive crossover shows up, it comes
with problems. One of these is power loss. Passive networks waste valuable power. The extra power
needed to make the drivers louder, instead boils off the components and comes out of the box as heat —
not sound. Therefore, passive units make you buy a bigger amp.
A couple of additional passive network problems has to do with their impedance. Impedance restricts
power transfer; it's like resistance, only frequency sensitive. In order for the passive network to work
exactly right, the source impedance (the amplifier's output plus the wiring impedance) must be as close
to zero as possible and not frequency-dependent, and the load impedance (the loudspeaker's
characteristics) must be fixed and not frequency-dependent (sorry — not in this universe; only on Star
Trek). Since these things are not possible, the passive network must be (at best), a simplified and
compromised solution to a very complex problem. Consequently, the crossover's behavior changes with
frequency — not something you want for a good sounding system.

One last thing to make matters worse. There is something called back-emf (back-electromotive force:
literally, back-voltage) which further contributes to poor sounding speaker systems. This is the
phenomena where, after the signal stops, the speaker cone continues moving, causing the voice coil to
move through the magnetic field (now acting like a microphone), creating a new voltage that tries to
drive the cable back to the amplifier's output! If the speaker is allowed to do this, the cone flops around
like a dying fish. It does not sound good! The only way to stop back-emf is to make the loudspeaker
"see" a dead short, i.e., zero ohms looking backward, or as close to it as possible — something that's not
gonna happen with a passive network slung between it and the power amp.

All this, and not to mention that inductors saturate at high signal levels causing distortion — another
reason you can't get enough loudness. Or the additional weight and bulk caused by the large inductors
required for good low frequency response. Or that it is almost impossible to get high-quality steep slopes
passively, so the response suffers. Or that inductors are way too good at picking up local radio, TV,
emergency, and cellular broadcasts, and joyfully mixing them into your audio.
Such is life with passive speaker systems.

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Active 2-Way Crossover

Active 3-Way Crossover

Active

Active crossover networks require a power supply to operate and usually come packaged in single-space,
rack-mount units. (Although of late, powered loudspeakers with built-in active crossovers and power
amplifiers are becoming increasingly popular.) Looking at the accompanying diagram shows how active
crossovers differ from their passive cousins. For a 2-way system instead of one power amp, you now
have two, but they can be smaller for the same loudness level. How much smaller depends on the
sensitivity rating of the drivers (more on this later). Likewise a 3-way system requires three power amps.
You also see and hear the terms bi-amped, and tri-amped applied to 2- and 3-way systems.
Active crossovers cure many ills of the passive systems. Since the crossover filters themselves are safely
tucked away inside their own box, away from the driving and loading impedance problems plaguing
passive units, they can be made to operate in an almost mathematically perfect manner. Extremely steep,
smooth and well-behaved crossover slopes are easily achieved by active circuitry.
There are no amplifier power loss problems, since active circuits operate from their own low voltage
power supplies. And with the inefficiencies of the passive network removed, the power amps more
easily achieve the loudness levels required. Loudspeaker jitters and tremors caused by inadequately
damped back-emf all but disappear once the passive network is removed. What remains is the
amplifier's inherent output impedance and that of the connecting wire. Here's where the term damping
factor comes up. [Note that the word is damp-ing, not damp-ning as is so often heard; impress your
friends.] Damping is a measure of a system's ability to control the motion of the loudspeaker cone after
the signal disappears. No more dying fish.

Siegfried & Russ

Active crossovers go by many names. First, they are either 2-way or 3-way (or even 4-way and 5-way).
Then there is the slope rate and order: 24 dB/oct (4th-order), or 18 dB/oct (3rd-order), and so on. And
finally there is a name for the kind of design. The two most common being Linkwitz-Riley and
Butterworth, named after Siegfried Linkwitz and Russ Riley who first proposed this application, and
Stanley Butterworth who first described the response in 1930. Up until the mid `80s, the 3rd-order (18
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dB/oct) Butterworth design dominated, but still had some problems. Since then, the development
(pioneered by Rane and Sundholm) of the 4th-order (24 dB/oct) Linkwitz-Riley design solved these
problems, and today is the norm.
What this adds up to is active crossovers are the rule. Luckily, the hardest thing about an active
crossover is getting the money to buy one. After that, most of the work is already done for you. At the
most basic level all you really need from an active crossover are two things: to let you set the correct
crossover point, and to let you balance driver levels. That's all. The first is done by consulting the
loudspeaker manufacturer's data sheet, and dialing it in on the front panel. (That's assuming a complete
factory-made 2-way loudspeaker cabinent, for example. If the box is homemade, then both drivers must
be carefully selected so they have the same crossover frequency, otherwise a severe response problem
can result.) Balancing levels is necessary because high frequency drivers are more efficient than low
frequency drivers. This means that if you put the same amount of power into each driver, one will sound
louder than the other. The one that is the most efficient plays louder. Several methods to balance drivers
are always outlined in any good owner's manual.

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Operation of A Live Sound System

Setting Sound System Level Controls

Importance

Correctly setting a sound system's gain structure is one of the most important contributors to creating an
excellent sounding system. Conversely, an improperly set gain structure is one of the leading
contributors to bad sounding systems. The cost of the system is secondary to proper setup. The most
expensive system set wrong never performs up to the level of a correctly set inexpensive system. Setting
all the various level controls is not difficult; however, it remains a very misunderstood topic.
The key to setting level controls lies in the simple understanding of what you are trying to do. A few
minutes spent in mastering this concept makes most set-ups intuitive. A little common sense goes a long
way in gain setting.
A dozen possible procedures exist for correctly setting the gain structure of any system. What follows is
but one of these, and is meant to demonstrate the principles involved. Once you master the fundamental
principles, you will know what to do when confronted with different system configurations.

Decibels, Dynamic Range & Maximizing Headroom

Audio-speak is full of jargon, but none so pervasive as the decibel. Mastering gain, or level control
settings also requires an understanding of dynamic range and headroom. Dynamic range is the ratio of
the loudest (undistorted) signal to that of the quietest (discernible) signal in a piece of equipment or a
complete system, expressed in decibels (dB). For signal processing equipment, the maximum output
signal is ultimately restricted by the size of the power supplies, i.e., it cannot swing more voltage than is
available. While the minimum output signal is determined by the noise floor of the unit, i.e., it cannot
put out a discernible signal smaller than the noise (generally speaking). Professional-grade analog signal
processing equipment can output maximum levels of +26 dBu, with the best noise floors being down
around -94 dBu. This gives a maximum unit dynamic range of 120 dB — a pretty impressive number
coinciding nicely with the 120 dB dynamic range of normal human hearing (from just audible to
painfully loud).

For sound systems, the maximum loudness level is what is achievable before acoustic feedback, or
system squeal begins while the minimum level is determined by the overall background noise. It is
significant that the audio equipment noise is usually swamped by the HVAC (heating, ventilating & air
conditioning) plus audience noise. Typical minimum noise levels are 35-45 dB SPL (sound pressure
level), with typical loudest sounds being in the 100-105 dB SPL area. (Sounds louder than this start
being very uncomfortable, causing audience complaints.) This yields a typical useable system dynamic
range on the order of only 55-70 dB -- quite different than unit dynamic ranges.
Note that the dynamic range of the system is largely out of your hands. The lower limit is set by the
HVAC and audience noise, while the upper end is determined by the comfort level of the audience. As
seen above, this useable dynamic range only averages about 65 dB. Anything more doesn't hurt, but it
doesn't help either.

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Headroom is the ratio of the largest undistorted signal possible through a unit or system, to that of the
average signal level. For example, if the average level is +4 dBu and the largest level is +26 dBu, then
there is 22 dB of headroom. Since you cannot do anything about the system dynamic range, your job
actually becomes easier. All you need worry about is maximizing unit headroom. Fine, but how much is
enough? An examination of all audio signals reveals music as being the most dynamic (big surprise)
with a crest factor of 4-10. Crest factor is the term used to represent the ratio of the peak (crest) value to
the rms (root mean square -- think average) value of a waveform. For example, a sine wave has a crest
factor of 1.4 (or 3 dB), since the peak value equals 1.414 times the rms value.
Music's wide crest factor of 4-10 translates into 12-20 dB. This means that musical peaks occur 12-20
dB higher than the "average" value. This is why headroom is so important. You need 12-20 dB of
headroom in each unit to avoid clipping.

Preset All Level Controls in the System

After all equipment is hooked-up, verify system operation by sending an audio signal through it. Do this
first before trying to set any gain/level controls. This is to make sure all wiring has been done correctly,
that there are no bad cables, and that there is no audible hum or buzz being picked up by improperly
grounded interconnections. Once you are sure the system is operating quietly and correctly, then you are
ready to proceed.
• Turn down all power amplifier level/sensitivity controls.
• Turn off all power amplifiers. (This allows you to set the maximum signal level through the
system without making yourself and others stark raving mad.)
• Position all gain/level controls to their off or minimum settings.
• Defeat all dynamic controllers such as compressors/limiters, gate/expanders, and enhancers by
setting the Ratio controls to 1:1, and/or turning the Threshold controls way up (or down for
gate/expanders).
• Leave all equalization until after correctly setting the gain structure.

Console/Mic Preamp Gain Settings

A detailed discussion of how to run a mixing console lies outside the range of this note, but a few
observations are relevant. Think about the typical mixer signal path. At its most basic, each input
channel consists of a mic stage, some EQ, routing assign switches and level controls, along with a
channel master fader. All of these input channels are then mixed together to form various outputs, each
with its own level control or fader. To set the proper mixer gain structure, you want to maximize the
overall S/N (signal-to-noise) ratio. Now think about that a little: because of the physics behind analog
electronics, each stage contributes noise as the signal travels through it. Therefore each stage works to
degrade the overall signal-to-noise ratio. Here's the important part: The amount of noise contributed by
each stage is (relatively) independent of the signal level passing through it. So, the bigger the input
signal, the better the output S/N ratio (in general). The rule here is to take as much gain as necessary to
bring the signal up to the desired average level, say, +4 dBu, as soon as possible. If you need 60 dB of
gain to bring up a mic input, you don't want to do it with 20 dB here, and 20 dB there, and 20 dB some
other place! You want to do it all at once at the input mic stage. For most applications, the entire system
S/N (more or less) gets fixed at the mic stage. Therefore set it for as much gain as possible without
excessive clipping. Note the wording excessive clipping. A little clipping is not audible in the overall
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scheme of things. Test the source for its expected maximum input level. This means, one at a time,
having the singers sing, and the players play, as loud as they expect to sing/play during the performance.
Or, if the source is recorded, or off-the-air, turn it up as loud as ever expected. Set the input mic gain
trim so the mic OL (overload) light just occasionally flickers. This is as much gain as can be taken with
this stage. Any more and it will clip all the time; any less and you are hurting your best possible S/N.
Once you've set all the input gains, and then created the overall desired mix (involving all sorts of art
and science I'm not going to get into), then you must set the output level controls in a similar manner:
advance the output control until the output OL light begins to flicker. This is the maximum output level.
(Note that a simple single mic preamp is set up in the same manner as a whole mixing console.)

Setting Outboard Gear Input/Output Level Controls

All outboard unit level controls (except active crossovers: see below) exist primarily for two reasons:
• They provide the flexibility to operate with all signal sizes. If the input signal is too small, a gain
control brings it up to the desired average level, and if the signal is too large, an attenuator
reduces it back to the desired average.
• Level controls for equalizers: the need to provide make-up gain in the case where significant
cutting of the signal makes it too small, or the opposite case, where a lot of boosting makes the
overall signal too large, requiring attenuation.

Many outboard units operate at "unity gain," and do not have any level controls -- what comes in
(magnitude-wise) is what comes out. For a perfect system, all outboard gear would operate in a unity
gain fashion. It is the main console's (or preamp's) job to add whatever gain is required to all input
signals. After that, all outboard compressors, limiters, equalizers, enhancers, effects, or what-have-you
need not provide gain beyond that required to offset the amplification or attenuation the box provides.
With that said, you can now move ahead with setting whatever level controls do exist in the system.
Whether the system contains one piece of outboard gear, or a dozen, gains are all set the same way.
Again, the rule is to maximize the S/N through each piece of equipment, thereby maximizing the S/N of
the whole system. And that means setting things such that your maximum system signal goes straight
through every box without clipping. Here's how:
Choose between one of the three methods OL Light, Oscilloscope, or AC Voltmeter described below.
With the console or preamp set up as above, you now need a convenient sound source. Use an oscillator
(built-in or external) and feed in a tone around 1 kHz. Or you can substitute pink noise for the "OL
Light" or "Oscilloscope" methods, but pink noise will not work for the "AC Voltmeter" method (the
ACVM will not respond fast enough to catch the peaks).

• OL Light Set the level of the oscillator (NOT the console or preamp's output level; it has
already been set!) by turning up its own control, if existing, or by using a spare channel on the
console. Turn it up such that the output OL indicator just begins to light. It's a large signal, on the
order of +20 dBu, and would be very loud if you had not already turned off the amps. What you
have now is the maximum expected signal level running through the system. From here on,
everything will be set so it does not clip with this signal. Once done, the operator can run the
system as loud as they want without fear of (feedback or) distortion.

• Oscilloscope Using the OL light is a fast and convenient way to set this level. However, a better
alternative is to use an oscilloscope and actually measure the output to see where excessive
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clipping really begins. This method gets around the many different ways that OL points are
detected and displayed by manufacturers. There is no standard for OL detection. If you want the
absolute largest signal possible before real clipping, you must use an oscilloscope. And, of
course, if the unit or console does not have an OL indicator, then an oscilloscope is mandatory to
establish the actually clipping point.

• AC Voltmeter If an oscilloscope is out of the question, another alternative is to use an AC

voltmeter (preferably with a "dB" scale). Here, instead of relying on the OL indicator to tell you
when you have a maximum signal, you choose a very large output level, say, +20 dBu (7.75
Vrms) and define that as your maximum level. Now set everything to not clip at this level. This is
a reasonable and accurate way to do it, but is it an appropriate maximum? Well, you already
know (from the above discussion) that you need 12-20 dB of headroom above your average
signal. It is normal pro audio practice to set your average level at +4 dBu (which, incidentally,
registers as "0 dB" on a true VU meter). And since all high quality pro audio equipment can
handle +20 dBu in and out, then this value becomes a safe maximum level for setting gains,
giving you 16 dB of headroom -- plenty for most systems.

Outboard gear falls into three categories regarding gain/level controls:

• No controls
• One control, either Input or Output
• Both Input & Output Controls

Obviously, the first category is not a problem!

If there is only one level control, regardless of its location, set it to give you the maximum output level
either by observing the OL light, or the oscilloscope, or by setting an output level of +20 dBu as shown
on your AC voltmeter. With two controls it is very important to set the Input control first. Do this by
turning up the Output control just enough to observe the signal. Set the Input control to barely light the
OL indicator, then back it down a hair, or set it just below clipping using your oscilloscope. Now set the
Output control also to just light the OL indicator, or just at clipping using the scope. (Note: there is no
good way to optimally set an input control on a unit with two level controls, using only an AC
voltmeter.)

Setting Power Amplifiers

If your system uses active crossovers, for the moment, set all the crossover output level controls to
maximum. Much confusion surrounds power amplifier controls. First, let's establish that power amplifier
"level/volume/gain" controls are input sensitivity controls. (no matter how they are calibrated.) They are
not power controls. They have absolutely nothing to do with output power. They are sensitivity controls,
i.e., these controls determine exactly what input level will cause the amplifier to produce full power. Or,
if you prefer, they determine just how sensitive the amplifier is. For example, they might be set such that
an input level of +4 dBu causes full power, or such that an input level of +20 dBu causes full power, or
whatever-input-level-your-system-may-require, causes full power. They do NOT change the available
output power. They only change the required input level to produce full output power. Clearly
understanding the above makes setting these controls elementary. You want the maximum system signal
to cause full power; therefore set the amplifier controls to give full power with your maximum input
signal using the following procedure:

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1. Turn the sensitivity controls all the way down (least sensitive; fully CCW; off).
2. Make sure the device driving the amp is delivering max (unclipped) signal.
3. Warn everyone you are about to make a LOT of noise!
4. Cover your ears and turn on the first power amplifier.
5. Slowly rotate the control until clipping just begins. Stop! This is the maximum possible power output
using the maximum system input signal. In general, if there is never a bigger input signal, this setting
guarantees the amplifier cannot clip. (Note: if this much power causes the loudspeaker to "bottom out,"
or distort in any manner, then you have a mismatch between your amplifier and your loudspeaker.
Matching loudspeakers and amplifiers is another subject beyond this note.)
6. Repeat the above process for each power amplifier.
7. Turn the test signal off.

ACTIVE CROSSOVER OUTPUT LEVEL CONTROLS

Setting the output attenuators on active crossovers differs from other outboard gear in that they serve a
different purpose. These attenuators allow setting different output levels to each driver to correct for
efficiency differences. This means that the same voltage applied to different drivers results in different
loudness levels. This is the loudspeaker sensitivity specification, usually stated as so many dB SPL at a
distance of one meter, when driven with one watt. Ergo, you want to set these controls for equal
maximum loudness in each driver section. Try this approach:
1. Turn down all the crossover outputs except for the lowest frequency band, typically labeled "Low-
Out." (Set one channel at a time for stereo systems.)
2. If available, use pink noise as a source for these settings; otherwise use a frequency tone that falls
mid-band for each section. Turn up the source until you verify the console is putting out the maximum
system signal level (somewhere around the console clipping point.) Using an SPL meter (Important:
turn off all weighting filters; the SPL meter must have a flat response mode) turn down this one output
level control until the maximum desired loudness level is reached, typically around 100-105 dB SPL.
Very loud, but not harmful. (1-2 hours is the Permissible Noise Exposure allowed by the U.S. Dept. of
Labor Noise Regulations for 100-105 dB SPL, A-weighted levels.)
Okay. You have established that with this maximum system signal this driver will not exceed your
desired maximum loudness level (at the location picked for measurement). Now, do the same for the
other output sections as follows:
1. Mute this output section -- do not turn down the level control; you just set it! If a Mute button is not
provided on the crossover, disconnect the cable going to the power amp.
2. Turn up the next output section: either "High-Out" for 2-way systems, or "Mid-Out" for 3-way
systems, until the same maximum loudness level is reached. Stop and mute this output.
3. Continue this procedure until all output level controls are set.

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4. Un-mute all sections, and turn off the test source.

Congratulations! You have finished correctly setting the gain structure for your system.
Now you are ready to adjust equalization and set all dynamic controllers. Remember, after EQ-ing to
always reset the EQ level controls for unity gain as required. Use the Bypass (or Engage) pushbuttons to
"A/B" between equalized and un-equalized sound, adjusting the overall level controls as required for
equal loudness in both positions.

Summary

Optimum performance requires correctly setting the gain structure of sound systems. It makes the
difference between excellent sounding systems and mediocre ones. The proper method begins by taking
all necessary gain in the console, or preamp. All outboard units operate with unity gain, and are set to
pass the maximum system signal without clipping. The power amplifier sensitivity controls are set for a
level appropriate to pass the maximum system signal without excessive clipping. Lastly, active
crossover output controls are set to correct for loudspeaker efficiency differences.

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