Audio Basics

Introduction

Audio is the department where the newcomer is most likely to be handed something and told to make it work. A microphone that is not working. A channel that hums. A wedge that squeals the moment somebody steps in front of it. And the honest truth is that almost all of those have a cause you can find, in an order you can learn, using equipment that behaves the same way in every room you will ever walk into.

That is what this book is for. Not the art of mixing, which is a different skill and a later course, but the equipment and the chain: what each box does, what it expects to receive, and what happens when it gets something else.

What this book actually does

It teaches the gear. What the three signal levels are and why plugging one into the other is the single most common audio fault on earth. What gain actually sets, and what headroom buys you. How a microphone turns air into electricity, which of the three kinds to reach for, and what a polar pattern decides. What a mixing console is doing behind that wall of identical strips. How a loudspeaker turns it all back into air, and why feedback is a loop rather than a mystery.

Every chapter shows you the thing on a real job, walks you through the procedure, warns you about the failure you will actually meet, and gives you something you can take to work.

How this sits with The Physics of Sound

This book stands on its own. Everything you need in order to do the work is in here, taught properly, and you do not need another book beside you to follow it. Chapter one carries the physics this book depends on rather than pointing somewhere else for it.

The Physics of Sound is a different book about the same air. It asks why: why a room sounds like that, why two loudspeakers are not twice as loud, why a wavelength decides what a wall does to it. Where the two overlap, they overlap on purpose — meeting an idea twice, once as a working fact and once as an explanation, is how it stops being something you looked up and starts being something you know.

Who this is for

Somebody who will be handed audio work in the next few months, or has already been handed it and is getting by on pattern-matching. You do not need to have mixed anything. You do need to be willing to look at a rack of unfamiliar boxes and work out what goes in and what comes out, which is a skill this book builds deliberately.

What this book will not do

It will not teach you to mix a band. That is Front-of-House Mixing, and it assumes what is in here. It will not teach you a particular console, because every manufacturer names things differently and a book that taught you one desk would be wrong in a year and useless on the next job. It teaches what all of them are doing, so the manual for any of them makes sense.

What it will do is take you from "audio is the scary department" to being genuinely useful on an audio crew, which is the step that actually gets you booked.

How to Use This Book

Each chapter follows the same shape, and the shape matters more than it might look. You read the ideas, then you see them play out in a real situation, then you get the procedure, then you find out what goes wrong, then you practice, then you get a checklist you can actually take to work.

1. Read the chapter body first. This is the understanding layer, and it is written to be read straight through.
2. Read On the job. The same material as a real situation on a real show, which is where it stops being abstract.
3. Work through How to do it. These are procedures you can follow on a job this week.
4. Study What goes wrong. Experienced technicians are not people who never make mistakes; they are people who recognise mistakes early.
5. Do the exercise before reading the answer key. Recognition feels like learning and is not.
6. Take the checklist with you. Photograph it on your phone. Use it until you no longer need it.

Keep a notebook

One habit is worth starting today, before the first chapter: keep a small notebook and write the technical details in it as you meet them. Audio is full of numbers you will need exactly and will not remember — what a line level actually is, how much pad a particular microphone needs on a loud source, which venue's wedge sends are numbered backwards, the phantom power switch that is on the back of the rack rather than the front.

Write it by hand rather than photographing this page. Copying a number puts it on paper; rewriting it in your own words is what puts it in your head, and deciding how to phrase it is most of the learning.

Then use it twice. Read a few pages on the way to a job, which takes five minutes and turns things you half-know into things you know. And carry it on the job, because the moment you need to know whether that input takes phantom power is never a moment when you can stop and look it up.

A note on the words

Audio vocabulary is inconsistent and you should expect that rather than be caught by it. Gain is trim is input level, depending on the desk. A bus is a group is a mix. A wedge is a floor monitor, and a monitor is also the person mixing them, and sometimes also a screen.

None of that is worth arguing about. What matters is that you can say what a thing does, because a person who can describe the function will be understood anywhere, and a person who only knows one desk's word for it will be understood in one building.

A note on safety

Audio can permanently damage the hearing you need to do the job, and it does it painlessly and cumulatively. Damage is a dose rather than a volume: a moderate level for a long day costs you as much as a loud one briefly. Carry plugs, wear them, and treat the people who do as the ones who intend to still be working in twenty years.

The rest is the usual: mains power near liquid, heavy cases, cable across walkways, things above your head. This book gives you awareness, not authorisation. If you are ever unsure whether you are qualified to do something, you are not — ask.

Where this book sits

This book assumes you have read AV Fundamentals 101, or that you already know what is in it.

Experience counts instead. If you have a few months on crews, where you have already learned what a signal is, what a source and a destination are, and roughly how a show goes together, start here.

If none of that is true yet, read AV Fundamentals 101 first.

What pairs with this

• The Physics of Sound — the why beneath the what — the same air, asked about differently

Where it leads

These are not steps in an order. Take whichever one fits.

• Front-of-House Mixing — to mix a band, which assumes everything in here
• A2 & Stage Audio Fundamentals — to work the stage end rather than the desk
• Wireless & RF Systems — because radio microphones are a discipline of their own

What Sound Actually Is

Every decision in the rest of this book rests on about a dozen facts about air. This chapter teaches them — not as theory to be admired, but as the handful of things you act on, often, at speed, and usually while somebody is waiting. None of it needs mathematics beyond doubling and halving.

Sound is moving air

Sound is a pattern of pressure travelling through air. Something moves, it squeezes the air next to it, that squeeze passes to the air next to that, and a moment later it arrives at an eardrum or a microphone diaphragm and pushes it.

The air itself barely goes anywhere. It is the pattern that crosses the room, the way a wave crosses water without the water travelling with it.

Three consequences matter on a job, and they are the reason this paragraph exists at all. Sound needs a medium, so it does not cross a vacuum and it travels differently through a wall than through air. It takes time to arrive — roughly three milliseconds per metre, which is why a distant loudspeaker sounds late. And the microphone is not hearing your source so much as being pushed by it, which is why what is between the two matters as much as either.

Frequency and pitch

How fast that pressure pattern repeats is its frequency, measured in hertz, and what you hear as pitch. Low numbers are low notes: a kick drum lives around 50 to 100 Hz. High numbers are high sounds: the air and detail on a cymbal runs past 10,000 Hz.

Human hearing runs roughly 20 Hz to 20,000 Hz when you are young, and the top end retreats with age and exposure. That is not a tragedy, it is a clock.

The number worth carrying is that speech intelligibility lives mostly between about 1,000 and 4,000 Hz. When somebody cannot be understood, that is the region to think about first — and it is why a system can be loud and useless at the same time.

Amplitude and loudness

How big the pressure swing is, is amplitude, and it is roughly what you hear as loudness. Bigger swing, louder sound.

Roughly, because your ears are not a measuring instrument. They are far less sensitive to low frequencies at quiet levels than at loud ones, which is why a mix that sounded full at rehearsal volume can sound thin when you turn it down for a speech, and why the answer is not always to add bass.

For the working technician the point is simply this: loudness is a perception, level is a measurement, and they do not track each other reliably. Trust the meters for what is happening and your ears for whether it is right.

Decibels, simply

Audio measures level in decibels, and the decibel is a ratio on a logarithmic scale rather than a unit like a metre. That sounds abstract and it has three entirely practical consequences.

Three decibels is double the power and is a change you can just hear. Six decibels is double the voltage. Ten decibels is roughly what most people call twice as loud, and it takes about ten times the power to get there. This is why the last few decibels of a system cost so much more than the first, and why the answer to "can we just make it louder" is usually no.

The other thing to know is that a decibel on its own means nothing, because it is a ratio — a ratio to what has to be stated. dBu and dBV describe electrical signals, dBFS describes how close a digital signal is to its ceiling, and dB SPL describes actual loudness in a room. Somebody saying "it is at minus twenty" is telling you nothing until you know which scale they mean, and asking is not a stupid question.

Sound has a size

A sound's frequency also gives it a physical length, and that length decides what the world can do to it. Divide the speed of sound — about 343 metres per second in ordinary air — by the frequency, and you get the wavelength.

A 1,000 Hz tone is about a third of a metre long, roughly a hand span. A 100 Hz tone is nearly three and a half metres. A 40 Hz tone, the bottom of a kick drum, is more than eight metres — longer than most rooms are tall.

This single fact explains a great deal that otherwise looks like superstition. Sound bends around anything smaller than its own wavelength and is blocked by anything larger, which is why a person standing in front of a loudspeaker takes the top end out of it but not the bottom, why a projection screen dulls the speech behind it while the bass passes through untouched, and why you can hear the bass from a room down the corridor but not the words.

It is also why low frequencies are the ones that leak into the neighbouring conference room, why a subwoofer can sit almost anywhere while a top box has to be aimed carefully, and why acoustic treatment for bass has to be thick — a thin panel is simply too small for an eight-metre wave to notice.

You will not calculate wavelengths on a job. You will constantly use the shape of the idea: big sounds go everywhere and are hard to stop, small sounds travel in straight lines and are easy to block.

What distance does to level

Sound spreads out as it travels, and the same energy covering a larger area means less of it arrives anywhere in particular. For a single source in the open, doubling the distance costs about six decibels.

That number is worth having in your hands, because it works in both directions and the useful direction is backwards. If a presenter is 200 mm from the microphone and you get her to 100 mm, you have gained six decibels for free — no gain, no extra loudspeaker, nothing to go wrong. If she drifts from 100 mm to 400 mm during her talk, she has quietly lost twelve, and the instinct to turn her up is how a room starts ringing.

This is the single most valuable thing you can do about almost any audio problem, and it is why an experienced technician's first move is so often to move something rather than to adjust something. Distance is free level, and it is the only free level there is.

Real rooms are kinder than this rule suggests, because reflected sound fills in behind the direct sound and the level stops falling away once you are far enough out. That is a comfort outdoors-versus-indoors, but it is not good news: the sound holding the level up at the back of the room is the reflected sound, and reflected sound is the enemy of understanding the words.

What a room does to it

Everything so far assumes sound going outwards and not coming back. Indoors, it comes back.

Sound meets a surface and some of it is absorbed, some is reflected, and a little passes through. Hard, heavy, flat surfaces — glass, plaster, a polished table — reflect nearly all of it. Soft, fibrous, deep ones — heavy curtain, carpet, an audience in coats — absorb it, and absorb the high frequencies far more readily than the low ones.

So a listener hears the direct sound first, then the same sound again, thousands of times over, arriving from every surface a few milliseconds later and progressively quieter. That pile-up of late arrivals is reverberation, and how long it takes to die away is what people mean by a live or a dead room.

A little of it is pleasant and makes music sound generous. Too much of it, and consonants are the casualty: the tail of one word is still arriving while the next word starts, and "fifteen" and "sixteen" become the same noise. This is precisely why a room can be unmistakably loud and still impossible to follow, and why turning it up makes it worse — you are amplifying the reflections in exact step with the thing you wanted to hear.

There are only three levers, and one of them is not yours. You can move the source closer to the microphone, you can aim loudspeakers at people rather than at walls, and somebody with a budget can add absorption. Level is not on the list.

The six numbers worth carrying

If you remember nothing else from this chapter, remember these. They are the numbers a working technician uses without stopping to think, and every one of them earns its place several times a week.

On the job

A conference room, one presenter, one lectern microphone, and a complaint that arrives before the coffee does: the people at the back cannot understand her.

The instinct is to turn it up, and it is wrong. She is already audible at the back — the complaint is not that they cannot hear her, it is that they cannot make out the words, and those are different problems with different fixes.

Audible is level. Intelligible lives between about one and four kilohertz, and it is being lost to two things: she is a foot further from the microphone than she was in rehearsal, and the room's own reflected sound is arriving a few milliseconds behind the direct sound and smearing the consonants.

So the fixes are, in order: get her closer to the microphone, which costs nothing and gains real level in the range that carries meaning; and if there is a delay speaker at the back, check it is actually delayed, because an undelayed one makes this worse rather than better.

Turning the system up would have made her louder and no clearer, which is the specific failure this chapter exists to prevent.

How to do it

Somebody cannot be understood. Work in this order — it goes cheapest and most effective first, and it almost never reaches step five.

1. Get the source closer to the microphone. Halving that distance is worth about six decibels and costs nothing.
2. Check the level is actually adequate at the back before assuming it is not. Walk there. This takes ninety seconds and settles the argument.
3. Ask whether the problem is loudness or clarity. They sound like one complaint and they have different fixes.
4. If it is clarity, think about the range that carries meaning — roughly one to four kilohertz — before touching anything else.
5. Check any distant loudspeaker is delayed. An undelayed one arrives before the stage sound and destroys intelligibility while adding level.
6. Only then consider more level, and expect it to help less than you hoped.

What goes wrong

Practice it

Five questions. All of them come up in the first month.

The Audio Signal & Gain Structure

If you learn one thing from this book, learn this chapter. Signal level is the most common audio fault in the world, it is entirely mechanical, and once you can say what each box sends and what the next one expects, a whole class of problems stops being mysterious and starts being arithmetic.

The three signal levels

Audio travels at three wildly different strengths, and confusing them is the single most common fault in the trade.

Mic level is tiny — thousandths of a volt. It is what comes out of a microphone, and it is so small that it needs amplifying before anything useful can be done with it, and so fragile that it picks up interference from anything nearby. This is why microphone cables are built the way they are and why they should never be coiled next to a mains lead.

Line level is about a thousand times larger, and it is the standard currency of audio equipment. Once a signal has been amplified to line level it is robust: it travels well, it is much harder to interfere with, and it is what almost every box on a rack expects to receive and to send. A laptop, a media player, a mixer output, an effects unit — all line level.

Speaker level is larger again, hundreds of times more than line, because it has to physically move a loudspeaker cone. It comes out of a power amplifier and goes into a passive loudspeaker, and nothing else on a stage sends or expects it.

Those three cover essentially everything you will meet. Instrument level, which comes out of an electric guitar, sits awkwardly between mic and line and is the reason DI boxes exist.

What matters is what happens when you get it wrong, because each mistake has its own signature and knowing the signature tells you the fault.

Plug a line-level source into a microphone input and the signal is a thousand times bigger than the input expects. It distorts immediately and horribly, and turning the gain down does not save it — the damage happens at the input stage before the gain control can help.

Plug a microphone into a line input and the opposite happens: the signal is a thousand times too small, so it is either inaudible or so faint that turning it up brings the noise floor up with it. What you hear is a thin, hissy, distant version of the source.

Send speaker level into anything that is not a passive loudspeaker and you will damage equipment. That one is not a sound problem; it is a repair bill.

Gain: the first knob in the chain

Gain is the first control on a channel and the most misunderstood. It is not volume. It sets how much the incoming signal is amplified to bring it up to the level the rest of the console expects to work with.

Everything downstream — the fader, the equaliser, the compressor, the sends — assumes the signal arriving is already at a sensible working level. Gain is what makes that true. Set it correctly and every other control behaves predictably. Set it badly and nothing downstream can rescue it, which is why gain is set first and then largely left alone.

The way it is set is by ear and eye together. You get the source doing what it will actually do during the show — a singer singing at performance volume, not talking politely — and you raise the gain until the meter is reading healthily with room to spare. Then you stop.

The mistake nearly everybody makes early is using gain as a volume control during the show. If a channel needs to be louder in the mix, that is what the fader is for. Reaching for gain changes the level going into everything downstream, which means the compressor now behaves differently, the monitor sends change, and the effects change — all at once, and none of it visible.

Headroom, noise floor, and clipping

Between the quietest thing a system can carry and the loudest, there is a window. Understanding that window is what gain structure is for.

At the bottom is the noise floor: the constant low-level hiss and hum every piece of electronics produces. It is always there. If your signal sits close to it, you have to turn everything up to hear the signal, and you bring the noise up with it.

At the top is clipping. Every stage has a ceiling, and a signal that tries to exceed it does not simply get louder — its peaks are squared off, which sounds like harsh, ugly distortion. In digital systems the ceiling is absolute and unforgiving; there is nothing above zero.

Headroom is the space you deliberately leave between where your signal normally sits and that ceiling. It exists because live sources are unpredictable: a singer leans in, a drummer hits harder in the chorus, a presenter suddenly gets enthusiastic. If you have set levels so that normal is nearly at the ceiling, the first genuinely loud moment of the show clips.

The practical target is to sit comfortably in the middle of that window — well above the noise, with real space above for the surprise. On a digital desk, peaking somewhere around minus eighteen to minus twelve with the loudest moments still short of zero is a sane place to live.

Cable management as signal protection

Cable is not just a way of getting a signal from A to B, and treating it as such is how avoidable faults are created.

Mic-level signals are tiny, so anything electrically noisy nearby can induce interference into them. A microphone cable run alongside a mains lead for twenty metres will often pick up hum. The fix costs nothing: cross power and audio at right angles rather than running them in parallel, and where they must run together, separate them physically.

Balanced cable is the other half of the answer, and it is why professional audio uses XLR. A balanced connection carries the signal twice, once inverted, and the receiving end subtracts one from the other. Interference picked up along the way arrives identically on both and cancels out in that subtraction. This is why a balanced line can run fifty metres and an unbalanced one starts humming after five.

Then there is the purely physical. A cable trodden on, run over by a case, or pulled by its connector will eventually fail — usually intermittently, usually during a show, and usually in a way that takes an hour to find. Coil it properly, do not walk on it, and disconnect it by the connector rather than by pulling the cable.

The signal levels, and what each mistake sounds like

The most useful table in this book. Every row is a fault you will meet, and the sound tells you the cause.

Where to sit in the window

Sane working targets, not laws. Every desk meters slightly differently and the manual beats this table.

On the job

A corporate breakfast. A lectern microphone, a laptop for the slides with sound on one video, and a small mixer feeding two speakers on sticks.

The microphone is fine. The laptop is not: the moment the video plays, the sound is a distorted mess, and turning the channel down does not help — it just gets quieter and still ugly.

That "quieter and still ugly" is the whole diagnosis. Distortion that survives turning it down happened before the volume control, which means it happened at the input. The laptop is a line-level source and it has been plugged into a microphone input, so a signal roughly a thousand times too big is being fed to a stage built for something tiny.

The fix is any one of three things depending on what is in the bag: use a line input if the mixer has one, engage the pad on that channel if it has one, or put a DI box in between. All three do the same job — bring the signal down to what the input expects.

What does not work, and what somebody will suggest, is turning the laptop down instead. It helps a little and it costs you the signal-to-noise on the way, so you end up with a quiet, hissy, slightly-less-distorted mess. Fix it at the input.

How to do it

Setting gain on a channel, in the order that makes every later control behave. Two minutes per channel and it is the two minutes that decide the show.

1. Check what the source actually sends before you plug it in. A laptop is line level, a microphone is mic level, a guitar needs a DI. Getting this right first prevents the whole class of fault.
2. Set the channel fader to its normal position and the master somewhere sensible, so you are setting gain into a chain that resembles the one you will use.
3. Get the source doing what it will do in the show. A singer singing, not talking; a presenter at presentation volume, not at conversation volume.
4. Bring the gain up until the meter reads healthily — around minus eighteen to minus twelve on a digital desk — with the peaks well short of the ceiling.
5. Ask for the loudest thing they will do, and check it still does not reach the top. This is the step people skip and it is the one the show tests.
6. Stop touching gain. From here the fader is the volume control, and everything downstream depends on the level you just set staying put.
7. If the channel needs a pad, engage it before setting gain rather than fighting the input with the gain control.

What goes wrong

Practice it

Six questions. Every one of them is something you will be asked to diagnose, usually while somebody waits.

Microphones

A microphone is the first thing in the chain and the one decision that cannot be undone later. No amount of skill downstream rescues a source captured badly, which is why choosing and placing well is worth more than most of what happens after it.

How a microphone works

A microphone turns moving air into a moving electrical signal, and every type does it the same way in outline: something extremely light — a diaphragm — is pushed back and forth by the pressure wave arriving at it, and that movement is converted into a voltage that copies its shape.

That is worth holding onto because it explains most microphone behaviour. The diaphragm is being physically pushed. It cannot tell the difference between the voice you want and the wedge pointing at it. It responds to whatever pressure reaches it, from wherever, and everything about choosing and placing a microphone is about controlling what reaches it.

It also explains why the signal is so small. Air pressure moving a tiny diaphragm produces thousandths of a volt, which is why microphones need a preamplifier before anything else can happen, and why their cables need protecting from interference.

The differences between types come down to how that movement becomes a voltage — and those differences decide how rugged the microphone is, how much detail it captures, and whether it needs power to work at all.

Dynamic, condenser, and ribbon

There are three kinds and, for a working technician, they divide by what job they are for rather than by how they are built.

A dynamic microphone moves a coil of wire in a magnetic field, which generates the voltage directly. It needs no power, it is mechanically simple, and it is extremely tough. It will survive being dropped, rained on, gripped by a singer and used as a prop. It is less sensitive to fine detail than the alternatives, and it handles very loud sources without complaint. This is why the standard vocal microphone on almost every stage in the world is a dynamic, and why it is what goes on a snare drum and a guitar cabinet.

A condenser microphone works electrostatically, with a very light diaphragm close to a fixed plate. That lightness makes it far more sensitive and much better at detail and high frequencies — but it needs power to work, and that power is phantom power, forty-eight volts sent up the microphone cable from the desk. Condensers are what you reach for when accuracy matters: overheads, acoustic instruments, choirs, lecterns, anything where you want to hear what is actually there.

A ribbon microphone suspends a thin metal ribbon in a magnetic field. It is prized for a particular smooth character, and it is delicate — older ribbons can be damaged by phantom power and by rough handling. You will meet them in studios far more often than on a stage, and the rule on a live job is to leave them to whoever brought them.

For most work the choice is a two-way one: dynamic when it will be loud, close, or handled roughly; condenser when detail matters and the source is not going to destroy it.

Polar patterns

A polar pattern is the shape of what a microphone listens to, and it is the single most useful thing to understand about microphone choice — on a live stage, on a film set, or on any job where a camera is running.

An omnidirectional microphone hears equally in every direction. That gives the most natural sound, and it makes it almost useless for a loud stage, because it hears the wedges and the drums as readily as the voice.

A cardioid pattern is heart-shaped: most sensitive in front, progressively less at the sides, and least directly behind. This is the workhorse of live sound, and its dead spot at the rear is not an accident — it is where you point the thing you do not want, which usually means a monitor wedge.

A supercardioid is tighter at the front, which rejects more from the sides, but it trades that for a small rear lobe: it hears a little of what is directly behind it again. That matters practically, because the dead spot is no longer straight back but off to either side, so a wedge placed dead behind it is now a feedback risk rather than a safe position.

A bidirectional pattern — almost always called a figure-of-eight, and sometimes bi-polar — hears equally in front and behind, and rejects the two sides more completely than any other pattern rejects anything. Those side nulls are the deepest you can buy, which makes it a precision instrument rather than a curiosity.

It earns its place in three jobs. Two people talking across a table share one microphone, each on a live lobe, with the room's sides rejected. It is the pattern every ribbon microphone has by construction, because a ribbon is open on both faces. And it is one half of mid-side recording, where a figure-of-eight pointed across the stage is combined with a forward-facing microphone to produce a stereo image whose width can be decided afterwards — a technique from broadcast and film that is now ordinary on any job capturing room sound for a video edit.

What to watch is the back. A figure-of-eight is as sensitive behind as in front, so whatever is behind it is in the shot acoustically even when it is not in the shot visually. It also has the strongest proximity effect of any pattern: move close and the bass rises steeply, which is a tool when you want warmth and a problem when you did not.

A shotgun is the long, thin microphone you have seen on a boom pole above an actor's head, and it is the workhorse of film and location sound. What makes it work is an interference tube: a slotted barrel in front of the capsule that lets sound arriving from the front pass straight through, while sound arriving from the sides enters at several slots at once, out of step with itself, and largely cancels. The result is a narrow forward lobe with strong side rejection — technically a lobar pattern, and like a supercardioid it keeps a rear lobe, so what is directly behind it is not safe.

The most important thing to understand about a shotgun is what it does not do. It does not reach further. Distance costs you the same six decibels per doubling that it costs everything else. What a shotgun buys is a better ratio: less of the room and less of the sides for the same amount of your subject. That is exactly why a boom operator still works the microphone as close to the actor as the frame allows rather than standing back and pointing.

Its weakness is rooms. The interference tube colours whatever it rejects rather than removing it, so off-axis sound comes back thin and harsh instead of merely quieter. Outdoors, where there is little off-axis sound, that hardly matters and a shotgun sounds superb. In a small hard room, where reflections arrive from every angle, a shotgun can sound worse than a plain cardioid — which is why interiors are often boomed with a hypercardioid instead, and why a shotgun clipped to a camera in a meeting room rarely sounds as good as its price suggests.

In corporate AV you will meet shotguns on camera mounts, on booms for interview setups, and hung or lectern-mounted where the microphone has to be far from the talker and out of the shot.

The reason to care about all of this is that the pattern is a free tool. Aiming a microphone's dead area at the loudest thing in the room costs nothing, needs no equipment, and buys more usable level before feedback than any equaliser will. On a live stage that loudest thing is a wedge; on a location shoot it is a road, a generator or an air-conditioning unit; and the move is identical.

Placement basics

Where a microphone goes matters more than which microphone it is, and the two rules that decide most of it are simple enough to apply without thinking.

The first is distance. Halving the distance between source and microphone gains you roughly six decibels of the sound you want, and costs you nothing in the sound you do not. That is a bigger improvement than most processing can deliver, and it is free. Close is almost always better on a live stage.

The second is the three-to-one rule, and it prevents a specific ugly problem. When two microphones pick up the same source at slightly different distances, their signals arrive slightly out of step and combine into a thin, hollow, phasey sound. The rule of thumb is to keep microphones at least three times further apart than each is from its own source.

A panel table is where this bites and where it is easy to satisfy. Four panellists, each with a microphone about 200 mm from their own mouth, means those microphones want to be at least 600 mm apart — which on a table of any normal size they comfortably are, provided nobody slides two of them together to make room for a water jug.

A lectern is the more instructive case, because the rule cannot be satisfied there at all. Two microphones on one lectern are perhaps 200 mm apart while each sits 200 mm from the same mouth: a ratio of one to one, when you wanted three to one. No placement fixes it, because the lectern is not wide enough to contain the answer. That is precisely why a lectern with two microphones is almost always a redundancy arrangement with only one of them open, and why opening both — which looks like it should give you more level — gives you a thinner sound instead.

Then there are the ordinary practicalities that decide whether a good placement survives the day. Point the dead side at the wedge. Keep the microphone out of the way of the performer's hands, music stand and sightline. Secure the cable so a foot cannot take the stand over. And check what the microphone can see besides the source — a cardioid pointed at a singer with a drum kit directly behind is a cardioid pointed at a drum kit.

Which microphone, and why

Choose by the job rather than by the specification. On a live stage this is nearly always a two-way decision.

Every polar pattern, and where it is deaf

The pattern is a free tool. Point the dead area at the loudest thing in the room and you have gained level before feedback for nothing.

On the job

A panel discussion in a hotel ballroom. Four speakers at a table, four microphones, and two wedges so the panel can hear questions from the floor during the Q&A — they hear each other perfectly well across a table a metre wide. The room starts ringing the moment all four microphones are open.

The first instinct is the equaliser. It is the wrong instinct, and the right one costs nothing.

Look at what the microphones can see. They are cardioids, so each has a dead spot directly behind it — and the wedges are in front of the table, pointing at the panel, which means pointing straight into the most sensitive part of every microphone. The pattern is being used backwards.

It is worth seeing how that happened, because it was not carelessness. Somebody aimed the wedges at the people who need to hear them, which is the right instinct and the wrong geometry: the panel's ears and the panel's microphones are facing the same way, so anything aimed at one is aimed at the other.

Turning the table around is not an option, so the fix is to move the wedges: out to the sides and angled in, so each microphone's rear null faces roughly where the wedge is. That alone buys back several decibels before anything rings.

Then the second free fix: four open microphones cost about six decibels of gain before feedback compared with one. On a panel that means riding them — closing whichever ones nobody is speaking into — rather than leaving all four open all night.

Only after both of those is it worth finding the one frequency that still rings and taking a narrow cut. If it needs more than a couple of those, something in the geometry is still wrong.

How to do it

Choosing and placing a microphone on a source. Two minutes, and it decides more about the result than anything you do afterwards.

1. Look at the source and ask how loud it is and how close you can get. Loud and close means a dynamic; detailed and controllable means a condenser.
2. Check whether the microphone needs phantom power, and whether that input can supply it. A condenser with no phantom is a dead channel that looks like a broken cable.
3. Get it as close as the performance allows. Halving the distance is worth about six decibels of the sound you want, free.
4. Work out where the pattern's dead area is, and point it at the loudest thing in the room — usually a wedge.
5. If two microphones will pick up the same source, keep them at least three times further from each other than each is from the source.
6. Look at what else the microphone can see. A cardioid aimed at a singer with a drum kit behind them is also aimed at the drum kit.
7. Secure the stand and dress the cable so a foot cannot take it over, then check the performer can still reach their music, their instrument and their water.

What goes wrong

Practice it

Six questions. All of them are decisions you will be asked to make with somebody waiting.

The Mixing Console & Signal Path

A console is the object people find most intimidating and the one that is most obviously the same thing repeated. It is one channel strip, printed forty times, feeding a handful of destinations. Learn the strip and the wall of knobs stops being a wall.

What the console does

A mixing console does three things, and everything on its surface serves one of them.

It takes many inputs and gets each one to a usable level and shape — that is the channel strip. It combines them in chosen proportions — that is the mix. And it sends different combinations to different places — that is the bus structure, and it is the part beginners notice last and need most.

That third one is worth dwelling on, because it is what makes a console more than a volume-control panel. The audience needs one mix. The performers on stage need a different one, usually several different ones, because a singer wants to hear themselves and the drummer wants to hear the bass. A recording may need another. A broadcast feed another again. The console's job is to build all of those simultaneously from the same set of inputs.

Everything else — the equaliser, the compressor, the effects — is refinement layered on top of those three functions. A console with none of it would still be a console.

The channel strip, top to bottom

One input's journey through the desk, in the order the signal actually travels. Every console lays this out the same way for the same reason: it is the order in which the decisions have to be made.

Gain comes first, at the top, setting the incoming signal to a workable level. Everything below assumes it has been done. On a digital desk it may be labelled trim.

Phantom power, usually a small switch near the gain, sends forty-eight volts up the cable for a condenser microphone. It does nothing for a dynamic and nothing good for some ribbons.

A pad, where fitted, drops the incoming signal by a fixed amount — usually twenty decibels — before the preamplifier, for sources too loud or too strong for the input.

A high-pass filter cuts everything below a set frequency. On a vocal channel this is nearly always worth engaging: it removes stage rumble, handling noise and footfall that contribute nothing but eat headroom.

The equaliser shapes the tone, cutting or boosting bands of frequency. On most desks this is three or four bands with the middle ones sweepable.

Auxiliary sends take a copy of the channel to somewhere else — a monitor wedge, an effects unit. This is where the performers' mixes are built, and the crucial part is whether each send is pre-fade or post-fade, which decides whether it follows the main fader or ignores it.

Then routing, deciding which mix this channel joins, and finally the fader, which sets how much of it goes there. Plus a mute, a pan, and usually a solo for listening to that channel alone.

Buses, mains, and monitors

A bus is a destination that several channels can be sent to at once. That is the whole concept, and the names on a desk are all instances of it.

The main bus — often called the left-right or the mains — is the mix the audience hears. Most channels go there, at whatever level the fader sets.

An auxiliary bus is a separate mix built from the same channels in different proportions. Its normal use is a monitor wedge: the singer's wedge is an aux bus with lots of the singer, some of the guitar and almost none of the drums, built independently of what the audience hears.

Whether a send is pre-fade or post-fade is the detail that decides whether this works. A pre-fade send takes its signal before the channel fader, so it is unaffected by what you do to the main mix — which is exactly what a monitor needs, because pulling a channel down in the room should not silence it on stage. A post-fade send takes it after, so it follows the fader — which is what an effect wants, because if a channel is out of the mix its reverb should go with it.

Getting those two backwards is one of the more memorable mistakes in live sound. Monitors on post-fade means every change you make for the audience is also made, unasked, in the performers' ears.

Groups are a third kind: a way of sending several related channels to one fader — all the drums, say — so they can be controlled together before joining the main mix.

Analog and digital consoles

Both do the same job, and the difference is not sound quality so much as what happens between shows.

An analog console gives you one physical control per function. Everything is visible at once, nothing is hidden in a menu, and what you see is what is set. That immediacy is why many engineers still prefer them for simple work. The cost is size, weight, no memory, and the fact that a channel's outboard equipment is separate hardware.

A digital console puts the same functions behind a screen and a smaller number of controls that change what they do depending on what is selected. That is the one thing to be careful about — the knob you are holding does something different for each channel — but in exchange you get things that change the job: the whole desk recalls in seconds, every channel has processing that would have been a rack of hardware, and multiple shows live in one box.

That phrase — processing that would have been a rack of hardware — is the whole comparison, and it is worth spelling out rather than leaving as an aside. To make an analog console do what a modest digital console does out of the box, you have to buy, carry, power and connect the missing pieces.

A compressor for each channel that needs one, so the loud singer does not have to be ridden by hand. An equaliser for each monitor send, which on a live stage is where most feedback is fought. An effects unit for reverb and delay, because a voice with no reverb in a big room sounds like an announcement. On a digital console every one of those is already in every channel and every output, included, recalled with the show.

And this is the part that decides how your evening goes: every one of those boxes has to be patched in. A compressor is not switched on, it is wired into the channel's insert point — out of the console, into the compressor, back into the console. That is two more cables per channel. An effects unit is fed from an aux send and returned to a pair of input channels, so it costs a send, two returns and four connectors. A rack of eight processors is thirty or so patch leads that all have to be right, and a patch bay exists precisely because doing that by reaching round the back does not scale.

Every one of those leads is somewhere a signal can be lost, and none of it is remembered. At the end of the night the settings live on the hardware and in your memory, and tomorrow's show starts by dialling them again — which is why analog engineers photograph their desks.

None of that makes analog worse. A four-microphone panel needs none of it, and an analog desk will be set up and working while the digital one is still booting. But it does mean the honest question is not "which console", it is which system — and a digital desk of the same channel count is usually smaller, lighter, faster to load in, and cheaper than the same capability assembled out of separate boxes.

For a beginner the meaningful advice is short. On analog, look at what is in front of you, and follow every insert lead with your hands before you doubt the desk. On digital, always check what is selected before you turn anything, because the fastest way to ruin a mix is to adjust the right control on the wrong channel.

The stage box, and where the microphone really plugs in

There is a second box you will meet on your first digital job, and it surprises people because nobody introduces it. The console is at the mix position, out in the room. The microphones are on stage. On an analog system those two are joined by a multicore — a single thick cable carrying every channel individually, heavy enough to need two people and expensive enough to be treated carefully.

A digital system moves the preamps to the stage instead. The stage box is a rack of XLR inputs and outputs that sits by the stage, converts everything to digital there, and sends the lot back to the console down one thin network cable. That is why a digital rig loads in faster: the heaviest, most awkward cable in audio has been replaced by something you can coil in one hand.

Three practical consequences follow, and they all catch people out. The microphone plugs into the stage box, not the console, so when a channel is dead the first place to look is on stage rather than at the desk. The gain control on the console is operating a preamp thirty metres away, which is why two consoles sharing one stage box have to agree about gain. And the whole show now depends on one network cable, which is why anybody who has done this a while runs a second one.

You will hear it called a stage box, a stage rack, an I/O box or a digital snake, and they all mean the same object.

What comes in the box, and what you patch

The same job, done two ways. The right-hand column is what an analog desk needs added to match a digital one — and every added box is cable, weight, and something else to be wrong at eight o'clock.

Pre-fade or post-fade, and why it matters

The one aux detail worth getting right first time. Backwards is a memorable mistake.

On the job

A small band in a function room. Soundcheck goes fine, the mix in the room sounds right, and then the singer starts waving at you during the first song because her wedge keeps dropping out.

Nothing is broken. Every time you pull the guitar down in the room — which you are doing constantly, because it is too loud out front — it is also disappearing from her wedge. She is hearing the room mix with the volume changes you make for the audience, which is exactly what a musician does not want.

The monitor sends are on post-fade. They should be pre-fade: a monitor mix is meant to be independent of what the audience gets, so that pulling a channel down in the room leaves the stage untouched.

On most desks this is one switch per aux, or one global setting, and it takes fifteen seconds between songs. The reason it is worth knowing before the show rather than during it is that a singer who has lost confidence in her wedge in song one is a different singer for the rest of the set, and no amount of correct wedge mix afterwards gets that back.

How to do it

Meeting a console you have never used. Fifteen minutes, and it works on any desk of any size.

1. Find one channel strip and read it top to bottom. Every other strip is identical, so this is most of the desk learned.
2. Establish where gain is, and whether the desk calls it trim. It is the first control and the one everything else assumes.
3. Find phantom power and work out whether it is per channel or in banks. A whole bank switched at once will damage nothing but will surprise you.
4. Find the aux sends and establish whether they are pre-fade or post-fade, and how to change that. Do this before soundcheck, not during the show.
5. Trace one channel all the way to the main output, so you know where it is routed and what could stop it arriving.
6. On a digital desk, learn how to see what is currently selected. Adjusting the right control on the wrong channel is the most common digital mistake.
7. Find the mute and the solo, and check whether solo affects the main output or only your headphones. Discovering that live is memorable.

What goes wrong

Practice it

Six questions about the desk. Answer before you look.

Speakers, Amplification & Feedback

The end of the chain, where the signal becomes air again. Coverage and feedback both get taught here in full, because you cannot work a stage without them — what the equipment is, and the handful of things you actually change when it misbehaves.

How a loudspeaker works

A loudspeaker is a microphone running backwards. An electrical signal passes through a coil in a magnetic field, the coil moves, it is attached to a cone, and the cone pushes air. The pressure pattern that started at the source is recreated at the other end of the chain.

That symmetry is not a curiosity — it is why a loudspeaker pointed at an open microphone produces feedback, and it is why the same words describe both: diaphragm, coil, magnet, sensitivity.

The consequence worth carrying is that moving air is physical work. A loudspeaker is a machine doing mechanical labour, which is why it needs real power to drive it, why it gets hot, why it has a limit, and why the limit is reached by asking it to move more air than the cone can move — which sounds like distortion and then, if you keep asking, like an expensive silence.

Drivers, mains, subs, and monitors

No single driver reproduces the whole audible range well, because the job at each end is physically different. Moving enough air for a low note needs a large, heavy cone; responding fast enough for a high one needs a small, light one. So a loudspeaker box contains more than one driver, each covering part of the range, and a crossover inside decides which frequencies go where.

A woofer handles the low end and is large. A tweeter, usually a compression driver on a horn in professional boxes, handles the top and is small. A three-way box adds a midrange between them.

Beyond the drivers, the boxes themselves have jobs. A main is what the audience hears, flown or on a stick, aimed at the seats. A subwoofer handles only the lowest frequencies, and it exists because those need a physically bigger machine than the mains can contain — and because low frequencies are omnidirectional, a sub can go where it fits rather than where it points.

A monitor is a box aimed at the performers, and the two kinds behave differently. A wedge sits on the floor angled up at the performer, which means it is also a loudspeaker pointed at the stage microphones, and that is the entire reason wedges cause feedback. In-ear monitors put the mix in the performer's ears instead, which removes the loop completely — no wedge, no feedback path — and is why they have taken over.

Powered vs passive, and matching

A passive loudspeaker has no amplifier in it. It needs speaker-level signal from an external power amplifier, which means an amp rack, speaker cable, and somebody who has matched the amplifier to the box.

A powered, or active, loudspeaker has its amplifier built in. You give it line level and mains power, and it handles the rest. The amplifier inside is chosen by the manufacturer to suit the drivers, which removes the matching question entirely and is why powered boxes dominate small and mid-sized work.

For a technician the practical differences are these. A powered box needs a mains socket wherever it stands, which shapes where you can put things. A passive box needs no power at the box but does need speaker cable back to a rack, which is heavier and must not be confused with anything else.

And that last point is where the danger is. Never send speaker-level signal into a line input, and never send line level to a passive box expecting it to work. The first damages equipment; the second produces almost nothing and leaves people hunting a fault that is a patching mistake.

If you are working with passive boxes and somebody else has chosen the amplifier, the matching has been done. If you are choosing it, that is a system-tech question and worth asking rather than guessing — an underpowered amplifier driven into clipping destroys more loudspeakers than an oversized one ever has.

Coverage and a word on system tuning

Coverage is where the sound actually goes, and it is decided by physics rather than by preference. High frequencies are directional and travel in a beam; low frequencies spread in all directions. That is why a loudspeaker aimed away from you still sounds bass-heavy, and why aiming matters far more for intelligibility than for warmth.

Every professional box has a stated coverage angle — say ninety degrees horizontal by sixty vertical — describing the wedge of space it is designed to serve. The practical job is to aim those wedges so that they cover the seats and, as far as possible, miss the walls and ceiling, because sound that hits a hard surface comes back late and blurs everything that follows it.

The other rule is that one box cannot serve a deep room. The people at the front will be too loud before the people at the back are loud enough, because level falls off with distance. The answer is either more boxes covering different depths, or delay speakers further back — and a delay speaker must be delayed, by roughly three milliseconds per metre, or it arrives before the stage sound and destroys the intelligibility it was installed to improve.

System tuning — measuring a system and correcting it — is a specialism with its own tools and its own course. What belongs in a basics book is knowing that it exists, that it happens after the boxes are aimed rather than instead of aiming them, and that no amount of it fixes a loudspeaker pointed at a wall.

Feedback: cause and control

Feedback is a loop, not a mystery. A microphone hears a loudspeaker, that sound goes back through the system to the same loudspeaker, and it goes round again. When the sound coming back is as loud as the sound going in, the loop sustains itself and you get the ring or the howl.

Naming it as a loop is what makes it fixable, because a loop can be broken in four places and you can go down them in order of cost.

Distance from the source to the microphone is first, and it is free. Halving it gains about six decibels of the sound you want without adding anything to the loop — and that last clause is the whole point of it. Every other way of making somebody louder raises the loop by the same amount it raises the voice, because a fader cannot tell the two apart. Distance is the only lever that improves the ratio instead of moving both ends of it.

This is also why a source drifting away from a microphone is the most common cause of feedback that nobody recognises as one. The drift does not ring. It makes the voice quieter, somebody restores the level, and the restoring is what rings — several minutes later, at a moment that looks unrelated to anything anyone did.

How many microphones are open is second, and also free. Every doubling of open microphones costs you roughly three decibels before feedback, so a panel with four open costs about six decibels compared with one. Closing what nobody is speaking into is the largest single thing you control.

Where the loudspeakers point is third. A wedge meeting the dead side of a microphone's pattern is worth more than any equaliser, and it costs nothing but a minute of moving things.

Equalisation is fourth and last. Find the frequency that rings and cut it narrowly — a third of an octave or less. A wide cut takes the ring out and the voice with it. And stop after four or five: ringing out buys a few decibels, not fifteen, and a system needing more cuts than that has a geometry problem wearing an equaliser problem's clothes.

Powered or passive

Both work. The difference is what you have to carry, plug in, and get right.

Breaking the feedback loop, cheapest first

Four places to break it. The first three are free, and the fourth is the one everybody reaches for first.

On the job

An awards evening in a hotel ballroom. Two powered mains on sticks either side of a low stage, aimed out at the tables, and a lectern with a condenser on a gooseneck standing between them and a little behind the line they sit on. No monitor for the presenter, because a presenter at a lectern hears themselves perfectly well in the room and a wedge of their own voice would only spend gain you need elsewhere.

Nothing rings during setup. Ten minutes before doors, with the room filling, the presenter starts stepping back from the lectern to read the card — and within a minute the system is ringing.

Follow it in order, because the step backwards is not what rang. At the lectern the microphone is 150 mm from their mouth; a step back makes it 300, which costs about six decibels of voice. On its own that changes nothing dangerous: the loop is exactly as it was, the system is simply quieter, and the room says it cannot hear them.

So somebody turns it up. That is the moment the system rings, and it is worth being blunt about why: the fader does not know which sound is the presenter. Six decibels added to get the voice back is six decibels added to everything that microphone hears — including the mains spilling back onto the stage, which is the loop. Whatever headroom stood between the system and feedback has just been spent.

There is a second thing happening in the same move. At 150 mm the microphone hears mostly mouth; at 300 mm it hears a much larger share of room. What the operator is raising is not only quieter, it is a worse mixture — more air, more reflection, more of exactly the sound that completes the loop. That is why turning it up buys so much less than it should, and why the ring arrives before the presenter sounds right again.

Three fixes, in order of cost. Ask them to stay close to the microphone. This is the only one that gives the voice back WITHOUT adding anything to the loop, which is precisely what makes it the right answer and not merely the cheap one. It is free, it works immediately, and it is the thing every presenter forgets within ninety seconds. Look at where the lectern is standing: if it is level with the mains or in front of them it is sitting in their coverage, and moving it a metre upstage, or angling the boxes further out into the room, is ten minutes of work and permanent. Only then, if it still rings, find the frequency and take one narrow cut.

What nobody should do is turn the mains up because the back tables are complaining. That raises the loudest term in the loop, and the ring that was a threat during rehearsal arrives during the speech.

How to do it

Setting up and aiming a small system, and getting level before it rings. Half an hour, and it prevents most of what goes wrong later.

1. Work out whether the boxes are powered or passive before you carry anything. It decides whether you need mains at each position or a rack and speaker cable.
2. Place the mains so they cover the seats rather than the walls. Sound that hits a hard surface comes back late and blurs what follows.
3. Get them high enough that the front row is not deafened before the back row is served. On sticks, above head height.
4. Put any subwoofer where it fits. Low frequencies are omnidirectional, so a sub does not need aiming the way a main does.
5. Aim any wedge so it meets the dead side of the microphone it is nearest. This is free level before feedback and it is easiest before anything is taped down.
6. Ask what each wedge is FOR before you feed it. A performer's wedge carries their own voice; a wedge at a lectern is for hearing questions from the floor, and the presenter's own channel must never go into it.
7. Delay any distant loudspeaker by roughly three milliseconds per metre. An undelayed one adds level and destroys intelligibility.
8. Bring the system up slowly until it just begins to ring, then back off. Now you know where the ceiling is rather than guessing.
9. If it needs ringing out, cut narrowly and stop after four or five. If that is not enough, go back and change a position.

What goes wrong

Practice it

Seven questions. The feedback ones are the ones you will be paid for.

Where to Go Next

Audio is the largest department in the industry and the one with the most doors into it. What you have here is the equipment; every direction below is a craft built on top of it.

• If you finish this book also wanting to know why — why a room does that to a voice, why two loudspeakers are not twice as loud — The Physics of Sound takes the same air and asks exactly that, in far more depth than a working chapter can. It adds to what you have just learned rather than replacing any of it.
• If the stage interested you, go to A2 & Stage Audio Fundamentals, which is the first audio job most people are actually hired into.
• If the mixing interested you, go to Front-of-House Mixing, and then to monitors, which is its own craft.
• If the wireless interested you, go to Wireless & RF Systems. RF coordination travels well and pays well.
• If the problem-solving interested you most, that instinct is worth more than any of the above. It is what makes a technician the person who gets called.

Choosing without guessing

Those are honest directions and they are not a decision. What decides whether a route suits you is more ordinary than the subject: how much of your week is travel, whether you would rather be judged on taste or on reliability, whether you like the show or the problem, and what you already have that you are discounting because it did not happen in AV.

Start with the Opportunity Map, at https://reboot-opportunity-map-v1.vercel.app, which draws the industry as a transit map and shows what is genuinely next to the station you are standing on. Somebody who can trace an audio chain is closer to several well-paid seats than they usually believe. Looking around costs nothing and takes a few minutes, which is why it comes first: it is far easier to judge a route once you can see where it goes.

Then, if something on the map has your attention and you want to know whether it actually suits you, take the AV Pathfinder, at https://reboot-pathfinder.vercel.app. It is a short personal quiz, and what comes back is three AV paths that suit you with a Reboot course journey mapped to each — the specific courses, in order, between you and that path.

Both live inside the Reboot community, which is also where the rest of the course library sits, so you will be asked to sign in.

About Reboot AV

Reboot AV exists because the audiovisual industry has a map problem. The work is there, the pay is real, and the ceiling is high, but almost nobody entering it can see the route from where they are to where they want to be. People do not fail in this trade for lack of talent. They fail because nobody ever showed them what the path looked like.

So Reboot AV builds the thing that was missing: a course library covering the full breadth of live-event and installed audiovisual work, an interactive map of the roles in the industry and the routes between them, and an assessment that helps people work out where they actually fit rather than guessing.

Glossary

	Aux send	A copy of a channel sent somewhere else — a monitor wedge or an effect
	Balanced	A connection carrying the signal twice, once inverted, so interference cancels. Why audio uses XLR
	Bus	A destination several channels can be sent to at once
	Cardioid	The heart-shaped pattern of the standard live microphone. Dead directly behind
	Insert	A break in a channel's path where a processor is wired in — signal out to it, and back again. Two leads, and both have to be right
	Outboard	Processing that lives in a rack rather than in the console. What an analog desk needs added to match a digital one
	Patch bay	A panel bringing every input and output to one place, so processors are connected at the front instead of round the back
	Clipping	A signal driven past a stage's ceiling. Peaks squared off, and it sounds harsh
	Compression driver	The small driver on a horn that handles the high end of a professional box
	Condenser	A sensitive microphone needing phantom power. For detail, not for abuse
	Crossover	Decides which frequencies go to which driver inside a loudspeaker
	dBFS	How close a digital signal is to its ceiling. There is nothing above zero
	dB SPL	Actual loudness in a room, as opposed to an electrical level
	DI box	Turns an unbalanced instrument or laptop output into a balanced signal at the right level
	Dynamic	The tough microphone that needs no power. Vocals, snare, guitar cabs
	Fader	How much of a channel goes to a mix. The volume control during a show
	Feedback	The self-sustaining loop of a microphone hearing its own loudspeaker
	Figure-of-eight	A pattern hearing front and back equally, with the deepest side nulls of any pattern. Also called bidirectional
	Shotgun	A microphone with an interference tube giving a narrow forward lobe. The film and location workhorse. It rejects the sides rather than reaching further
	Gain	Sets how much the incoming signal is amplified to a workable level. Set first, then left alone
	Group	Several related channels on one fader before they join the main mix
	Headroom	The space deliberately left between the normal signal and the ceiling
	High-pass filter	Cuts everything below a set frequency. Nearly always worth it on a vocal
	In-ear monitors	A performer's mix in their ears rather than a wedge. Removes the feedback loop entirely
	Instrument level	Between mic and line. What a guitar sends, and the reason DI boxes exist
	Line level	The working currency of audio equipment, about a thousand times mic level
	Main bus	The mix the audience hears. Also the left-right, or the mains
	Mic level	Thousandths of a volt, straight from a microphone. Tiny and fragile
	Noise floor	The constant low-level hiss every piece of electronics produces
	Omnidirectional	A pattern hearing equally in every direction. Rarely right on a loud stage
	Pad	A switch dropping the input, usually by 20 dB, before the preamp
	Passive loudspeaker	A box with no amplifier in it. Needs speaker level from an amp rack
	Phantom power	48 V sent up the microphone cable to run a condenser
	Power amplifier	Drives a passive loudspeaker at speaker level. An underpowered one driven into clipping destroys boxes
	Polar pattern	The shape of what a microphone listens to
	Post-fade	A send taken after the fader, so it follows it. What an effect wants
	Powered loudspeaker	A box with its amplifier built in. Takes line level and a mains socket
	Pre-fade	A send taken before the fader, so it ignores it. What a monitor wants
	Ribbon	A delicate microphone with a particular smooth character. Mostly a studio tool
	Speaker level	Hundreds of times line level, from an amplifier to a passive box. Nothing else
	Subwoofer	A box for the lowest frequencies only. Can go where it fits, being omnidirectional
	Supercardioid	Tighter at the front than a cardioid, with a rear lobe. Its dead spots are off to the sides
	Three-to-one rule	Keep two microphones three times further apart than each is from its source
	Wedge	A floor monitor angled up at a performer, and therefore a loudspeaker pointed at the microphones
	Woofer	The large driver handling the low end of a loudspeaker box

What the gear actually looks like

Every device named in this book, in one place. Recognition is the whole point of Chapter 4 — use this page to attach a picture to each name before you meet it on a dock at six in the morning.

Audio

Dynamic microphone

Condenser microphone

Wireless handheld

Radio rack

Mixing console

Stage box / snake

Power amplifier

Main loudspeaker

Subwoofer

Stage monitor wedge

In-ear monitors

DI box

Video

Camera

Video switcher

Scaler / processor

Projector

LED wall panel

Confidence monitor

Lighting & control

Conventional fixture

Moving-head fixture

Lighting console

Dimmer

Show controller

Media server

Networking & infrastructure

Network switch

Patch bay

Equipment rack

Power distribution

Cable tester

A properly coiled cable

Connectors, close up

XLR

TRS / TS ¼-inch

SDI on BNC

HDMI

Cat5e / Cat6 (RJ45)

Fiber optic

DMX (5-pin XLR)

AC power — not signal